Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits)
  ALSA: usb - Use strlcat() correctly
  ALSA: Fix invalid __exit in sound/mips/*.c
  ALSA: hda - Fix / improve ALC66x parser
  ALSA: ctxfi: Swapped SURROUND-SIDE mute
  sound: Make keywest_driver static
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
  ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs
  ASoC: fix kconfig order of Blackfin drivers
  ALSA: hda - Added quirk to enable sound on Toshiba NB200
  ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
  ALSA: Don't assume i2c device probing always succeeds
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
  ALSA: echoaudio - Re-enable the line-out control for the Mia card
  ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
  ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
  ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
  ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
  ASoC: DaVinci: Correct McASP FIFO initialization
  ASoC: Davinci: Fix race with cpu_dai->dma_data
  ASoC: DaVinci: Fix divide by zero error during 1st execution
  ...

24 files changed, 512 insertions, 256 deletions

diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index f1708b79f963..75fddb40f416 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -209,6 +209,7 @@ AD1884A / AD1883 / AD1984A / AD1984B
   laptop        laptop with HP jack sensing
   mobile        mobile devices with HP jack sensing
   thinkpad      Lenovo Thinkpad X300
+  touchsmart    HP Touchsmart
 
 AD1884
 ======
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index f0ebc971c686..1dd66ddffcaf 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -897,6 +897,15 @@ static int tas_create(struct i2c_adapter *adapter,
         client = i2c_new_device(adapter, &info);
         if (!client)
                 return -ENODEV;
+        /*
+         * We know the driver is already loaded, so the device should be
+         * already bound. If not it means binding failed, and then there
+         * is no point in keeping the device instantiated.
+         */
+        if (!client->driver) {
+                i2c_unregister_device(client);
+                return -ENODEV;
+        }
 
         /*
          * Let i2c-core delete that device on driver removal.
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index c52691c2fc46..9a88cdfd952a 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -915,7 +915,7 @@ static int __devinit hal2_probe(struct platform_device *pdev)
         return 0;
 }
 
-static int __exit hal2_remove(struct platform_device *pdev)
+static int __devexit hal2_remove(struct platform_device *pdev)
 {
         struct snd_card *card = platform_get_drvdata(pdev);
 
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index e497525bc11b..8691f4cf6191 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -973,7 +973,7 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
         return 0;
 }
 
-static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
 {
         struct snd_card *card = platform_get_drvdata(pdev);
 
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index b1b3a644f738..75454648d50c 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1037,7 +1037,7 @@ static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
 
 static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
 {
-        return atc_daio_unmute(atc, state, LINEO4);
+        return atc_daio_unmute(atc, state, LINEO2);
 }
 
 static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
@@ -1047,7 +1047,7 @@ static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
 
 static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
 {
-        return atc_daio_unmute(atc, state, LINEO2);
+        return atc_daio_unmute(atc, state, LINEO4);
 }
 
 static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index da2065cd2c0d..1305f7ca02c3 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,7 +950,7 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
         Control interface
 ******************************************************************************/
 
-#ifndef ECHOCARD_HAS_VMIXER
+#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
 
 /******************* PCM output volume *******************/
 static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
@@ -1003,6 +1003,19 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
         return changed;
 }
 
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+/* On the Mia this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+        .name = "Line Playback Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+                  SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+        .info = snd_echo_output_gain_info,
+        .get = snd_echo_output_gain_get,
+        .put = snd_echo_output_gain_put,
+        .tlv = {.p = db_scale_output_gain},
+};
+#else
 static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
         .name = "PCM Playback Volume",
         .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1012,9 +1025,10 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
         .put = snd_echo_output_gain_put,
         .tlv = {.p = db_scale_output_gain},
 };
-
 #endif
 
+#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
+
 
 
 #ifdef ECHOCARD_HAS_INPUT_GAIN
@@ -2030,10 +2044,18 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
         snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
         if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
                 goto ctl_error;
-#else
-        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
+        err = snd_ctl_add(chip->card,
+                          snd_ctl_new1(&snd_echo_line_output_gain, chip));
+        if (err < 0)
                 goto ctl_error;
 #endif
+#else /* ECHOCARD_HAS_VMIXER */
+        err = snd_ctl_add(chip->card,
+                          snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
+        if (err < 0)
+                goto ctl_error;
+#endif /* ECHOCARD_HAS_VMIXER */
 
 #ifdef ECHOCARD_HAS_INPUT_GAIN
         if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
index f3b9b45c9c1b..f05c8c097aa8 100644
--- a/sound/pci/echoaudio/mia.c
+++ b/sound/pci/echoaudio/mia.c
@@ -29,6 +29,7 @@
 #define ECHOCARD_HAS_ADAT       FALSE
 #define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
 #define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_LINE_OUT_GAIN
 
 /* Pipe indexes */
 #define PX_ANALOG_OUT   0       /* 8 */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 20a66f85f0a4..c9ad182e1b4b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2303,6 +2303,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  * white-list for enable_msi
  */
 static struct snd_pci_quirk msi_white_list[] __devinitdata = {
+        SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1),
         SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
         {}
 };
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 215e72a87113..2d603f6aba63 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -4032,6 +4032,127 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec)
 }
 
