Merge tag 'sound-4.0-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai...

Pull sound fixes from Takashi Iwai:
 "Here are a few more ASoC changes that have been gathered since rc1,
  but it's still fairly calm over all.  The only largish LOC is found in
  atmel driver, and it's just a removal of broken non-DT stuff.  The
  rest are all small driver-specific fixes, nothing to worry much"

* tag 'sound-4.0-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
  ALSA: hda - One more Dell macine needs DELL1_MIC_NO_PRESENCE quirk
  ALSA: opl3: small array underflow
  ALSA: line6: Clamp values correctly
  ALSA: msnd: add some missing curly braces
  ASoC: omap-pcm: Correct dma mask
  ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of
  ASoC: sam9g20_wm8731: drop machine_is_xxx
  ALSA: dice: fix wrong offsets for Dice interface
  ALSA: oxfw: fix a condition and return code in start_stream()
  ASoC: OMAP: mcbsp: Fix CLKX and CLKR pinmux when used as inputs
  ASoC: rt5677: Correct the routing paths of that after IF1/2 DACx Mux
  ASoC: sta32x: fix register range in regmap.
  ASoC: rt5670: Set RT5670_IRQ_CTRL1 non volatile
  ASoC: Intel: reset the DSP while suspending
  ASoC: Intel: save and restore the CSR register
  ASoC: Intel: update MMX ID to 3
  ASoC: max98357a: Add missing header files
  ASoC: cirrus: tlv320aic23 needs I2C
  ASoC: Samsung: add missing I2C/SPI dependencies
  ASoC: rt5670: Fix the speaker mono output issue
  ...

diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index f62780ed64adcc85465447838ee3b5d794690937..7821b07415a785c70982e03803ac34601770e3a9 100644 (file)
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -105,6 +105,8 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum,
                int pitchbend = chan->midi_pitchbend;
                int segment;
 
+               if (pitchbend < -0x2000)
+                       pitchbend = -0x2000;
                if (pitchbend > 0x1FFF)
                        pitchbend = 0x1FFF;
 

diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 27b044f84c816f5b4f15efa9782b7c4fbd1a217d..de7602bd69b569176ab5b41ab9065a0e04c7688a 100644 (file)
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -298,24 +298,24 @@
  */
 #define RX_ISOCHRONOUS                 0x008
 
-/*
- * Index of first quadlet to be interpreted; read/write.  If > 0, that many
- * quadlets at the beginning of each data block will be ignored, and all the
- * audio and MIDI quadlets will follow.
- */
-#define RX_SEQ_START                   0x00c
-
 /*
  * The number of audio channels; read-only.  There will be one quadlet per
  * channel.
  */
-#define RX_NUMBER_AUDIO                        0x010
+#define RX_NUMBER_AUDIO                        0x00c
 
 /*
  * The number of MIDI ports, 0-8; read-only.  If > 0, there will be one
  * additional quadlet in each data block, following the audio quadlets.
  */
-#define RX_NUMBER_MIDI                 0x014
+#define RX_NUMBER_MIDI                 0x010
+
+/*
+ * Index of first quadlet to be interpreted; read/write.  If > 0, that many
+ * quadlets at the beginning of each data block will be ignored, and all the
+ * audio and MIDI quadlets will follow.
+ */
+#define RX_SEQ_START                   0x014
 
 /*
  * Names of all audio channels; read-only.  Quadlets are byte-swapped.  Names

diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index f5c1d1bced59fee2e0012d9a5f96b85553bb53f8..ecfe20fd4de57868cefff26e8317ec54f180509a 100644 (file)
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry,
                } tx;
                struct {
                        u32 iso;
-                       u32 seq_start;
                        u32 number_audio;
                        u32 number_midi;
+                       u32 seq_start;
                        char names[RX_NAMES_SIZE];
                        u32 ac3_caps;
                        u32 ac3_enable;
@@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry,
                        break;
                snd_iprintf(buffer, "rx %u:\n", stream);
                snd_iprintf(buffer, "  iso channel: %d\n", (int)buf.rx.iso);
-               snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                snd_iprintf(buffer, "  audio channels: %u\n",
                            buf.rx.number_audio);
                snd_iprintf(buffer, "  midi ports: %u\n", buf.rx.number_midi);
+               snd_iprintf(buffer, "  sequence start: %u\n", buf.rx.seq_start);
                if (quadlets >= 68) {
                        dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE);
                        snd_iprintf(buffer, "  names: %s\n", buf.rx.names);

diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 29ccb3637164f82846022c37b2818e1a6ad4989e..e6757cd8572422813b1627f3c5718ec4c53f2aac 100644 (file)
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -171,9 +171,10 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
        }
 
        /* Wait first packet */
-       err = amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT);
-       if (err < 0)
+       if (!amdtp_stream_wait_callback(stream, CALLBACK_TIMEOUT)) {
                stop_stream(oxfw, stream);
+               err = -ETIMEDOUT;
+       }
 end:
        return err;
 }

diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
index 17e49a071af4497e5f39e0341cdee1db229cbc59..b408540798c1648d04a4ec1c36086844326affe1 100644 (file)
--- a/sound/isa/msnd/msnd_pinnacle_mixer.c
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -306,11 +306,12 @@ int snd_msndmix_new(struct snd_card *card)
        spin_lock_init(&chip->mixer_lock);
        strcpy(card->mixername, "MSND Pinnacle Mixer");
 
-       for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+       for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) {
                err = snd_ctl_add(card,
                                  snd_ctl_new1(snd_msnd_controls + idx, chip));
                if (err < 0)
                        return err;
+       }
 
        return 0;
 }

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2b24a8b3dac8c49d2bb55c3142eecd967e73608..526398a4a4428da11d5792c9c7f2a299cac9bd43 100644 (file)
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5209,6 +5209,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
                {0x17, 0x40000000},
                {0x1d, 0x40700001},
                {0x21, 0x02211040}),
+       SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+               ALC255_STANDARD_PINS,
+               {0x12, 0x90a60170},
+               {0x14, 0x90170140},
+               {0x17, 0x40000000},
+               {0x1d, 0x40700001},
+               {0x21, 0x02211050}),
        SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
                {0x12, 0x90a60130},
                {0x13, 0x40000000},

diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f5ad214663f98b4dbbdc3c2a4180d6142d09688c..8de836165cf2ed1978c4846ac56e84bcbae4e6d1 100644 (file)
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,8 +46,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/mach-types.h>
-
 #include "../codecs/wm8731.h"
 #include "atmel-pcm.h"
 #include "atmel_ssc_dai.h"
@@ -171,9 +169,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
        int ret;
 
        if (!np) {
-               if (!(machine_is_at91sam9g20ek() ||
-                       machine_is_at91sam9g20ek_2mmc()))
-                       return -ENODEV;
+               return -ENODEV;
        }
 
        ret = atmel_ssc_set_audio(0);
@@ -210,39 +206,37 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
        card->dev = &pdev->dev;
 
