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Message-ID: <s5hk4a0y3qc.wl%tiwai@suse.de>
Date: Fri, 26 Aug 2011 10:30:19 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Mark Brown <broonie@...nsource.wolfsonmicro.com>,
Liam Girdwood <lrg@...mlogic.co.uk>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 3.1-rc4
Linus,
please pull sound fixes for v3.1-rc4 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
Most of commits are small and trivial fixes for ASoC, and a few other
HD-audio regression fixes.
Thanks!
Takashi
===
Axel Lin (3):
ASoC: soc-jack: Fix checking return value of request_any_context_irq
ASoC: sta32x: Fix a memory leak if snd_soc_register_codec fails
ASoC: soc-core: use GFP_KERNEL flag for kmalloc in snd_soc_cnew
David Henningsson (1):
ALSA: hda: Conexant: Allow different output types to share DAC
Jarkko Nikula (1):
ASoC: omap: Fix build errors in ams-delta
Joseph Pentland (1):
ASoC: Add Springbank I/O card to Speyside Kconfig
Julia Lawall (4):
sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
sound/soc/ep93xx/ep93xx-i2s.c: add missing kfree
sound/soc/fsl/p1022_ds.c: add missing of_node_put
sound/soc/fsl/mpc8610_hpcd.c: add missing of_node_put
Mark Brown (7):
ASoC: Move WM8962 CLKREG_OVD earlier
ASoC: WM8996 record paths need AIFCLK
ASoC: Fix configuration of WM8996 input enables
ASoC: Ensure we only run Speyside WM8962 bias level callbacks once
ASoC: Clear any outstanding WM8962 FLL lock completions before waiting
ASoC: Clear completions from late WM8996 FLL lock IRQs
ASoC: Correct element count for WM8996 sidetone HPF
Sangbeom Kim (1):
ASoC: Add samsung maintainer
Sascha Hauer (1):
ASoC: Fix check for symmetric rate enforcement
Scott Jiang (5):
ASoC: ad193x: fix registers definition
ASoC: ad193x: fix dac word len setting
ASoC: ad193x: fix system clock
ASoC: Add spi hw read function for 16 addr 8 data mode for ad193x fix
ASoC: ad193x: remove cache support
Stephen Warren (1):
ASoC: Tegra: wm8903 machine driver: Drop Ventana support
Takashi Iwai (4):
ALSA: hda - Fix output-path initialization for Realtek auto-parser
ALSA: hda - Update jack-sense info even when no automute is set
ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser
ALSA: hda/conexant - Enable ADC-switching for auto-mic mode, too
Timur Tabi (1):
sound/soc/fsl/fsl_dma.c: add missing of_node_put
Vasily Khoruzhick (2):
ASoC: h1940: Fix compilation error due to missing header
ASoC: rx1950: Fix compilation error due to missing header
---
MAINTAINERS | 1 +
sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++-------------
sound/pci/hda/patch_realtek.c | 28 ++++++++++-------
sound/soc/blackfin/bf5xx-ad193x.c | 2 +-
sound/soc/codecs/ad193x.c | 11 +-----
sound/soc/codecs/ad193x.h | 5 ++-
sound/soc/codecs/sta32x.c | 1 +
sound/soc/codecs/wm8962.c | 12 +++++--
sound/soc/codecs/wm8996.c | 28 +++++++++++------
sound/soc/ep93xx/ep93xx-i2s.c | 5 ++-
sound/soc/fsl/fsl_dma.c | 2 +
sound/soc/fsl/mpc8610_hpcd.c | 18 +++++-----
sound/soc/fsl/p1022_ds.c | 4 ++-
sound/soc/kirkwood/kirkwood-i2s.c | 2 +-
sound/soc/omap/ams-delta.c | 6 ++-
sound/soc/samsung/Kconfig | 1 +
sound/soc/samsung/h1940_uda1380.c | 1 +
sound/soc/samsung/rx1950_uda1380.c | 1 +
sound/soc/samsung/speyside_wm8962.c | 6 ++++
sound/soc/soc-core.c | 2 +-
sound/soc/soc-io.c | 23 ++++++++++++++
sound/soc/soc-jack.c | 2 +-
sound/soc/soc-pcm.c | 3 ++
sound/soc/tegra/tegra_wm8903.c | 4 +-
24 files changed, 148 insertions(+), 77 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index d942920..1a8cc60 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5532,6 +5532,7 @@ F: include/media/*7146*
SAMSUNG AUDIO (ASoC) DRIVERS
M: Jassi Brar <jassisinghbrar@...il.com>
+M: Sangbeom Kim <sbkim73@...sung.com>
L: alsa-devel@...a-project.org (moderated for non-subscribers)
S: Supported
F: sound/soc/samsung