 /*
+ * HP Touchsmart
+ * port-A (0x11)      - front hp-out
+ * port-B (0x14)      - unused
+ * port-C (0x15)      - unused
+ * port-D (0x12)      - rear line out
+ * port-E (0x1c)      - front mic-in
+ * port-F (0x16)      - Internal speakers
+ * digital-mic (0x17) - Internal mic
+ */
+
+static struct hda_verb ad1984a_touchsmart_verbs[] = {
+        /* DACs; unmute as default */
+        {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+        {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
+        /* Port-A (HP) mixer - route only from analog mixer */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        /* Port-A pin */
+        {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        /* Port-A (HP) pin - always unmuted */
+        {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        /* Port-E (int speaker) mixer - route only from analog mixer */
+        {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
+        /* Port-E pin */
+        {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+        {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        /* Port-F (int speaker) mixer - route only from analog mixer */
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        /* Port-F pin */
+        {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Analog mixer; mute as default */
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+        {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+        /* Analog Mix output amp */
+        {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* capture sources */
+        /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* unsolicited event for pin-sense */
+        {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+        {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+        /* allow to touch GPIO1 (for mute control) */
+        {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+        {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+        {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
+        /* internal mic - dmic */
+        {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+        /* set magic COEFs for dmic */
+        {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+        {0x01, AC_VERB_SET_PROC_COEF, 0x08},
+        { } /* end */
+};
+
+static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
+        HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
+/*      HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Master Playback Switch",
+                .info = snd_hda_mixer_amp_switch_info,
+                .get = snd_hda_mixer_amp_switch_get,
+                .put = ad1884a_mobile_master_sw_put,
+                .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+        },
+        HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
+        HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
+        HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
+        { } /* end */
+};
+
+/* switch to external mic if plugged */
+static void ad1984a_touchsmart_automic(struct hda_codec *codec)
+{
+        if (snd_hda_codec_read(codec, 0x1c, 0,
+                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) {
+                snd_hda_codec_write(codec, 0x0c, 0,
+                                     AC_VERB_SET_CONNECT_SEL, 0x4);
+        } else {
+                snd_hda_codec_write(codec, 0x0c, 0,
+                                     AC_VERB_SET_CONNECT_SEL, 0x5);
+        }
+}
+
+
+/* unsolicited event for HP jack sensing */
+static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
+        unsigned int res)
+{
+        switch (res >> 26) {
+        case AD1884A_HP_EVENT:
+                ad1884a_hp_automute(codec);
+                break;
+        case AD1884A_MIC_EVENT:
+                ad1984a_touchsmart_automic(codec);
+                break;
+        }
+}
+
+/* initialize jack-sensing, too */
+static int ad1984a_touchsmart_init(struct hda_codec *codec)
+{
+        ad198x_init(codec);
+        ad1884a_hp_automute(codec);
+        ad1984a_touchsmart_automic(codec);
+        return 0;
+}
+
+
+/*
  */
 
 enum {
@@ -4039,6 +4160,7 @@ enum {
         AD1884A_LAPTOP,
         AD1884A_MOBILE,
         AD1884A_THINKPAD,
+        AD1984A_TOUCHSMART,
         AD1884A_MODELS
 };
 
@@ -4047,6 +4169,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = {
         [AD1884A_LAPTOP]        = "laptop",
         [AD1884A_MOBILE]        = "mobile",
         [AD1884A_THINKPAD]      = "thinkpad",
+        [AD1984A_TOUCHSMART]    = "touchsmart",
 };
 
 static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
@@ -4059,6 +4182,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
         SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
         SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
         SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
+        SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
         {}
 };
 
@@ -4142,6 +4266,21 @@ static int patch_ad1884a(struct hda_codec *codec)
                 codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
                 codec->patch_ops.init = ad1984a_thinkpad_init;
                 break;
+        case AD1984A_TOUCHSMART:
+                spec->mixers[0] = ad1984a_touchsmart_mixers;
+                spec->init_verbs[0] = ad1984a_touchsmart_verbs;
+                spec->multiout.dig_out_nid = 0;
+                codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
+                codec->patch_ops.init = ad1984a_touchsmart_init;
+                /* set the upper-limit for mixer amp to 0dB for avoiding the
+                 * possible damage by overloading
+                 */
+                snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
+                                          (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+                                          (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+                                          (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+                                          (1 << AC_AMPCAP_MUTE_SHIFT));
+                break;
         }
 
         return 0;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 9d899eda44d7..3fbbc8c01e70 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -682,11 +682,13 @@ static struct hda_input_mux cxt5045_capture_source = {
 };
 
 static struct hda_input_mux cxt5045_capture_source_benq = {
-        .num_items = 3,
+        .num_items = 5,
         .items = {
                 { "IntMic", 0x1 },
                 { "ExtMic", 0x2 },
                 { "LineIn", 0x3 },
+                { "CD",     0x4 },
+                { "Mixer",  0x0 },
         }
 };
 
@@ -811,11 +813,19 @@ static struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
+        HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT),
+
         HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT),
         HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT),
         HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT),
         HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT),
 
+        HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT),
+
         {}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 129605819560..7810d3dcad83 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12660,7 +12660,7 @@ static struct alc_config_preset alc268_presets[] = {
                 .init_hook = alc268_toshiba_automute,
         },
         [ALC268_ACER] = {
-                .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
+                .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
                             alc268_beep_mixer },
                 .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
                                 alc268_acer_verbs },
@@ -16852,6 +16852,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
         SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
         SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
                       ALC662_3ST_6ch_DIG),
+        SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
         SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
         SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
                       ALC662_3ST_6ch_DIG),
@@ -17145,70 +17146,145 @@ static struct alc_config_preset alc662_presets[] = {
  * BIOS auto configuration
  */
 