        /* Parse device node info */
-       if (np) {
-               ret = snd_soc_of_parse_card_name(card, "atmel,model");
-               if (ret)
-                       goto err;
-
-               ret = snd_soc_of_parse_audio_routing(card,
-                       "atmel,audio-routing");
-               if (ret)
-                       goto err;
-
-               /* Parse codec info */
-               at91sam9g20ek_dai.codec_name = NULL;
-               codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
-               if (!codec_np) {
-                       dev_err(&pdev->dev, "codec info missing\n");
-                       return -EINVAL;
-               }
-               at91sam9g20ek_dai.codec_of_node = codec_np;
-
-               /* Parse dai and platform info */
-               at91sam9g20ek_dai.cpu_dai_name = NULL;
-               at91sam9g20ek_dai.platform_name = NULL;
-               cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
-               if (!cpu_np) {
-                       dev_err(&pdev->dev, "dai and pcm info missing\n");
-                       return -EINVAL;
-               }
-               at91sam9g20ek_dai.cpu_of_node = cpu_np;
-               at91sam9g20ek_dai.platform_of_node = cpu_np;
-
-               of_node_put(codec_np);
-               of_node_put(cpu_np);
+       ret = snd_soc_of_parse_card_name(card, "atmel,model");
+       if (ret)
+               goto err;
+
+       ret = snd_soc_of_parse_audio_routing(card,
+               "atmel,audio-routing");
+       if (ret)
+               goto err;
+
+       /* Parse codec info */
+       at91sam9g20ek_dai.codec_name = NULL;
+       codec_np = of_parse_phandle(np, "atmel,audio-codec", 0);
+       if (!codec_np) {
+               dev_err(&pdev->dev, "codec info missing\n");
+               return -EINVAL;
+       }
+       at91sam9g20ek_dai.codec_of_node = codec_np;
+
+       /* Parse dai and platform info */
+       at91sam9g20ek_dai.cpu_dai_name = NULL;
+       at91sam9g20ek_dai.platform_name = NULL;
+       cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
+       if (!cpu_np) {
+               dev_err(&pdev->dev, "dai and pcm info missing\n");
+               return -EINVAL;
        }
+       at91sam9g20ek_dai.cpu_of_node = cpu_np;
+       at91sam9g20ek_dai.platform_of_node = cpu_np;
+
+       of_node_put(codec_np);
+       of_node_put(cpu_np);
 
        ret = snd_soc_register_card(card);
        if (ret) {

diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 7b7fbcd49e5e424c25e47d9e6a1aee77065358d5..c7cd60f009e93e69ef2220eec3eb6ec97a9e6500 100644 (file)
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -16,7 +16,7 @@ config SND_EP93XX_SOC_AC97
 
 config SND_EP93XX_SOC_SNAPPERCL15
         tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
-        depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+        depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C
         select SND_EP93XX_SOC_I2S
         select SND_SOC_TLV320AIC23_I2C
         help

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 064e6c18e10923fd75609b750405dbc33f9db6af..ea9f0e31f9d40d4cf4d8e626371698cb857fcb3a 100644 (file)
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -69,7 +69,7 @@ config SND_SOC_ALL_CODECS
        select SND_SOC_MAX98088 if I2C
        select SND_SOC_MAX98090 if I2C
        select SND_SOC_MAX98095 if I2C
-       select SND_SOC_MAX98357A
+       select SND_SOC_MAX98357A if GPIOLIB
        select SND_SOC_MAX9850 if I2C
        select SND_SOC_MAX9768 if I2C
        select SND_SOC_MAX9877 if I2C

diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 1806333ea29e5a6b49e2a0adb8c58257d34b7508..e9e6efbc21dd527299fb459ec01903fa8c24dae2 100644 (file)
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -12,9 +12,19 @@
  * max98357a.c -- MAX98357A ALSA SoC Codec driver
  */
 
-#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/err.h>
 #include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/kernel.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
 #include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
 
 #define DRV_NAME "max98357a"
 

diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index e1a4a45c57e229b12dcd174ac8453f8772b6e3b1..fd102613d20d8747c7554f5a3882c027fd0024c8 100644 (file)
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -225,7 +225,6 @@ static bool rt5670_volatile_register(struct device *dev, unsigned int reg)
        case RT5670_ADC_EQ_CTRL1:
        case RT5670_EQ_CTRL1:
        case RT5670_ALC_CTRL_1:
-       case RT5670_IRQ_CTRL1:
        case RT5670_IRQ_CTRL2:
        case RT5670_INT_IRQ_ST:
        case RT5670_IL_CMD:
@@ -2703,6 +2702,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
 
        regmap_write(rt5670->regmap, RT5670_RESET, 0);
 
+       regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val);
+       if (val >= 4)
+               regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980);
+       else
+               regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00);
+
        ret = regmap_register_patch(rt5670->regmap, init_list,
                                    ARRAY_SIZE(init_list));
        if (ret != 0)

