Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 - A series of fixes for Conexant 20549 HD-audio codec chip
 - A workaround for HDMI hotplug debug prints that annoyed people
 - A fix for the new support of platform DAPM contexts
 - Many driver-specific minor fixes

* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - hide HDMI/ELD printks unless snd.debug=2
  ALSA: sound/isa/sscape.c: add missing resource-release code
  sound: sound/oss/msnd_pinnacle.c: add vfrees
  ALSA: hda - clean up CX20549 test mixer setup
  ALSA: hda - CX20549 doesn't need pin_amp_workaround.
  ALSA: hda - Remove CD control from model=benq for CX20549
  ALSA: hda - fix record volume controls of CX20459 ("Venice")
  ALSA: hda - Rename capture sources of CX20549 to match common conventions
  ALSA: hda - Fix proc output for ADC amp values of CX20549
  ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
  ASoC: set idle_bias_off=1 for all platform DAPM contexts
  ASoC: imx-audmux: Check for NULL pointer
  ASoC: imx-audmux: Fix ssi port numbers in sysfs
  ASoC: ak4642: fixup: mute needs +1 step
  MAINTAINERS: Don't list everyone working on Wolfson drivers
  MAINTAINERS: Add missing ASoC OMAP co-maintainer
  ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
  ASoC: tegra: ensure clocks are enabled when touching registers
  ASoC: sgtl5000: Enable VAG when DAC/ADC up
  ALSA: asihpi - fix return value of hpios_locked_mem_alloc()

diff --git a/MAINTAINERS b/MAINTAINERS
index 2dcfca8..a127097 100644 (file)
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -4803,6 +4803,7 @@ F:        arch/arm/mach-omap2/clockdomain2xxx_3xxx.c
 F:     arch/arm/mach-omap2/clockdomain44xx.c
 
 OMAP AUDIO SUPPORT
+M:     Peter Ujfalusi <peter.ujfalusi@ti.com>
 M:     Jarkko Nikula <jarkko.nikula@bitmer.com>
 L:     alsa-devel@alsa-project.org (subscribers-only)
 L:     linux-omap@vger.kernel.org
@@ -7461,8 +7462,7 @@ F:        include/linux/wm97xx.h
 
 WOLFSON MICROELECTRONICS DRIVERS
 M:     Mark Brown <broonie@opensource.wolfsonmicro.com>
-M:     Ian Lartey <ian@opensource.wolfsonmicro.com>
-M:     Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+L:     patches@opensource.wolfsonmicro.com
 T:     git git://opensource.wolfsonmicro.com/linux-2.6-asoc
 T:     git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
 W:     http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices

diff --git a/include/sound/core.h b/include/sound/core.h
index b6e0f57..bc05668 100644 (file)
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -325,6 +325,13 @@ void release_and_free_resource(struct resource *res);
 
 /* --- */
 
+/* sound printk debug levels */
+enum {
+       SND_PR_ALWAYS,
+       SND_PR_DEBUG,
+       SND_PR_VERBOSE,
+};
+
 #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
 __printf(4, 5)
 void __snd_printk(unsigned int level, const char *file, int line,
@@ -354,6 +361,8 @@ void __snd_printk(unsigned int level, const char *file, int line,
  */
 #define snd_printd(fmt, args...) \
        __snd_printk(1, __FILE__, __LINE__, fmt, ##args)
+#define _snd_printd(level, fmt, args...) \
+       __snd_printk(level, __FILE__, __LINE__, fmt, ##args)
 
 /**
  * snd_BUG - give a BUG warning message and stack trace
@@ -383,6 +392,7 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #else /* !CONFIG_SND_DEBUG */
 
 #define snd_printd(fmt, args...)       do { } while (0)
+#define _snd_printd(level, fmt, args...) do { } while (0)
 #define snd_BUG()                      do { } while (0)
 static inline int __snd_bug_on(int cond)
 {

diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index b4a6aa9..8490f59 100644 (file)
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
        irq_cfg = get_irq_config(sscape->type, irq[dev]);
        if (irq_cfg == INVALID_IRQ) {
                snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]);
-               return -ENXIO;
+               err = -ENXIO;
+               goto _release_dma;
        }
 
        mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]);
        if (mpu_irq_cfg == INVALID_IRQ) {
                snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
-               return -ENXIO;
+               err = -ENXIO;
+               goto _release_dma;
        }
 