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 502fc94..7696d05 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin,
#define MAX_AUTO_DACS 5
+#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */
+
/* fill analog DAC list from the widget tree */
static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
{
@@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs)
/* fill pin_dac_pair list from the pin and dac list */
static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins,
int num_pins, hda_nid_t *dacs, int *rest,
- struct pin_dac_pair *filled, int type)
+ struct pin_dac_pair *filled, int nums,
+ int type)
{
- int i, nums;
+ int i, start = nums;
- nums = 0;
- for (i = 0; i < num_pins; i++) {
+ for (i = 0; i < num_pins; i++, nums++) {
filled[nums].pin = pins[i];
filled[nums].type = type;
filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest);
- nums++;
+ if (filled[nums].dac)
+ continue;
+ if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) {
+ filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) {
+ filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG;
+ continue;
+ }
+ snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]);
}
return nums;
}
@@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec)
rest = fill_cx_auto_dacs(codec, dacs);
/* parse all analog output pins */
nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs,
- dacs, &rest, spec->dac_info,
- AUTO_PIN_LINE_OUT);
- nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_HP_OUT);
- nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
- dacs, &rest, spec->dac_info + nums,
- AUTO_PIN_SPEAKER_OUT);
+ dacs, &rest, spec->dac_info, 0,
+ AUTO_PIN_LINE_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_HP_OUT);
+ nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs,
+ dacs, &rest, spec->dac_info, nums,
+ AUTO_PIN_SPEAKER_OUT);
spec->dac_info_filled = nums;
/* fill multiout struct */
for (i = 0; i < nums; i++) {
hda_nid_t dac = spec->dac_info[i].dac;
- if (!dac)
+ if (!dac || (dac & DAC_SLAVE_FLAG))
continue;
switch (spec->dac_info[i].type) {
case AUTO_PIN_LINE_OUT:
@@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec)
}
if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items)
cx_auto_check_auto_mic(codec);
- if (imux->num_items > 1 && !spec->auto_mic) {
+ if (imux->num_items > 1) {
for (i = 1; i < imux->num_items; i++) {
if (spec->imux_info[i].adc != spec->imux_info[0].adc) {
spec->adc_switching = 1;
@@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec)
nid = spec->dac_info[i].dac;
if (!nid)
nid = spec->multiout.dac_nids[0];
+ else if (nid & DAC_SLAVE_FLAG)
+ nid &= ~DAC_SLAVE_FLAG;
select_connection(codec, spec->dac_info[i].pin, nid);
}
if (spec->auto_mute) {
@@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac,
hda_nid_t pin, const char *name, int idx)
{
unsigned int caps;
- caps = query_amp_caps(codec, dac, HDA_OUTPUT);
- if (caps & AC_AMPCAP_NUM_STEPS)
- return cx_auto_add_pb_volume(codec, dac, name, idx);
+ if (dac && !(dac & DAC_SLAVE_FLAG)) {
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ if (caps & AC_AMPCAP_NUM_STEPS)
+ return cx_auto_add_pb_volume(codec, dac, name, idx);
+ }
caps = query_amp_caps(codec, pin, HDA_OUTPUT);
if (caps & AC_AMPCAP_NUM_STEPS)
return cx_auto_add_pb_volume(codec, pin, name, idx);
@@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
for (i = 0; i < spec->dac_info_filled; i++) {
const char *label;
int idx, type;
- if (!spec->dac_info[i].dac)
- continue;
+ hda_nid_t dac = spec->dac_info[i].dac;
type = spec->dac_info[i].type;
if (type == AUTO_PIN_LINE_OUT)
type = spec->autocfg.line_out_type;
@@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec)
idx = num_spk++;
break;
}
- err = try_add_pb_volume(codec, spec->dac_info[i].dac,
+ err = try_add_pb_volume(codec, dac,
spec->dac_info[i].pin,
label, idx);