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 5d0bb8748dd1df5cd6a262d7e3a925fe4b4b5efd..fb9c20eace3fbe9d7bb009862a17c0e8644ec296 100644 (file)
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3284,8 +3284,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
        { "IB45 Bypass Mux", "Bypass", "IB45 Mux" },
        { "IB45 Bypass Mux", "Pass SRC", "IB45 Mux" },
 
-       { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6" },
-       { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6" },
+       { "IB6 Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+       { "IB6 Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
        { "IB6 Mux", "SLB DAC 6", "SLB DAC6" },
        { "IB6 Mux", "STO4 ADC MIX L", "Stereo4 ADC MIXL" },
        { "IB6 Mux", "IF4 DAC L", "IF4 DAC L" },
@@ -3293,8 +3293,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
        { "IB6 Mux", "STO2 ADC MIX L", "Stereo2 ADC MIXL" },
        { "IB6 Mux", "STO3 ADC MIX L", "Stereo3 ADC MIXL" },
 
-       { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7" },
-       { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7" },
+       { "IB7 Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+       { "IB7 Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
        { "IB7 Mux", "SLB DAC 7", "SLB DAC7" },
        { "IB7 Mux", "STO4 ADC MIX R", "Stereo4 ADC MIXR" },
        { "IB7 Mux", "IF4 DAC R", "IF4 DAC R" },
@@ -3635,15 +3635,15 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
        { "DAC1 FS", NULL, "DAC1 MIXL" },
        { "DAC1 FS", NULL, "DAC1 MIXR" },
 
-       { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2" },
-       { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2" },
+       { "DAC2 L Mux", "IF1 DAC 2", "IF1 DAC2 Mux" },
+       { "DAC2 L Mux", "IF2 DAC 2", "IF2 DAC2 Mux" },
        { "DAC2 L Mux", "IF3 DAC L", "IF3 DAC L" },
        { "DAC2 L Mux", "IF4 DAC L", "IF4 DAC L" },
        { "DAC2 L Mux", "SLB DAC 2", "SLB DAC2" },
        { "DAC2 L Mux", "OB 2", "OutBound2" },
 
-       { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3" },
-       { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3" },
+       { "DAC2 R Mux", "IF1 DAC 3", "IF1 DAC3 Mux" },
+       { "DAC2 R Mux", "IF2 DAC 3", "IF2 DAC3 Mux" },
        { "DAC2 R Mux", "IF3 DAC R", "IF3 DAC R" },
        { "DAC2 R Mux", "IF4 DAC R", "IF4 DAC R" },
        { "DAC2 R Mux", "SLB DAC 3", "SLB DAC3" },
@@ -3651,29 +3651,29 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
        { "DAC2 R Mux", "Haptic Generator", "Haptic Generator" },
        { "DAC2 R Mux", "VAD ADC", "VAD ADC Mux" },
 
-       { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4" },
-       { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4" },
+       { "DAC3 L Mux", "IF1 DAC 4", "IF1 DAC4 Mux" },
+       { "DAC3 L Mux", "IF2 DAC 4", "IF2 DAC4 Mux" },
        { "DAC3 L Mux", "IF3 DAC L", "IF3 DAC L" },
        { "DAC3 L Mux", "IF4 DAC L", "IF4 DAC L" },
        { "DAC3 L Mux", "SLB DAC 4", "SLB DAC4" },
        { "DAC3 L Mux", "OB 4", "OutBound4" },
 
-       { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC4" },
-       { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC4" },
+       { "DAC3 R Mux", "IF1 DAC 5", "IF1 DAC5 Mux" },
+       { "DAC3 R Mux", "IF2 DAC 5", "IF2 DAC5 Mux" },
        { "DAC3 R Mux", "IF3 DAC R", "IF3 DAC R" },
        { "DAC3 R Mux", "IF4 DAC R", "IF4 DAC R" },
        { "DAC3 R Mux", "SLB DAC 5", "SLB DAC5" },
        { "DAC3 R Mux", "OB 5", "OutBound5" },
 
