aboutsummaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-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_proc.c13
-rw-r--r--sound/pci/hda/patch_conexant.c108
-rw-r--r--sound/pci/riptide/riptide.c3
-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
-rw-r--r--sound/usb/mixer.c2
-rw-r--r--sound/usb/mixer_maps.c13
-rw-r--r--sound/usb/mixer_quirks.c67
18 files changed, 183 insertions, 99 deletions
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index b4a6aa960f4..8490f59709b 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 2c79d60a725..536c4c0514d 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 8c63200cf33..bc86cb726d7 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 87f4385fe8c..5ef4fe96436 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 9a9f372e1be..56b4f74c0b1 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_proc.c b/sound/pci/hda/hda_proc.c
index 254ab520460..e59e2f059b6 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 8c6523bbc79..a36488d94aa 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/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 0481d94aac9..cbeb3f77350 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1837,8 +1837,7 @@ static int snd_riptide_free(struct snd_riptide *chip)
}
if (chip->irq >= 0)
free_irq(chip->irq, chip);
- if (chip->fw_entry)
- release_firmware(chip->fw_entry);
+ release_firmware(chip->fw_entry);
release_and_free_resource(chip->res_port);
kfree(chip);
return 0;
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index f8e10ced244..b3e24f28942 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 d1926266fe0..8e92fb88ed0 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 601df809a26..0fe66c3dde1 100644
--- a/sound/soc/imx/imx-audmux.c
+++ b/sound/soc/imx/imx-audmux.c
@@ -79,6 +79,9 @@ 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);
@@ -158,7 +161,7 @@ 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 609abd51e55..d08583790d2 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 a4deebc0801..8d2ebf502df 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1087,6 +1087,8 @@ 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 33509de5254..e53349912b2 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 475428cf270..9ff2c601445 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;
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index ab23869c01b..c374c7242ab 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1388,7 +1388,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *r
for (pin = 0; pin < input_pins; pin++) {
err = parse_audio_unit(state, desc->baSourceID[pin]);
if (err < 0)
- return err;
+ continue;
err = check_input_term(state, desc->baSourceID[pin], &iterm);
if (err < 0)
return err;
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index f1324c42383..41daaa24c25 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -288,6 +288,15 @@ static struct usbmix_name_map scratch_live_map[] = {
{ 0 } /* terminator */
};
+static struct usbmix_name_map ebox44_map[] = {
+ { 4, NULL }, /* FU */
+ { 6, NULL }, /* MU */
+ { 7, NULL }, /* FU */
+ { 10, NULL }, /* FU */
+ { 11, NULL }, /* MU */
+ { 0 }
+};
+
/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+"
* most importand difference is SU[8], it should be set to "Capture Source"
* to make alsamixer and PA working properly.
@@ -371,6 +380,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
.map = scratch_live_map,
.ignore_ctl_error = 1,
},
+ {
+ .id = USB_ID(0x200c, 0x1018),
+ .map = ebox44_map,
+ },
{ 0 } /* terminator */
};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index ab125ee0b0f..e2072edc777 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -557,6 +557,69 @@ static int snd_maudio_ftu_create_mixer(struct usb_mixer_interface *mixer)
return 0;
}
+static int snd_ebox44_create_ctl(struct usb_mixer_interface *mixer,
+ int unitid, int control, int cmask,
+ int val_type, const char *name)
+{
+ struct usb_mixer_elem_info *cval;
+ struct snd_kcontrol *kctl;
+
+ cval = kzalloc(sizeof(*cval), GFP_KERNEL);
+ if (!cval)
+ return -ENOMEM;
+
+ cval->id = unitid;
+ cval->mixer = mixer;
+
+ cval->val_type = val_type;
+ cval->channels = 1;
+ cval->control = control;
+ cval->cmask = cmask;
+
+ /* Volume controls will override these values */
+ cval->min = 0;
+ cval->max = 1;
+ cval->res = 0;
+
+ cval->dBmin = 0;
+ cval->dBmax = 0;
+
+ kctl = snd_ctl_new1(snd_usb_feature_unit_ctl, cval);
+ if (!kctl) {
+ kfree(cval);
+ return -ENOMEM;
+ }
+
+ snprintf(kctl->id.name, sizeof(kctl->id.name), name);
+ kctl->private_free = usb_mixer_elem_free;
+ return snd_usb_mixer_add_control(mixer, kctl);
+}
+
+/*
+ * Create mixer for Electrix Ebox-44
+ *
+ * The mixer units from this device are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we create a good mixer in code.
+ */
+
+static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
+{
+ snd_ebox44_create_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Headphone Playback Switch");
+ snd_ebox44_create_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16, "Headphone A Mix Playback Volume");
+ snd_ebox44_create_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16, "Headphone B Mix Playback Volume");
+
+ snd_ebox44_create_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Output Playback Switch");
+ snd_ebox44_create_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16, "Output A Playback Volume");
+ snd_ebox44_create_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16, "Output B Playback Volume");
+
+ snd_ebox44_create_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN, "Input Capture Switch");
+ snd_ebox44_create_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16, "Input A Capture Volume");
+ snd_ebox44_create_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16, "Input B Capture Volume");
+
+ return 0;
+}
+
void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
unsigned char samplerate_id)
{
@@ -619,6 +682,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
snd_nativeinstruments_ta10_mixers,
ARRAY_SIZE(snd_nativeinstruments_ta10_mixers));
break;
+
+ case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
+ err = snd_ebox44_create_mixer(mixer);
+ break;
}
return err;
generated by cgit v1.2.3-18-g5258 (git 2.32.0) at 2024-12-23 14:07:31 +0000