if (err < 0)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index fcb11af..7cabd73 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute)
- return;
spec->jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
spec->autocfg.hp_pins);
+ if (!spec->automute)
+ return;
update_speakers(codec);
}
@@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (!spec->automute || !spec->detect_line)
- return;
spec->line_jack_present =
detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
spec->autocfg.line_out_pins);
+ if (!spec->automute || !spec->detect_line)
+ return;
update_speakers(codec);
}
@@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec)
static void alc_auto_init_extra_out(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- hda_nid_t pin;
+ hda_nid_t pin, dac;
pin = spec->autocfg.hp_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_HP,
- spec->multiout.hp_nid);
+ if (pin) {
+ dac = spec->multiout.hp_nid;
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac);
+ }
pin = spec->autocfg.speaker_pins[0];
- if (pin)
- alc_auto_set_output_and_unmute(codec, pin, PIN_OUT,
- spec->multiout.extra_out_nid[0]);
+ if (pin) {
+ dac = spec->multiout.extra_out_nid[0];
+ if (!dac)
+ dac = spec->multiout.dac_nids[0];
+ alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac);
+ }
}
/*
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index d6651c0..a118a0f 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
switch (params_rate(params)) {
case 48000:
- clk = 12288000;
+ clk = 24576000;
break;
}
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 2374ca5..eedb6f5 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -27,11 +27,6 @@ struct ad193x_priv {
int sysclk;
};
-/* ad193x register cache & default register settings */
-static const u8 ad193x_reg[AD193X_NUM_REGS] = {
- 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
-};
-
/*
* AD193X volume/mute/de-emphasis etc. controls
*/
@@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
- reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
+ reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+ | (word_len << AD193X_DAC_WORD_LEN_SHFT);
snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
@@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
.probe = ad193x_probe,
- .reg_cache_default = ad193x_reg,
- .reg_cache_size = AD193X_NUM_REGS,
- .reg_word_size = sizeof(u16),
};
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 9747b54..cccc2e8 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -34,7 +34,8 @@
#define AD193X_DAC_LEFT_HIGH (1 << 3)
#define AD193X_DAC_BCLK_INV (1 << 7)
#define AD193X_DAC_CTRL2 0x804
-#define AD193X_DAC_WORD_LEN_MASK 0xC
+#define AD193X_DAC_WORD_LEN_SHFT 3
+#define AD193X_DAC_WORD_LEN_MASK 0x18
#define AD193X_DAC_MASTER_MUTE 1
#define AD193X_DAC_CHNL_MUTE 0x805
#define AD193X_DACL1_MUTE 0
@@ -63,7 +64,7 @@
#define AD193X_ADC_CTRL1 0x80f
#define AD193X_ADC_SERFMT_MASK 0x60
#define AD193X_ADC_SERFMT_STEREO (0 << 5)
-#define AD193X_ADC_SERFMT_TDM (1 << 2)
+#define AD193X_ADC_SERFMT_TDM (1 << 5)
#define AD193X_ADC_SERFMT_AUX (2 << 5)
#define AD193X_ADC_WORD_LEN_MASK 0x3
#define AD193X_ADC_CTRL2 0x810
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 409d89d..fbd7eb9 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret);
+ kfree(sta32x);
return ret;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 60d740e..1725550 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (fll) {
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_ENA, WM8962_FLL_ENA);
if (wm8962->irq) {
@@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec,
WM8962_BIAS_ENA | 0x180);
msleep(5);
-
- snd_soc_update_bits(codec, WM8962_CLOCKING2,
- WM8962_CLKREG_OVD,
- WM8962_CLKREG_OVD);
}
/* VMID 2*250k */
@@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
+ try_wait_for_completion(&wm8962->fll_lock);
+
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