-       { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6" },
-       { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6" },
+       { "DAC4 L Mux", "IF1 DAC 6", "IF1 DAC6 Mux" },
+       { "DAC4 L Mux", "IF2 DAC 6", "IF2 DAC6 Mux" },
        { "DAC4 L Mux", "IF3 DAC L", "IF3 DAC L" },
        { "DAC4 L Mux", "IF4 DAC L", "IF4 DAC L" },
        { "DAC4 L Mux", "SLB DAC 6", "SLB DAC6" },
        { "DAC4 L Mux", "OB 6", "OutBound6" },
 
-       { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7" },
-       { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7" },
+       { "DAC4 R Mux", "IF1 DAC 7", "IF1 DAC7 Mux" },
+       { "DAC4 R Mux", "IF2 DAC 7", "IF2 DAC7 Mux" },
        { "DAC4 R Mux", "IF3 DAC R", "IF3 DAC R" },
        { "DAC4 R Mux", "IF4 DAC R", "IF4 DAC R" },
        { "DAC4 R Mux", "SLB DAC 7", "SLB DAC7" },

diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 3a1343fa109b482860811baea1566d354e574399..007a0e3bc2735c0c6f9d8325f15445657fcfd374 100644 (file)
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -106,13 +106,11 @@ static const struct reg_default sta32x_regs[] = {
 };
 
 static const struct regmap_range sta32x_write_regs_range[] = {
-       regmap_reg_range(STA32X_CONFA,  STA32X_AUTO2),
-       regmap_reg_range(STA32X_C1CFG,  STA32X_FDRC2),
+       regmap_reg_range(STA32X_CONFA,  STA32X_FDRC2),
 };
 
 static const struct regmap_range sta32x_read_regs_range[] = {
-       regmap_reg_range(STA32X_CONFA,  STA32X_AUTO2),
-       regmap_reg_range(STA32X_C1CFG,  STA32X_FDRC2),
+       regmap_reg_range(STA32X_CONFA,  STA32X_FDRC2),
 };
 
 static const struct regmap_range sta32x_volatile_regs_range[] = {

diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 2595611e8a6ded3876345767bde2ce51e079c9ee..b9fabbf69db6c9830ebf0de211f62735a24297b8 100644 (file)
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -603,10 +603,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
        factor = (div2 + 1) * (7 * psr + 1) * 2;
 
        for (i = 0; i < 255; i++) {
-               /* The bclk rate must be smaller than 1/5 sysclk rate */
-               if (factor * (i + 1) < 5)
-                       continue;
-
                tmprate = freq * factor * (i + 2);
 
                if (baudclk_is_used)
@@ -614,6 +610,13 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream,
                else
                        clkrate = clk_round_rate(ssi_private->baudclk, tmprate);
 
+               /*
+                * Hardware limitation: The bclk rate must be
+                * never greater than 1/5 IPG clock rate
+                */
+               if (clkrate * 5 > clk_get_rate(ssi_private->clk))
+                       continue;
+
                clkrate /= factor;
                afreq = clkrate / (i + 1);
 

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index f7c6734bd5daee1dafd76ba8b6feafa05b2a6c52..fb550b5869d21f73307d36d2edc88ebcffa61f0c 100644 (file)
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node,
                            strlen(dai_link->cpu_dai_name)   +
                            strlen(dai_link->codec_dai_name) + 2,
                            GFP_KERNEL);
+       if (!name) {
+               ret = -ENOMEM;
+               goto dai_link_of_err;
+       }
+
        sprintf(name, "%s-%s", dai_link->cpu_dai_name,
                                dai_link->codec_dai_name);
        dai_link->name = dai_link->stream_name = name;

diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h
index dfebfdd5eb2aaa65e1f21aa5d65d2c9052486b75..daecc58f28afafb28c8172649355ff9b30749783 100644 (file)
--- a/sound/soc/intel/sst-atom-controls.h
+++ b/sound/soc/intel/sst-atom-controls.h
@@ -150,7 +150,7 @@ enum sst_cmd_type {
 
