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-rw-r--r--MAINTAINERS4
-rw-r--r--include/sound/core.h10
-rw-r--r--sound/isa/sscape.c6
-rw-r--r--sound/oss/msnd_pinnacle.c8
-rw-r--r--sound/pci/asihpi/hpi_internal.h4
-rw-r--r--sound/pci/asihpi/hpios.c10
-rw-r--r--sound/pci/hda/hda_codec.h3
-rw-r--r--sound/pci/hda/hda_eld.c6
-rw-r--r--sound/pci/hda/hda_proc.c13
-rw-r--r--sound/pci/hda/patch_conexant.c108
-rw-r--r--sound/pci/hda/patch_hdmi.c9
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c25
-rw-r--r--sound/soc/imx/imx-audmux.c5
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c1
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/tegra/tegra_i2s.c6
-rw-r--r--sound/soc/tegra/tegra_spdif.c4
18 files changed, 120 insertions, 106 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index 2dcfca850639..a1270978eb41 100644
--- 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 b6e0f57d451d..bc056687f647 100644
--- 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 b4a6aa960f4b..8490f59709bb 100644
--- 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 2c79d60a725f..536c4c0514d3 100644
--- 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 8c63200cf339..bc86cb726d79 100644
--- 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 87f4385fe8c7..5ef4fe964366 100644
--- 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 9a9f372e1be4..56b4f74c0b13 100644
--- 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 b58b4b1687fa..4c054f4486b9 100644
--- 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 254ab5204603..e59e2f059b6e 100644
--- 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 8c6523bbc797..a36488d94aaa 100644
--- 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 540cd13f7f15..83f345f3c961 100644
--- 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 f8e10ced244a..b3e24f289421 100644
--- 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 d1926266fe00..8e92fb88ed09 100644
--- 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 1765a197acb0..f23700359c67 100644
--- 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 609abd51e55f..d08583790d23 100644
--- 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 e19c24ade414..accdcb7d4d9d 100644
--- 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 33509de52540..e53349912b2e 100644
--- 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 475428cf270e..9ff2c601445f 100644
--- 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;
}
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-26 15:54:53 +0300