+/* convert from MIX nid to DAC */
+static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
+{
+        if (nid == 0x0f)
+                return 0x02;
+        else if (nid >= 0x0c && nid <= 0x0e)
+                return nid - 0x0c + 0x02;
+        else
+                return 0;
+}
+
+/* get MIX nid connected to the given pin targeted to DAC */
+static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
+                                   hda_nid_t dac)
+{
+        hda_nid_t mix[4];
+        int i, num;
+
+        num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
+        for (i = 0; i < num; i++) {
+                if (alc662_mix_to_dac(mix[i]) == dac)
+                        return mix[i];
+        }
+        return 0;
+}
+
+/* look for an empty DAC slot */
+static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+        struct alc_spec *spec = codec->spec;
+        hda_nid_t srcs[5];
+        int i, j, num;
+
+        num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
+        if (num < 0)
+                return 0;
+        for (i = 0; i < num; i++) {
+                hda_nid_t nid = alc662_mix_to_dac(srcs[i]);
+                if (!nid)
+                        continue;
+                for (j = 0; j < spec->multiout.num_dacs; j++)
+                        if (spec->multiout.dac_nids[j] == nid)
+                                break;
+                if (j >= spec->multiout.num_dacs)
+                        return nid;
+        }
+        return 0;
+}
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc662_auto_fill_dac_nids(struct hda_codec *codec,
+                                     const struct auto_pin_cfg *cfg)
+{
+        struct alc_spec *spec = codec->spec;
+        int i;
+        hda_nid_t dac;
+
+        spec->multiout.dac_nids = spec->private_dac_nids;
+        for (i = 0; i < cfg->line_outs; i++) {
+                dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]);
+                if (!dac)
+                        continue;
+                spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+        }
+        return 0;
+}
+
+static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx,
+                              hda_nid_t nid, unsigned int chs)
+{
+        char name[32];
+        sprintf(name, "%s Playback Volume", pfx);
+        return add_control(spec, ALC_CTL_WIDGET_VOL, name,
+                           HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
+static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx,
+                             hda_nid_t nid, unsigned int chs)
+{
+        char name[32];
+        sprintf(name, "%s Playback Switch", pfx);
+        return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+                           HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT));
+}
+
+#define alc662_add_stereo_vol(spec, pfx, nid) \
+        alc662_add_vol_ctl(spec, pfx, nid, 3)
+#define alc662_add_stereo_sw(spec, pfx, nid) \
+        alc662_add_sw_ctl(spec, pfx, nid, 3)
+
 /* add playback controls from the parsed DAC table */
-static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec,
                                              const struct auto_pin_cfg *cfg)
 {
-        char name[32];
+        struct alc_spec *spec = codec->spec;
         static const char *chname[4] = {
                 "Front", "Surround", NULL /*CLFE*/, "Side"
         };
-        hda_nid_t nid;
+        hda_nid_t nid, mix;
         int i, err;
 
         for (i = 0; i < cfg->line_outs; i++) {
-                if (!spec->multiout.dac_nids[i])
+                nid = spec->multiout.dac_nids[i];
+                if (!nid)
+                        continue;
+                mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid);
+                if (!mix)
                         continue;
-                nid = alc880_idx_to_dac(i);
                 if (i == 2) {
                         /* Center/LFE */
-                        err = add_control(spec, ALC_CTL_WIDGET_VOL,
-                                          "Center Playback Volume",
-                                          HDA_COMPOSE_AMP_VAL(nid, 1, 0,
-                                                              HDA_OUTPUT));
+                        err = alc662_add_vol_ctl(spec, "Center", nid, 1);
                         if (err < 0)
                                 return err;
-                        err = add_control(spec, ALC_CTL_WIDGET_VOL,
-                                          "LFE Playback Volume",
-                                          HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-                                                              HDA_OUTPUT));
+                        err = alc662_add_vol_ctl(spec, "LFE", nid, 2);
                         if (err < 0)
                                 return err;
-                        err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-                                          "Center Playback Switch",
-                                          HDA_COMPOSE_AMP_VAL(0x0e, 1, 0,
-                                                              HDA_INPUT));
+                        err = alc662_add_sw_ctl(spec, "Center", mix, 1);
                         if (err < 0)
                                 return err;
-                        err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-                                          "LFE Playback Switch",
-                                          HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
-                                                              HDA_INPUT));
+                        err = alc662_add_sw_ctl(spec, "LFE", mix, 2);
                         if (err < 0)
                                 return err;
                 } else {
                         const char *pfx;
                         if (cfg->line_outs == 1 &&
                             cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
-                                if (!cfg->hp_pins)
+                                if (cfg->hp_outs)
                                         pfx = "Speaker";
                                 else
                                         pfx = "PCM";
                         } else
                                 pfx = chname[i];
-                        sprintf(name, "%s Playback Volume", pfx);
-                        err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-                                          HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-                                                              HDA_OUTPUT));
+                        err = alc662_add_vol_ctl(spec, pfx, nid, 3);
                         if (err < 0)
                                 return err;
                         if (cfg->line_outs == 1 &&
                             cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
                                 pfx = "Speaker";
-                        sprintf(name, "%s Playback Switch", pfx);
-                        err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-                                HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
-                                                    3, 0, HDA_INPUT));
+                        err = alc662_add_sw_ctl(spec, pfx, mix, 3);
                         if (err < 0)
                                 return err;
                 }
@@ -17217,54 +17293,38 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 /* add playback controls for speaker and HP outputs */
-static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
+/* return DAC nid if any new DAC is assigned */
+static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin,
                                         const char *pfx)
 {
-        hda_nid_t nid;
+        struct alc_spec *spec = codec->spec;
+        hda_nid_t nid, mix;
         int err;
-        char name[32];
 