 enum sst_task {
        SST_TASK_SBA = 1,
-       SST_TASK_MMX,
+       SST_TASK_MMX = 3,
 };
 
 enum sst_type {

diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c
index 8a8d56a146e75a19d4587e169cb967b730b3eb36..11c578651c1c8cc778904ebb7665587cde44beef 100644 (file)
--- a/sound/soc/intel/sst/sst.c
+++ b/sound/soc/intel/sst/sst.c
@@ -350,7 +350,9 @@ static inline void sst_save_shim64(struct intel_sst_drv *ctx,
 
        spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
 
-       shim_regs->imrx = sst_shim_read64(shim, SST_IMRX),
+       shim_regs->imrx = sst_shim_read64(shim, SST_IMRX);
+       shim_regs->csr = sst_shim_read64(shim, SST_CSR);
+
 
        spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
 }
@@ -367,6 +369,7 @@ static inline void sst_restore_shim64(struct intel_sst_drv *ctx,
         */
        spin_lock_irqsave(&ctx->ipc_spin_lock, irq_flags);
        sst_shim_write64(shim, SST_IMRX, shim_regs->imrx),
+       sst_shim_write64(shim, SST_CSR, shim_regs->csr),
        spin_unlock_irqrestore(&ctx->ipc_spin_lock, irq_flags);
 }
 
@@ -379,6 +382,10 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx)
         * initially active. So change the state to active before
         * enabling the pm
         */
+
+       if (!acpi_disabled)
+               pm_runtime_set_active(ctx->dev);
+
        pm_runtime_enable(ctx->dev);
 
        if (acpi_disabled)
@@ -409,6 +416,7 @@ static int intel_sst_runtime_suspend(struct device *dev)
        synchronize_irq(ctx->irq_num);
        flush_workqueue(ctx->post_msg_wq);
 
+       ctx->ops->reset(ctx);
        /* save the shim registers because PMC doesn't save state */
        sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64);
 

diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c
index ccfb41c22e53b11cedbf5d8272a56a757ad283d1..f7eb42aa3f3893ca7cfd76fd2f8a96ff986d68b2 100644 (file)
--- a/sound/soc/omap/omap-hdmi-audio.c
+++ b/sound/soc/omap/omap-hdmi-audio.c
@@ -352,6 +352,9 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
                return ret;
 
        card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+       if (!card)
+               return -ENOMEM;
+
        card->name = devm_kasprintf(dev, GFP_KERNEL,
                                    "HDMI %s", dev_name(ad->dssdev));
        card->owner = THIS_MODULE;

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index c7eb9dd67f608c47ffa93a97c1824e3dae3c867b..fd99d89de6a854606357b50c44deeb43bcfb04ab 100644 (file)
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
        case OMAP_MCBSP_SYSCLK_CLKX_EXT:
                regs->srgr2     |= CLKSM;
+               regs->pcr0      |= SCLKME;
+               /*
+                * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+                * disable output on those pins. This enables to inject the
+                * reference clock through CLKX/CLKR. For this to work
+                * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+                */
+               regs->pcr0      &= ~CLKXM;
+               break;
        case OMAP_MCBSP_SYSCLK_CLKR_EXT:
                regs->pcr0      |= SCLKME;
+               /* Disable ouput on CLKR pin in master mode */
+               regs->pcr0      &= ~CLKRM;
                break;
        default:
                err = -ENODEV;

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index f4b05bc23e4bfb4d27adf8dea4702539c2696609..1343ecbf0bd5ee307d78d05f8a2e44abadc6c877 100644 (file)
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -201,7 +201,7 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
        struct snd_pcm *pcm = rtd->pcm;
        int ret;
 