        /*

diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c
index 2c79d60..536c4c0 100644 (file)
--- a/sound/oss/msnd_pinnacle.c
+++ b/sound/oss/msnd_pinnacle.c
@@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate)
 
 static int upload_dsp_code(void)
 {
+       int ret = 0;
+
        msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS);
 #ifndef HAVE_DSPCODEH
        INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE);
@@ -1312,7 +1314,8 @@ static int upload_dsp_code(void)
        memcpy_toio(dev.base, PERMCODE, PERMCODESIZE);
        if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) {
                printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n");
-               return -ENODEV;
+               ret = -ENODEV;
+               goto out;
        }
 #ifdef HAVE_DSPCODEH
        printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n");
@@ -1320,12 +1323,13 @@ static int upload_dsp_code(void)
        printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n");
 #endif
 
+out:
 #ifndef HAVE_DSPCODEH
        vfree(INITCODE);
        vfree(PERMCODE);
 #endif
 
-       return 0;
+       return ret;
 }
 
 #ifdef MSND_CLASSIC

diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index 8c63200..bc86cb7 100644 (file)
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned.
 If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and
 HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle.
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
+u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle,
                                                           /**< memory handle */
        u32 size, /**< Size in bytes to allocate */
        struct pci_dev *p_os_reference

diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index 87f4385..5ef4fe9 100644 (file)
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -1,7 +1,7 @@
 /******************************************************************************
 
     AudioScience HPI driver
-    Copyright (C) 1997-2011  AudioScience Inc. <support@audioscience.com>
+    Copyright (C) 1997-2012  AudioScience Inc. <support@audioscience.com>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of version 2 of the GNU General Public License as
@@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec)
 
 }
 
-/** Allocated an area of locked memory for bus master DMA operations.
+/** Allocate an area of locked memory for bus master DMA operations.
 
-On error, return -ENOMEM, and *pMemArea.size = 0
+If allocation fails, return 1, and *pMemArea.size = 0
 */
-int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
+u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
        struct pci_dev *pdev)
 {
        /*?? any benefit in using managed dmam_alloc_coherent? */
@@ -62,7 +62,7 @@ int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size,
                HPI_DEBUG_LOG(WARNING,
                        "failed to allocate %d bytes locked memory\n", size);
                p_mem_area->size = 0;
-               return -ENOMEM;
+               return 1;
        }
 }
 

diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 9a9f372..56b4f74 100644 (file)
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -851,6 +851,9 @@ struct hda_codec {
        unsigned int pin_amp_workaround:1; /* pin out-amp takes index
                                            * (e.g. Conexant codecs)
                                            */
+       unsigned int single_adc_amp:1; /* adc in-amp takes no index
+                                       * (e.g. CX20549 codec)
+                                       */
        unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */
        unsigned int pins_shutup:1;     /* pins are shut up */
        unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index b58b4b1..4c054f4 100644 (file)
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a)
        else
                buf2[0] = '\0';
 
-       printk(KERN_INFO "HDMI: supports coding type %s:"
+       _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:"
                        " channels = %d, rates =%s%s\n",
                        cea_audio_coding_type_names[a->format],
                        a->channels,
@@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e)
 {
        int i;
 