         if (!pin)
                 return 0;
-
-        if (pin == 0x17) {
-                /* ALC663 has a mono output pin on 0x17 */
+        nid = alc662_look_for_dac(codec, pin);
+        if (!nid) {
+                char name[32];
+                /* the corresponding DAC is already occupied */
+                if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP))
+                        return 0; /* no way */
+                /* create a switch only */
                 sprintf(name, "%s Playback Switch", pfx);
-                err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-                                  HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT));
-                return err;
+                return add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+                                   HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
         }
 
-        if (alc880_is_fixed_pin(pin)) {
-                nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-                /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
-                /* specify the DAC as the extra output */
-                if (!spec->multiout.hp_nid)
-                        spec->multiout.hp_nid = nid;
-                else
-                        spec->multiout.extra_out_nid[0] = nid;
-                /* control HP volume/switch on the output mixer amp */
-                nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-                sprintf(name, "%s Playback Volume", pfx);
-                err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-                                  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
-                if (err < 0)
-                        return err;
-                sprintf(name, "%s Playback Switch", pfx);
-                err = add_control(spec, ALC_CTL_BIND_MUTE, name,
-                                  HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT));
-                if (err < 0)
-                        return err;
-        } else if (alc880_is_multi_pin(pin)) {
-                /* set manual connection */
-                /* we have only a switch on HP-out PIN */
-                sprintf(name, "%s Playback Switch", pfx);
-                err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
-                                  HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT));
-                if (err < 0)
-                        return err;
-        }
-        return 0;
+        mix = alc662_dac_to_mix(codec, pin, nid);
+        if (!mix)
+                return 0;
+        err = alc662_add_vol_ctl(spec, pfx, nid, 3);
+        if (err < 0)
+                return err;
+        err = alc662_add_sw_ctl(spec, pfx, mix, 3);
+        if (err < 0)
+                return err;
+        return nid;
 }
 
 /* create playback/capture controls for input pins */
@@ -17273,30 +17333,35 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 
 static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
                                               hda_nid_t nid, int pin_type,
-                                              int dac_idx)
+                                              hda_nid_t dac)
 {
+        int i, num;
+        hda_nid_t srcs[4];
+
         alc_set_pin_output(codec, nid, pin_type);
         /* need the manual connection? */
-        if (alc880_is_multi_pin(nid)) {
-                struct alc_spec *spec = codec->spec;
-                int idx = alc880_multi_pin_idx(nid);
-                snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
-                                    AC_VERB_SET_CONNECT_SEL,
-                                    alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+        num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs));
+        if (num <= 1)
+                return;
+        for (i = 0; i < num; i++) {
+                if (alc662_mix_to_dac(srcs[i]) != dac)
+                        continue;
+                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i);
+                return;
         }
 }
 
 static void alc662_auto_init_multi_out(struct hda_codec *codec)
 {
         struct alc_spec *spec = codec->spec;
+        int pin_type = get_pin_type(spec->autocfg.line_out_type);
         int i;
 
         for (i = 0; i <= HDA_SIDE; i++) {
                 hda_nid_t nid = spec->autocfg.line_out_pins[i];
-                int pin_type = get_pin_type(spec->autocfg.line_out_type);
                 if (nid)
                         alc662_auto_set_output_and_unmute(codec, nid, pin_type,
-                                                          i);
+                                        spec->multiout.dac_nids[i]);
         }
 }
 
@@ -17306,12 +17371,13 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
         hda_nid_t pin;
 
         pin = spec->autocfg.hp_pins[0];
-        if (pin) /* connect to front */
-                /* use dac 0 */
-                alc662_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+        if (pin)
+                alc662_auto_set_output_and_unmute(codec, pin, PIN_HP,
+                                                  spec->multiout.hp_nid);
         pin = spec->autocfg.speaker_pins[0];
         if (pin)
-                alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+                alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+                                        spec->multiout.extra_out_nid[0]);
 }
 
 #define ALC662_PIN_CD_NID               ALC880_PIN_CD_NID
@@ -17349,21 +17415,25 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
         if (!spec->autocfg.line_outs)
                 return 0; /* can't find valid BIOS pin config */
 
-        err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+        err = alc662_auto_fill_dac_nids(codec, &spec->autocfg);
         if (err < 0)
                 return err;
-        err = alc662_auto_create_multi_out_ctls(spec, &spec->autocfg);
+        err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg);
         if (err < 0)
                 return err;
-        err = alc662_auto_create_extra_out(spec,
+        err = alc662_auto_create_extra_out(codec,
                                            spec->autocfg.speaker_pins[0],
                                            "Speaker");
         if (err < 0)
                 return err;
-        err = alc662_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+        if (err)
+                spec->multiout.extra_out_nid[0] = err;
+        err = alc662_auto_create_extra_out(codec, spec->autocfg.hp_pins[0],
                                            "Headphone");
         if (err < 0)
                 return err;
+        if (err)
+                spec->multiout.hp_nid = err;
         err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
         if (err < 0)
                 return err;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 826137ec3002..a9b26828a651 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -182,8 +182,8 @@ struct sigmatel_jack {
 