WM8962_FLL_ENA, fll1);
@@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec)
*/
snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0);
+ /* Ensure we have soft control over all registers */
+ snd_soc_update_bits(codec, WM8962_CLOCKING2,
+ WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
+
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
if (pdata) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index ab8e9d1..0cdb9d1 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = {
};
static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text);
+ SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
@@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0),
-SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux),
-SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux),
-
-SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0),
+SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux),
+SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux),
+SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux),
SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0),
@@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
{ "AIF2RX0", NULL, "AIFCLK" },
{ "AIF2RX1", NULL, "AIFCLK" },
+ { "AIF1TX0", NULL, "AIFCLK" },
+ { "AIF1TX1", NULL, "AIFCLK" },
+ { "AIF1TX2", NULL, "AIFCLK" },
+ { "AIF1TX3", NULL, "AIFCLK" },
+ { "AIF1TX4", NULL, "AIFCLK" },
+ { "AIF1TX5", NULL, "AIFCLK" },
+
+ { "AIF2TX0", NULL, "AIFCLK" },
+ { "AIF2TX1", NULL, "AIFCLK" },
+
{ "DSP1RXL", NULL, "SYSDSPCLK" },
{ "DSP1RXR", NULL, "SYSDSPCLK" },
{ "DSP2RXL", NULL, "SYSDSPCLK" },
@@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
+ /* Clear any pending completions (eg, from failed startups) */
+ try_wait_for_completion(&wm8996->fll_lock);
+
snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
WM8996_FLL_ENA, WM8996_FLL_ENA);
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index 56efa0c..099614e 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
if (!res) {
err = -ENODEV;
- goto fail;
+ goto fail_free_info;
}
info->mem = request_mem_region(res->start, resource_size(res),
pdev->name);
if (!info->mem) {
err = -EBUSY;
- goto fail;
+ goto fail_free_info;
}
info->regs = ioremap(info->mem->start, resource_size(info->mem));
@@ -435,6 +435,7 @@ fail_unmap_mem:
iounmap(info->regs);
fail_release_mem:
release_mem_region(info->mem->start, resource_size(info->mem));
+fail_free_info:
kfree(info);
fail:
return err;
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 732208c..cb50598 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np)
* assume that device_node pointers are a valid comparison.
*/
np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0);
+ of_node_put(np);
if (np == dma_channel_np)
return ssi_np;
}
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a192979..358f0ba 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL);
- if (!machine_data)
- return -ENOMEM;
+ if (!machine_data) {
+ ret = -ENOMEM;
+ goto error_alloc;
+ }
machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
machine_data->dai[0].ops = &mpc8610_hpcd_ops;
@@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
ret = platform_device_add(sound_device);
if (ret) {
dev_err(&pdev->dev, "platform device add failed\n");
- goto error;
+ goto error_sound;
}
dev_set_drvdata(&pdev->dev, sound_device);
@@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
return 0;
+error_sound:
+ platform_device_unregister(sound_device);
error:
- of_node_put(codec_np);
-
- if (sound_device)
- platform_device_unregister(sound_device);
-
kfree(machine_data);
-
+error_alloc:
+ of_node_put(codec_np);
return ret;
}
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 8fa4d5f..fcb862e 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np,
* dai->platform name should already point to an allocated buffer.