-       printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n",
+       _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n",
                        e->monitor_name,
                        eld_connection_type_names[e->conn_type]);
 
        if (e->spk_alloc) {
                char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE];
                snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf));
-               printk(KERN_INFO "HDMI: available speakers:%s\n", buf);
+               _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf);
        }
 
        for (i = 0; i < e->sad_count; i++)

diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 254ab52..e59e2f0 100644 (file)
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -651,9 +651,16 @@ static void print_codec_info(struct snd_info_entry *entry,
                        snd_iprintf(buffer, "  Amp-In caps: ");
                        print_amp_caps(buffer, codec, nid, HDA_INPUT);
                        snd_iprintf(buffer, "  Amp-In vals: ");
-                       print_amp_vals(buffer, codec, nid, HDA_INPUT,
-                                      wid_caps & AC_WCAP_STEREO,
-                                      wid_type == AC_WID_PIN ? 1 : conn_len);
+                       if (wid_type == AC_WID_PIN ||
+                           (codec->single_adc_amp &&
+                            wid_type == AC_WID_AUD_IN))
+                               print_amp_vals(buffer, codec, nid, HDA_INPUT,
+                                              wid_caps & AC_WCAP_STEREO,
+                                              1);
+                       else
+                               print_amp_vals(buffer, codec, nid, HDA_INPUT,
+                                              wid_caps & AC_WCAP_STEREO,
+                                              conn_len);
                }
                if (wid_caps & AC_WCAP_OUT_AMP) {
                        snd_iprintf(buffer, "  Amp-Out caps: ");

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8c6523b..a36488d 100644 (file)
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -141,7 +141,6 @@ struct conexant_spec {
        unsigned int hp_laptop:1;
        unsigned int asus:1;
        unsigned int pin_eapd_ctrls:1;
-       unsigned int single_adc_amp:1;
 
        unsigned int adc_switching:1;
 
@@ -687,27 +686,26 @@ static const struct hda_channel_mode cxt5045_modes[1] = {
 static const struct hda_input_mux cxt5045_capture_source = {
        .num_items = 2,
        .items = {
-               { "IntMic", 0x1 },
-               { "ExtMic", 0x2 },
+               { "Internal Mic", 0x1 },
+               { "Mic",          0x2 },
        }
 };
 
 static const struct hda_input_mux cxt5045_capture_source_benq = {
-       .num_items = 5,
+       .num_items = 4,
        .items = {
-               { "IntMic", 0x1 },
-               { "ExtMic", 0x2 },
-               { "LineIn", 0x3 },
-               { "CD",     0x4 },
-               { "Mixer",  0x0 },
+               { "Internal Mic", 0x1 },
+               { "Mic",          0x2 },
+               { "Line",         0x3 },
+               { "Mixer",        0x0 },
        }
 };
 
 static const struct hda_input_mux cxt5045_capture_source_hp530 = {
        .num_items = 2,
        .items = {
-               { "ExtMic", 0x1 },
-               { "IntMic", 0x2 },
+               { "Mic",          0x1 },
+               { "Internal Mic", 0x2 },
        }
 };
 
@@ -798,10 +796,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static const struct snd_kcontrol_new cxt5045_mixers[] = {
-       HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
-       HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
-       HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
        HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
@@ -822,27 +818,15 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = {
 };
 
 static const 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),
+       HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT),
+       HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT),
 
        {}
 };
 
 static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-       HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
-       HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
-       HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
        HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
@@ -946,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
        /* Output controls */
        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT),
        HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-       HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-       HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-       HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-       HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+       HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+       HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT),
        
        /* Modes for retasking pin widgets */
        CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT),
@@ -960,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
 
        /* Loopback mixer controls */
 
-       HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT),
-       HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT),
-       HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT),
-       HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT),
-       HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT),
-       HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT),
+       HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT),
+       HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT),
+       HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT),
+       HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT),
+       HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT),
+       HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT),
+       HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT),
+       HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT),
        {
                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
                .name = "Input Source",
@@ -978,16 +962,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = {
                .put = conexant_mux_enum_put,
        },
        /* Audio input controls */
-       HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT),
-       HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT),
-       HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT),
-       HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT),
-       HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT),
-       HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT),
-       HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT),
-       HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT),
-       HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
        { } /* end */
 };
 
@@ -1009,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
        {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
        {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0},
 
-       /* Start with output sum widgets muted and their output gains at min */
-       {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
        /* Unmute retasking pin widget output buffers since the default
         * state appears to be output.  As the pin mode is changed by the
         * user the pin mode control will take care of enabling the pin's
@@ -1027,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = {
        /* Set ADC connection select to match default mixer setting (mic1
         * pin)
         */
-       {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
-       {0x17, AC_VERB_SET_CONNECT_SEL, 0x00},
+       {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+       {0x17, AC_VERB_SET_CONNECT_SEL, 0x01},
 