 struct sigmatel_mic_route {
         hda_nid_t pin;
-        unsigned char mux_idx;
-        unsigned char dmux_idx;
+        signed char mux_idx;
+        signed char dmux_idx;
 };
 
 struct sigmatel_spec {
@@ -3469,18 +3469,26 @@ static int set_mic_route(struct hda_codec *codec,
                         break;
         if (i <= AUTO_PIN_FRONT_MIC) {
                 /* analog pin */
-                mic->dmux_idx = 0;
                 i = get_connection_index(codec, spec->mux_nids[0], pin);
                 if (i < 0)
                         return -1;
                 mic->mux_idx = i;
+                mic->dmux_idx = -1;
+                if (spec->dmux_nids)
+                        mic->dmux_idx = get_connection_index(codec,
+                                                             spec->dmux_nids[0],
+                                                             spec->mux_nids[0]);
         }  else if (spec->dmux_nids) {
                 /* digital pin */
-                mic->mux_idx = 0;
                 i = get_connection_index(codec, spec->dmux_nids[0], pin);
                 if (i < 0)
                         return -1;
                 mic->dmux_idx = i;
+                mic->mux_idx = -1;
+                if (spec->mux_nids)
+                        mic->mux_idx = get_connection_index(codec,
+                                                            spec->mux_nids[0],
+                                                            spec->dmux_nids[0]);
         }
         return 0;
 }
@@ -4557,11 +4565,11 @@ static void stac92xx_mic_detect(struct hda_codec *codec)
                 mic = &spec->ext_mic;
         else
                 mic = &spec->int_mic;
-        if (mic->dmux_idx)
+        if (mic->dmux_idx >= 0)
                 snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
                                           AC_VERB_SET_CONNECT_SEL,
                                           mic->dmux_idx);
-        else
+        if (mic->mux_idx >= 0)
                 snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
                                           AC_VERB_SET_CONNECT_SEL,
                                           mic->mux_idx);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 171ada535209..754867ed4785 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1954,6 +1954,18 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
                 .name = "Sony S1XP",
                 .type = AC97_TUNE_INV_EAPD
         },
+        {
+                .subvendor = 0x104d,
+                .subdevice = 0x81c0,
+                .name = "Sony VAIO VGN-T350P", /*AD1981B*/
+                .type = AC97_TUNE_INV_EAPD
+        },
+        {
+                .subvendor = 0x104d,
+                .subdevice = 0x81c5,
+                .name = "Sony VAIO VGN-B1VP", /*AD1981B*/
+                .type = AC97_TUNE_INV_EAPD
+        },
         {
                 .subvendor = 0x1043,
                 .subdevice = 0x80f3,
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index 835fa19ed461..d06f780bd7e8 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -59,6 +59,18 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter)
         strlcpy(info.type, "keywest", I2C_NAME_SIZE);
         info.addr = keywest_ctx->addr;
         keywest_ctx->client = i2c_new_device(adapter, &info);
+        if (!keywest_ctx->client)
+                return -ENODEV;
+        /*
+         * We know the driver is already loaded, so the device should be
+         * already bound. If not it means binding failed, and then there
+         * is no point in keeping the device instantiated.
+         */
+        if (!keywest_ctx->client->driver) {
+                i2c_unregister_device(keywest_ctx->client);
+                keywest_ctx->client = NULL;
+                return -ENODEV;
+        }
         
         /*
          * Let i2c-core delete that device on driver removal.
@@ -86,7 +98,7 @@ static const struct i2c_device_id keywest_i2c_id[] = {
         { }
 };
 
-struct i2c_driver keywest_driver = {
+static struct i2c_driver keywest_driver = {
         .driver = {
                 .name = "PMac Keywest Audio",
         },
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index ac927ffdc961..97f1a251e446 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -7,15 +7,6 @@ config SND_BF5XX_I2S
           mode (supports single stereo In/Out).
           You will also need to select the audio interfaces to support below.
 
-config SND_BF5XX_TDM
-        tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
-        depends on (BLACKFIN && SND_SOC)
-        help
-          Say Y or M if you want to add support for codecs attached to
-          the Blackfin SPORT (synchronous serial ports) interface in TDM
-          mode.
-          You will also need to select the audio interfaces to support below.
-
 config SND_BF5XX_SOC_SSM2602
         tristate "SoC SSM2602 Audio support for BF52x ezkit"
         depends on SND_BF5XX_I2S
@@ -41,6 +32,31 @@ config SND_BFIN_AD73311_SE
           Enter the GPIO used to control AD73311's SE pin. Acceptable
           values are 0 to 7
 
+config SND_BF5XX_TDM
+        tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip"
+        depends on (BLACKFIN && SND_SOC)
+        help
+          Say Y or M if you want to add support for codecs attached to
+          the Blackfin SPORT (synchronous serial ports) interface in TDM
+          mode.
+          You will also need to select the audio interfaces to support below.
+
+config SND_BF5XX_SOC_AD1836
+        tristate "SoC AD1836 Audio support for BF5xx"
+        depends on SND_BF5XX_TDM
+        select SND_BF5XX_SOC_TDM
+        select SND_SOC_AD1836
+        help
+          Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
+config SND_BF5XX_SOC_AD1938
+        tristate "SoC AD1938 Audio support for Blackfin"
+        depends on SND_BF5XX_TDM
+        select SND_BF5XX_SOC_TDM
+        select SND_SOC_AD1938
+        help
+          Say Y if you want to add support for AD1938 codec on Blackfin.
+
 config SND_BF5XX_AC97
         tristate "SoC AC97 Audio for the ADI BF5xx chip"
         depends on BLACKFIN
@@ -71,6 +87,30 @@ config SND_BF5XX_MULTICHAN_SUPPORT
           Say y if you want AC97 driver to support up to 5.1 channel audio.
           this mode will consume much more memory for DMA.
 