*/
ret = of_address_to_resource(dma_channel_np, 0, &res);
- if (ret)
+ if (ret) {
+ of_node_put(dma_channel_np);
return ret;
+ }
snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
(unsigned long long) res.start, dma_channel_np->name);
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index a33fc51..8f16cd3 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
if (!priv->mem) {
dev_err(&pdev->dev, "request_mem_region failed\n");
err = -EBUSY;
- goto error;
+ goto error_alloc;
}
priv->io = ioremap(priv->mem->start, SZ_16K);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 30fe0d0..0aa475f 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
}
/* Set codec bias level */
- ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY);
/* Add hook switch - can be used to control the codec from userspace
* even if line discipline fails */
@@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void)
ams_delta_hook_switch_gpios);
/* Keep modem power on */
- ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+ ams_delta_set_bias_level(&ams_delta_audio_card,
+ &ams_delta_audio_card.rtd[0].codec->dapm,
+ SND_SOC_BIAS_STANDBY);
platform_device_unregister(cx20442_platform_device);
platform_device_unregister(ams_delta_audio_platform_device);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index b99091f..65f980e 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE
select SND_SAMSUNG_I2S
select SND_SOC_WM8996
select SND_SOC_WM9081
+ select SND_SOC_WM1250_EV1
config SND_SOC_SPEYSIDE_WM8962
tristate "Audio support for Wolfson Speyside with WM8962"
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index 241f55d..c6c6589 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -13,6 +13,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 1e574a5..bc8c167 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -17,6 +17,7 @@
*
*/
+#include <linux/types.h>
#include <linux/gpio.h>
#include <sound/soc.h>
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 0b9eb5f..72535f2 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
@@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
int ret;
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
switch (level) {
case SND_SOC_BIAS_STANDBY:
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 83ad8ca..b085d8e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
if (prefix) {
name_len = strlen(long_name) + strlen(prefix) + 2;
- name = kmalloc(name_len, GFP_ATOMIC);
+ name = kmalloc(name_len, GFP_KERNEL);
if (!name)
return NULL;
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index cca490c..a62f7dd 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
#define snd_soc_16_8_read_i2c NULL
#endif
+#if defined(CONFIG_SPI_MASTER)
+static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
+ unsigned int r)
+{
+ struct spi_device *spi = codec->control_data;
+
+ const u16 reg = cpu_to_be16(r | 0x100);
+ u8 data;
+ int ret;
+
+ ret = spi_write_then_read(spi, ®, 2, &data, 1);
+ if (ret < 0)
+ return 0;
+ return data;
+}
+#else
+#define snd_soc_16_8_read_spi NULL
+#endif
+
static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
@@ -295,6 +314,7 @@ static struct {
int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
+ unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
} io_types[] = {
{
.addr_bits = 4, .data_bits = 12,
@@ -318,6 +338,7 @@ static struct {
.addr_bits = 16, .data_bits = 8,
.write = snd_soc_16_8_write,
.i2c_read = snd_soc_16_8_read_i2c,
+ .spi_read = snd_soc_16_8_read_spi,
},
{
.addr_bits = 16, .data_bits = 16,
@@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
#ifdef CONFIG_SPI_MASTER
codec->hw_write = do_spi_write;
#endif
+ if (io_types[i].spi_read)
+ codec->hw_read = io_types[i].spi_read;
codec->control_data = container_of(codec->dev,
struct spi_device,
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7c17b98..38b0013 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
IRQF_TRIGGER_FALLING,
gpios[i].name,
&gpios[i]);
- if (ret)
+ if (ret < 0)
goto err;
if (gpios[i].wake) {
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b575939..2879c88 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
codec_dai->active--;
codec->active--;
+ if (!cpu_dai->active && !codec_dai->active)
+ rtd->rate = 0;
+
/* Muting the DAC suppresses artifacts caused during digital
* shutdown, for example from stopping clocks.
*/
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 661373c..be27f1d 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
/* FIXME: Calculate automatically based on DAPM routes? */
- if (!machine_is_harmony() && !machine_is_ventana())
+ if (!machine_is_harmony())
snd_soc_dapm_nc_pin(dapm, "IN1L");
if (!machine_is_seaboard() && !machine_is_aebl())
snd_soc_dapm_nc_pin(dapm, "IN1R");
@@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
platform_set_drvdata(pdev, card);
snd_soc_card_set_drvdata(card, machine);
- if (machine_is_harmony() || machine_is_ventana()) {
+ if (machine_is_harmony()) {
card->dapm_routes = harmony_audio_map;
card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
} else if (machine_is_seaboard()) {
--
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