        /* Mute all inputs to mixer widget (even unconnected ones) */
-       {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */
+       {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */
        {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */
        {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */
        {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */
@@ -1110,7 +1082,7 @@ static int patch_cxt5045(struct hda_codec *codec)
        if (!spec)
                return -ENOMEM;
        codec->spec = spec;
-       codec->pin_amp_workaround = 1;
+       codec->single_adc_amp = 1;
 
        spec->multiout.max_channels = 2;
        spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -4220,7 +4192,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid,
                int idx = get_input_connection(codec, adc_nid, nid);
                if (idx < 0)
                        continue;
-               if (spec->single_adc_amp)
+               if (codec->single_adc_amp)
                        idx = 0;
                return cx_auto_add_volume_idx(codec, label, pfx,
                                              cidx, adc_nid, HDA_INPUT, idx);
@@ -4275,7 +4247,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
                if (cidx < 0)
                        continue;
                input_conn[i] = spec->imux_info[i].adc;
-               if (!spec->single_adc_amp)
+               if (!codec->single_adc_amp)
                        input_conn[i] |= cidx << 8;
                if (i > 0 && input_conn[i] != input_conn[0])
                        multi_connection = 1;
@@ -4466,15 +4438,17 @@ static int patch_conexant_auto(struct hda_codec *codec)
        if (!spec)
                return -ENOMEM;
        codec->spec = spec;
-       codec->pin_amp_workaround = 1;
 
        switch (codec->vendor_id) {
        case 0x14f15045:
-               spec->single_adc_amp = 1;
+               codec->single_adc_amp = 1;
                break;
        case 0x14f15051:
                add_cx5051_fake_mutes(codec);
+               codec->pin_amp_workaround = 1;
                break;
+       default:
+               codec->pin_amp_workaround = 1;
        }
 
        apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 540cd13..83f345f 100644 (file)
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
        struct hdmi_spec *spec = codec->spec;
        int tag = res >> AC_UNSOL_RES_TAG_SHIFT;
        int pin_nid;
-       int pd = !!(res & AC_UNSOL_RES_PD);
-       int eldv = !!(res & AC_UNSOL_RES_ELDV);
        int pin_idx;
        struct hda_jack_tbl *jack;
 
@@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
        pin_nid = jack->nid;
        jack->jack_dirty = 1;
 
-       printk(KERN_INFO
+       _snd_printd(SND_PR_VERBOSE,
                "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-               codec->addr, pin_nid, pd, eldv);
+               codec->addr, pin_nid,
+               !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
        pin_idx = pin_nid_to_pin_index(spec, pin_nid);
        if (pin_idx < 0)
@@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll)
        if (eld->monitor_present)
                eld_valid       = !!(present & AC_PINSENSE_ELDV);
 
-       printk(KERN_INFO
+       _snd_printd(SND_PR_VERBOSE,
                "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
                codec->addr, pin_nid, eld->monitor_present, eld_valid);
 

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f8e10ce..b3e24f2 100644 (file)
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -140,7 +140,7 @@
  * min : 0xFE : -115.0 dB
  * mute: 0xFF
  */
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
 
 static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d192626..8e92fb8 100644 (file)
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 }
 
 /*
- * using codec assist to small pop, hp_powerup or lineout_powerup
- * should stay setting until vag_powerup is fully ramped down,
- * vag fully ramped down require 400ms.
+ * As manual described, ADC/DAC only works when VAG powerup,
+ * So enabled VAG before ADC/DAC up.
+ * In power down case, we need wait 400ms when vag fully ramped down.
  */
-static int small_pop_event(struct snd_soc_dapm_widget *w,
+static int power_vag_event(struct snd_soc_dapm_widget *w,
        struct snd_kcontrol *kcontrol, int event)
 {
        switch (event) {
@@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w,
                        SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
                break;
 