+config SND_BF5XX_HAVE_COLD_RESET
+        bool "BOARD has COLD Reset GPIO"
+        depends on SND_BF5XX_AC97
+        default y if BFIN548_EZKIT
+        default n if !BFIN548_EZKIT
+
+config SND_BF5XX_RESET_GPIO_NUM
+        int "Set a GPIO for cold reset"
+        depends on SND_BF5XX_HAVE_COLD_RESET
+        range 0 159
+        default 19 if BFIN548_EZKIT
+        default 5 if BFIN537_STAMP
+        default 0
+        help
+          Set the correct GPIO for RESET the sound chip.
+
+config SND_BF5XX_SOC_AD1980
+        tristate "SoC AD1980/1 Audio support for BF5xx"
+        depends on SND_BF5XX_AC97
+        select SND_BF5XX_SOC_AC97
+        select SND_SOC_AD1980
+        help
+          Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
+
 config SND_BF5XX_SOC_SPORT
         tristate
 
@@ -88,30 +128,6 @@ config SND_BF5XX_SOC_AC97
         select SND_SOC_AC97_BUS
         select SND_BF5XX_SOC_SPORT
 
-config SND_BF5XX_SOC_AD1836
-        tristate "SoC AD1836 Audio support for BF5xx"
-        depends on SND_BF5XX_TDM
-        select SND_BF5XX_SOC_TDM
-        select SND_SOC_AD1836
-        help
-          Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1980
-        tristate "SoC AD1980/1 Audio support for BF5xx"
-        depends on SND_BF5XX_AC97
-        select SND_BF5XX_SOC_AC97
-        select SND_SOC_AD1980
-        help
-          Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT.
-
-config SND_BF5XX_SOC_AD1938
-        tristate "SoC AD1938 Audio support for Blackfin"
-        depends on SND_BF5XX_TDM
-        select SND_BF5XX_SOC_TDM
-        select SND_SOC_AD1938
-        help
-          Say Y if you want to add support for AD1938 codec on Blackfin.
-
 config SND_BF5XX_SPORT_NUM
         int "Set a SPORT for Sound chip"
         depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
@@ -120,19 +136,3 @@ config SND_BF5XX_SPORT_NUM
         default 0
         help
           Set the correct SPORT for sound chip.
-
-config SND_BF5XX_HAVE_COLD_RESET
-        bool "BOARD has COLD Reset GPIO"
-        depends on SND_BF5XX_AC97
-        default y if BFIN548_EZKIT
-        default n if !BFIN548_EZKIT
-
-config SND_BF5XX_RESET_GPIO_NUM
-        int "Set a GPIO for cold reset"
-        depends on SND_BF5XX_HAVE_COLD_RESET
-        range 0 159
-        default 19 if BFIN548_EZKIT
-        default 5 if BFIN537_STAMP
-        default 0
-        help
-          Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 1e9d161c76c4..084b68884ada 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -77,12 +77,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 3096badf09a5..ff546e91a22e 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -78,12 +78,12 @@ static struct sport_param sport_params[2] = {
  * TFS. When Port G is selected and EMAC then there is a conflict between
  * the PHY interrupt line and TFS.  Current settings prevent the conflict
  * by ignoring the TFS pin when Port G is selected. This allows both
- * ssm2602 using Port G and EMAC concurrently.
+ * codecs and EMAC using Port G concurrently.
  */
-#ifdef CONFIG_BF527_SPORT0_PORTF
-#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
-#else
+#ifdef CONFIG_BF527_SPORT0_PORTG
 #define LOCAL_SPORT0_TFS (0)
+#else
+#define LOCAL_SPORT0_TFS (P_SPORT0_TFS)
 #endif
 
 static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS,
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 12a6c549ee6e..4ae707048021 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,22 +97,19 @@ enum {
         DAVINCI_MCBSP_WORD_32,
 };
 
-static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
-        .name = "I2S PCM Stereo out",
-};
-
-static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
-        .name = "I2S PCM Stereo in",
-};
-
 struct davinci_mcbsp_dev {
+        /*
+         * dma_params must be first because rtd->dai->cpu_dai->private_data
+         * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+         * davinci_pcm_open.
+         */
+        struct davinci_pcm_dma_params   dma_params[2];
         void __iomem                    *base;
 #define MOD_DSP_A       0
 #define MOD_DSP_B       1
         int                             mode;
         u32                             pcr;
         struct clk                      *clk;
-        struct davinci_pcm_dma_params   *dma_params[2];
 };
 
 static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -215,14 +212,6 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
         toggle_clock(dev, playback);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream,
-                               struct snd_soc_dai *cpu_dai)
-{
-        struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
-        cpu_dai->dma_data = dev->dma_params[substream->stream];
-        return 0;
-}
-
 #define DEFAULT_BITPERSAMPLE    16
 