-       ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(64));
+       ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
        if (ret)
                return ret;
 

diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3cebf6ca03dfb91891cd6e6d5eb51649b504c0b1..0632a36852c8476416f8fcb78d625abb98d658cf 100644 (file)
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -174,7 +174,7 @@ config SND_SOC_SMDK_WM8994_PCM
 
 config SND_SOC_SPEYSIDE
        tristate "Audio support for Wolfson Speyside"
-       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C && SPI_MASTER
        select SND_SAMSUNG_I2S
        select SND_SOC_WM8996
        select SND_SOC_WM9081
@@ -189,7 +189,7 @@ config SND_SOC_TOBERMORY
 
 config SND_SOC_BELLS
        tristate "Audio support for Wolfson Bells"
-       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA
+       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER
        select SND_SAMSUNG_I2S
        select SND_SOC_WM5102
        select SND_SOC_WM5110
@@ -206,7 +206,7 @@ config SND_SOC_LOWLAND
 
 config SND_SOC_LITTLEMILL
        tristate "Audio support for Wolfson Littlemill"
-       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
+       depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C
        select SND_SAMSUNG_I2S
        select MFD_WM8994
        select SND_SOC_WM8994
@@ -223,7 +223,7 @@ config SND_SOC_SNOW
 
 config SND_SOC_ODROIDX2
        tristate "Audio support for Odroid-X2 and Odroid-U3"
-       depends on SND_SOC_SAMSUNG
+       depends on SND_SOC_SAMSUNG && I2C
        select SND_SOC_MAX98090
        select SND_SAMSUNG_I2S
        help
@@ -231,6 +231,6 @@ config SND_SOC_ODROIDX2
 
 config SND_SOC_ARNDALE_RT5631_ALC5631
         tristate "Audio support for RT5631(ALC5631) on Arndale Board"
-        depends on SND_SOC_SAMSUNG
+        depends on SND_SOC_SAMSUNG && I2C
         select SND_SAMSUNG_I2S
         select SND_SOC_RT5631

diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 1b53605f7154394d1746cb0a692953e296848c2a..110577c523179c3974a2ebe45ef577ea651aa5ce 100644 (file)
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -1252,6 +1252,8 @@ static int rsnd_probe(struct platform_device *pdev)
                        goto exit_snd_probe;
        }
 
+       dev_set_drvdata(dev, priv);
+
        /*
         *      asoc register
         */
@@ -1268,8 +1270,6 @@ static int rsnd_probe(struct platform_device *pdev)
                goto exit_snd_soc;
        }
 
-       dev_set_drvdata(dev, priv);
-
        pm_runtime_enable(dev);
 
        dev_info(dev, "probed\n");

diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 05dee690f4876167ace542fe500bb077bec26762..97ed593f6010f6dccf4e5eb5db39e4a2381a7692 100644 (file)
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -39,7 +39,7 @@ static void change_volume(struct urb *urb_out, int volume[],
                for (; p < buf_end; ++p) {
                        short pv = le16_to_cpu(*p);
                        int val = (pv * volume[chn & 1]) >> 8;
-                       pv = clamp(val, 0x7fff, -0x8000);
+                       pv = clamp(val, -0x8000, 0x7fff);
                        *p = cpu_to_le16(pv);
                        ++chn;
                }
@@ -54,7 +54,7 @@ static void change_volume(struct urb *urb_out, int volume[],
 
                        val = p[0] + (p[1] << 8) + ((signed char)p[2] << 16);
                        val = (val * volume[chn & 1]) >> 8;
-                       val = clamp(val, 0x7fffff, -0x800000);
+                       val = clamp(val, -0x800000, 0x7fffff);
                        p[0] = val;
                        p[1] = val >> 8;
                        p[2] = val >> 16;
@@ -126,7 +126,7 @@ static void add_monitor_signal(struct urb *urb_out, unsigned char *signal,
                        short pov = le16_to_cpu(*po);
                        short piv = le16_to_cpu(*pi);
                        int val = pov + ((piv * volume) >> 8);
-                       pov = clamp(val, 0x7fff, -0x8000);
+                       pov = clamp(val, -0x8000, 0x7fff);
                        *po = cpu_to_le16(pov);
                }
        }