-       case SND_SOC_DAPM_PRE_PMD:
+       case SND_SOC_DAPM_POST_PMD:
                snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
                        SGTL5000_VAG_POWERUP, 0);
                msleep(400);
@@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
                                mic_bias_event,
                                SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
-       SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
-                       small_pop_event,
-                       SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
-       SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
-                       small_pop_event,
-                       SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+       SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0),
+       SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0),
 
        SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
        SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
@@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
                                0, SGTL5000_CHIP_DIG_POWER,
                                1, 0),
 
-       SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+       SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0,
+                           power_vag_event,
+                           SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
+       SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
        SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
 };
 
@@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
        {"Capture Mux", "LINE_IN", "LINE_IN"},  /* line_in --> adc_mux */
        {"Capture Mux", "MIC_IN", "MIC_IN"},    /* mic_in --> adc_mux */
 
+       {"ADC", NULL, "VAG_POWER"},
        {"ADC", NULL, "Capture Mux"},           /* adc_mux --> adc */
        {"AIFOUT", NULL, "ADC"},                /* adc --> i2s_out */
 
+       {"DAC", NULL, "VAG_POWER"},
        {"DAC", NULL, "AIFIN"},                 /* i2s-->dac,skip audio mux */
        {"Headphone Mux", "DAC", "DAC"},        /* dac --> hp_mux */
        {"LO", NULL, "DAC"},                    /* dac --> line_out */

diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c
index 1765a19..f237003 100644 (file)
--- a/sound/soc/imx/imx-audmux.c
+++ b/sound/soc/imx/imx-audmux.c
@@ -73,6 +73,9 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
        if (!buf)
                return -ENOMEM;
 
+       if (!audmux_base)
+               return -ENOSYS;
+
        if (audmux_clk)
                clk_prepare_enable(audmux_clk);
 
@@ -152,7 +155,7 @@ static void __init audmux_debugfs_init(void)
                return;
        }
 
-       for (i = 1; i < 8; i++) {
+       for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
                snprintf(buf, sizeof(buf), "ssi%d", i);
                if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
                                         (void *)i, &audmux_debugfs_fops))

diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 609abd5..d085837 100644 (file)
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -17,6 +17,7 @@
 #include <linux/delay.h>
 #include <linux/clk.h>
 #include <linux/platform_device.h>
+#include <linux/io.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e19c24a..accdcb7 100644 (file)
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1081,6 +1081,8 @@ static int soc_probe_platform(struct snd_soc_card *card,
                snd_soc_dapm_new_controls(&platform->dapm,
                        driver->dapm_widgets, driver->num_dapm_widgets);
 
+       platform->dapm.idle_bias_off = 1;
+
        if (driver->probe) {
                ret = driver->probe(platform);
                if (ret < 0) {

diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 33509de..e533499 100644 (file)
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused)
        struct tegra_i2s *i2s = s->private;
        int i;
 
+       clk_enable(i2s->clk_i2s);
+
        for (i = 0; i < ARRAY_SIZE(regs); i++) {
                u32 val = tegra_i2s_read(i2s, regs[i].offset);
                seq_printf(s, "%s = %08x\n", regs[i].name, val);
        }
 
+       clk_disable(i2s->clk_i2s);
+
        return 0;
 }
 
@@ -112,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
                debugfs_remove(i2s->debug);
 }
 #else
-static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
+static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
 {
 }
 

diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c
index 475428c..9ff2c60 100644 (file)
--- a/sound/soc/tegra/tegra_spdif.c
+++ b/sound/soc/tegra/tegra_spdif.c
@@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused)
        struct tegra_spdif *spdif = s->private;
        int i;
 
+       clk_enable(spdif->clk_spdif_out);
+
        for (i = 0; i < ARRAY_SIZE(regs); i++) {
                u32 val = tegra_spdif_read(spdif, regs[i].offset);
                seq_printf(s, "%s = %08x\n", regs[i].name, val);
        }
 
+       clk_disable(spdif->clk_spdif_out);
+
        return 0;
 }