 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -353,8 +342,9 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
                                  struct snd_pcm_hw_params *params,
                                  struct snd_soc_dai *dai)
 {
-        struct davinci_pcm_dma_params *dma_params = dai->dma_data;
         struct davinci_mcbsp_dev *dev = dai->private_data;
+        struct davinci_pcm_dma_params *dma_params =
+                                        &dev->dma_params[substream->stream];
         struct snd_interval *i = NULL;
         int mcbsp_word_length;
         unsigned int rcr, xcr, srgr;
@@ -472,7 +462,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 #define DAVINCI_I2S_RATES       SNDRV_PCM_RATE_8000_96000
 
 static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
-        .startup        = davinci_i2s_startup,
         .shutdown       = davinci_i2s_shutdown,
         .prepare        = davinci_i2s_prepare,
         .trigger        = davinci_i2s_trigger,
@@ -534,12 +523,10 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 
         dev->base = (void __iomem *)IO_ADDRESS(mem->start);
 
-        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
-        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
             (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
 
-        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
-        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
             (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
 
         /* first TX, then RX */
@@ -549,7 +536,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
                 ret = -ENXIO;
                 goto err_free_mem;
         }
-        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start;
+        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
 
         res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
         if (!res) {
@@ -557,7 +544,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
                 ret = -ENXIO;
                 goto err_free_mem;
         }
-        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start;
+        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
         davinci_i2s_dai.private_data = dev;
         ret = snd_soc_register_dai(&davinci_i2s_dai);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 7a06c0a86665..5d1f98a4c978 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -332,14 +332,6 @@ static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val)
                 printk(KERN_ERR "GBLCTL write error\n");
 }
 
-static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
-                                                struct snd_soc_dai *cpu_dai)
-{
-        struct davinci_audio_dev *dev = cpu_dai->private_data;
-        cpu_dai->dma_data = dev->dma_params[substream->stream];
-        return 0;
-}
-
 static void mcasp_start_rx(struct davinci_audio_dev *dev)
 {
         mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
@@ -386,17 +378,17 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev)
 
 static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream)
 {
-        if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+        if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                if (dev->txnumevt)      /* enable FIFO */
+                        mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+                                                                FIFO_ENABLE);
                 mcasp_start_tx(dev);
-        else
+        } else {
+                if (dev->rxnumevt)      /* enable FIFO */
+                        mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+                                                                FIFO_ENABLE);
                 mcasp_start_rx(dev);
-
-        /* enable FIFO */
-        if (dev->txnumevt)
-                mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-        if (dev->rxnumevt)
-                mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+        }
 }
 
 static void mcasp_stop_rx(struct davinci_audio_dev *dev)
@@ -413,17 +405,17 @@ static void mcasp_stop_tx(struct davinci_audio_dev *dev)
 
 static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream)
 {
-        if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+        if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                if (dev->txnumevt)      /* disable FIFO */
+                        mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
+                                                                FIFO_ENABLE);
                 mcasp_stop_tx(dev);
-        else
+        } else {
+                if (dev->rxnumevt)      /* disable FIFO */
+                        mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
+                                                                FIFO_ENABLE);
                 mcasp_stop_rx(dev);
-
-        /* disable FIFO */
-        if (dev->txnumevt)
-                mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
-
-        if (dev->rxnumevt)
-                mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
+        }
 }
 
 static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
@@ -720,7 +712,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 {
         struct davinci_audio_dev *dev = cpu_dai->private_data;
         struct davinci_pcm_dma_params *dma_params =
-                                        dev->dma_params[substream->stream];
+                                        &dev->dma_params[substream->stream];
         int word_length;
         u8 numevt;
 
@@ -798,7 +790,6 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 }
 
 static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
-        .startup        = davinci_mcasp_startup,
         .trigger        = davinci_mcasp_trigger,
         .hw_params      = davinci_mcasp_hw_params,
         .set_fmt        = davinci_mcasp_set_dai_fmt,
@@ -849,20 +840,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
         struct resource *mem, *ioarea, *res;
         struct snd_platform_data *pdata;
         struct davinci_audio_dev *dev;
-        int count = 0;
         int ret = 0;
 
         dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL);
         if (!dev)
                 return  -ENOMEM;
 
-        dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2,
-                                                                GFP_KERNEL);
-        if (!dma_data) {
-                ret = -ENOMEM;
-                goto err_release_dev;
-        }
-
         mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
         if (!mem) {
                 dev_err(&pdev->dev, "no mem resource?\n");
@@ -897,11 +880,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
         dev->txnumevt = pdata->txnumevt;
         dev->rxnumevt = pdata->rxnumevt;
 
-        dma_data[count].name = "I2S PCM Stereo out";
-        dma_data[count].eventq_no = pdata->eventq_no;
-        dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
+        dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+        dma_data->eventq_no = pdata->eventq_no;
+        dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
                                                         io_v2p(dev->base));
-        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count];
 
         /* first TX, then RX */
         res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
@@ -910,13 +892,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
                 goto err_release_region;
         }
 
-        dma_data[count].channel = res->start;
-        count++;
-        dma_data[count].name = "I2S PCM Stereo in";
-        dma_data[count].eventq_no = pdata->eventq_no;
-        dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
+        dma_data->channel = res->start;
+
+        dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+        dma_data->eventq_no = pdata->eventq_no;
+        dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
                                                         io_v2p(dev->base));
-        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count];
 
         res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
         if (!res) {
@@ -924,7 +905,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
                 goto err_release_region;
         }
 
-        dma_data[count].channel = res->start;
+        dma_data->channel = res->start;
         davinci_mcasp_dai[pdata->op_mode].private_data = dev;
         davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
         ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
@@ -936,8 +917,6 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 err_release_region:
         release_mem_region(mem->start, (mem->end - mem->start) + 1);
 err_release_data:
-        kfree(dma_data);
-err_release_dev:
         kfree(dev);
 
         return ret;
@@ -946,7 +925,6 @@ err_release_dev:
 static int davinci_mcasp_remove(struct platform_device *pdev)
 {
         struct snd_platform_data *pdata = pdev->dev.platform_data;
-        struct davinci_pcm_dma_params *dma_data;
         struct davinci_audio_dev *dev;
         struct resource *mem;
 
@@ -959,8 +937,6 @@ static int davinci_mcasp_remove(struct platform_device *pdev)
         mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
         release_mem_region(mem->start, (mem->end - mem->start) + 1);
 
-        dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
-        kfree(dma_data);
         kfree(dev);
 
         return 0;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 554354c1cc2f..9d179cc88f7b 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,10 +39,15 @@ enum {
 };
 
 struct davinci_audio_dev {
+        /*
+         * dma_params must be first because rtd->dai->cpu_dai->private_data
+         * is cast to a pointer of an array of struct davinci_pcm_dma_params in
+         * davinci_pcm_open.
+         */
+        struct davinci_pcm_dma_params dma_params[2];
         void __iomem *base;
         int sample_rate;
         struct clk *clk;
-        struct davinci_pcm_dma_params *dma_params[2];
         unsigned int codec_fmt;
 
         /* McASP specific data */
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 2f7da49ed34f..c73a915f233f 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -126,16 +126,9 @@ static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 {
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
         struct edmacc_param p_ram;
         int ret;
 
-        if (!dma_data)
-                return -ENODEV;
-
-        prtd->params = dma_data;
-
         /* Request master DMA channel */
         ret = edma_alloc_channel(prtd->params->channel,
                                   davinci_pcm_dma_irq, substream,
@@ -244,6 +237,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct davinci_runtime_data *prtd;
         int ret = 0;
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
+        struct davinci_pcm_dma_params *params = &pa[substream->stream];
+        if (!params)
+                return -ENODEV;
 
         snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
         /* ensure that buffer size is a multiple of period size */
@@ -257,6 +255,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
                 return -ENOMEM;
 
         spin_lock_init(&prtd->lock);
+        prtd->params = params;
 
         runtime->private_data = prtd;
 
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 63d96253c73a..8746606efc89 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -17,7 +17,6 @@
 
 
 struct davinci_pcm_dma_params {
-        char *name;                     /* stream identifier */
         int channel;                    /* sync dma channel ID */
         unsigned short acnt;
         dma_addr_t dma_addr;            /* device physical address for DMA */
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 6375b4ea525d..dcb3181bb340 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -138,7 +138,7 @@ config SND_PXA2XX_SOC_MIOA701
 
 config SND_PXA2XX_SOC_IMOTE2
        tristate "SoC Audio support for IMote 2"
-       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C
        select SND_PXA2XX_SOC_I2S
        select SND_SOC_WM8940
        help
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index ab5a3ac2ac47..9efcfd08d747 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -898,6 +898,11 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = {
  * build a feature control
  */
 
+static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
+{
+        return strlcat(kctl->id.name, str, sizeof(kctl->id.name));
+}
+
 static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
                               unsigned int ctl_mask, int control,
                               struct usb_audio_term *iterm, int unitid)
@@ -978,13 +983,13 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc,
                  */
                 if (! mapped_name && ! (state->oterm.type >> 16)) {
                         if ((state->oterm.type & 0xff00) == 0x0100) {
-                                len = strlcat(kctl->id.name, " Capture", sizeof(kctl->id.name));
+                                len = append_ctl_name(kctl, " Capture");
                         } else {
-                                len = strlcat(kctl->id.name + len, " Playback", sizeof(kctl->id.name));
+                                len = append_ctl_name(kctl, " Playback");
                         }
                 }
-                strlcat(kctl->id.name + len, control == USB_FEATURE_MUTE ? " Switch" : " Volume",
-                        sizeof(kctl->id.name));
+                append_ctl_name(kctl, control == USB_FEATURE_MUTE ?
+                                " Switch" : " Volume");
                 if (control == USB_FEATURE_VOLUME) {
                         kctl->tlv.c = mixer_vol_tlv;
                         kctl->vd[0].access |= 
@@ -1143,7 +1148,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc,
                 len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0);
         if (! len)
                 len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
-        strlcat(kctl->id.name + len, " Volume", sizeof(kctl->id.name));
+        append_ctl_name(kctl, " Volume");
 
         snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
                     cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1400,8 +1405,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned
                         if (! len)
                                 strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
                 }
-                strlcat(kctl->id.name, " ", sizeof(kctl->id.name));
-                strlcat(kctl->id.name, valinfo->suffix, sizeof(kctl->id.name));
+                append_ctl_name(kctl, " ");
+                append_ctl_name(kctl, valinfo->suffix);
 
                 snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
                             cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
@@ -1610,9 +1615,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi
                         strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
                 if ((state->oterm.type & 0xff00) == 0x0100)
-                        strlcat(kctl->id.name, " Capture Source", sizeof(kctl->id.name));
+                        append_ctl_name(kctl, " Capture Source");
                 else
-                        strlcat(kctl->id.name, " Playback Source", sizeof(kctl->id.name));
+                        append_ctl_name(kctl, " Playback Source");
         }
 
         snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",