diff options
author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2012-10-25 18:36:02 +0200 |
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committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2012-10-25 18:36:02 +0200 |
commit | 456ba5a7802e58eccb5aa9751b3ab515ef99b9ca (patch) | |
tree | 4ca4dd3726b34dead51af13b67475af7bf857893 /sound | |
parent | ASoC: ux500_msp_i2s: Fix devm_* and return code merge error (diff) | |
parent | ASoC: wm8994: Only enable extra BCLK cycles when required (diff) | |
download | linux-456ba5a7802e58eccb5aa9751b3ab515ef99b9ca.tar.xz linux-456ba5a7802e58eccb5aa9751b3ab515ef99b9ca.zip |
Merge remote-tracking branches 'asoc/fix/ux500' and 'asoc/fix/wm8994' into for-3.7
Diffstat (limited to 'sound')
-rw-r--r-- | sound/pci/ac97/ac97_codec.c | 2 | ||||
-rw-r--r-- | sound/pci/emu10k1/emu10k1_main.c | 9 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 42 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 4 | ||||
-rw-r--r-- | sound/soc/Makefile | 5 | ||||
-rw-r--r-- | sound/soc/codecs/da9055.c | 22 | ||||
-rw-r--r-- | sound/soc/codecs/twl6040.c | 8 | ||||
-rw-r--r-- | sound/soc/codecs/wm2200.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 18 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.h | 1 | ||||
-rw-r--r-- | sound/soc/omap/ams-delta.c | 63 | ||||
-rw-r--r-- | sound/soc/omap/omap-abe-twl6040.c | 2 | ||||
-rw-r--r-- | sound/soc/omap/omap-mcpdm.c | 9 | ||||
-rw-r--r-- | sound/soc/pxa/mmp-pcm.c | 2 | ||||
-rw-r--r-- | sound/soc/samsung/bells.c | 4 | ||||
-rw-r--r-- | sound/soc/sh/fsi.c | 15 | ||||
-rw-r--r-- | sound/soc/soc-jack.c | 7 |
17 files changed, 136 insertions, 80 deletions
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 9473fca9681d..8b0f99688303 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1271,6 +1271,8 @@ static int snd_ac97_cvol_new(struct snd_card *card, char *name, int reg, unsigne tmp.index = ac97->num; kctl = snd_ctl_new1(&tmp, ac97); } + if (!kctl) + return -ENOMEM; if (reg >= AC97_PHONE && reg <= AC97_PCM) set_tlv_db_scale(kctl, db_scale_5bit_12db_max); else diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index bed4485f34f6..c21adb6ef1d5 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1416,6 +1416,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ + /* Tested by Maxim Kachur <mcdebugger@duganet.ru> 17th Oct 2012. */ + /* This is MAEM8986, 0202 is MAEM8980 */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, + .driver = "Audigy2", .name = "E-mu 1010 PCIe [MAEM8986]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James@superbug.co.uk 8th July 2005. */ /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6833835a218b..72b085ae7d46 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -501,6 +501,7 @@ struct azx { /* VGA-switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ unsigned int disabled:1; /* disabled by VGA-switcher */ @@ -2157,9 +2158,12 @@ static unsigned int azx_get_position(struct azx *chip, if (delay < 0) delay += azx_dev->bufsize; if (delay >= azx_dev->period_bytes) { - snd_printdd("delay %d > period_bytes %d\n", - delay, azx_dev->period_bytes); - delay = 0; /* something is wrong */ + snd_printk(KERN_WARNING SFX + "Unstable LPIB (%d >= %d); " + "disabling LPIB delay counting\n", + delay, azx_dev->period_bytes); + delay = 0; + chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } azx_dev->substream->runtime->delay = bytes_to_frames(azx_dev->substream->runtime, delay); @@ -2640,7 +2644,9 @@ static void azx_vs_set_state(struct pci_dev *pci, if (disabled) { azx_suspend(&pci->dev); chip->disabled = true; - snd_hda_lock_devices(chip->bus); + if (snd_hda_lock_devices(chip->bus)) + snd_printk(KERN_WARNING SFX + "Cannot lock devices!\n"); } else { snd_hda_unlock_devices(chip->bus); chip->disabled = false; @@ -2683,14 +2689,20 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { static int __devinit register_vga_switcheroo(struct azx *chip) { + int err; + if (!chip->use_vga_switcheroo) return 0; /* FIXME: currently only handling DIS controller * is there any machine with two switchable HDMI audio controllers? */ - return vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, VGA_SWITCHEROO_DIS, chip->bus != NULL); + if (err < 0) + return err; + chip->vga_switcheroo_registered = 1; + return 0; } #else #define init_vga_switcheroo(chip) /* NOP */ @@ -2712,7 +2724,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(chip)) { if (chip->disabled && chip->bus) snd_hda_unlock_devices(chip->bus); - vga_switcheroo_unregister_client(chip->pci); + if (chip->vga_switcheroo_registered) + vga_switcheroo_unregister_client(chip->pci); } if (chip->initialized) { @@ -2813,8 +2826,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF), - SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), @@ -3062,14 +3073,6 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } ok: - err = register_vga_switcheroo(chip); - if (err < 0) { - snd_printk(KERN_ERR SFX - "Error registering VGA-switcheroo client\n"); - azx_free(chip); - return err; - } - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); @@ -3340,6 +3343,13 @@ static int __devinit azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + err = register_vga_switcheroo(chip); + if (err < 0) { + snd_printk(KERN_ERR SFX + "Error registering VGA-switcheroo client\n"); + goto out_free; + } + dev++; return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8253b4eeb6a1..48d9d609f89b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2598,8 +2598,10 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return "PCM"; break; } - if (snd_BUG_ON(ch >= ARRAY_SIZE(channel_name))) + if (ch >= ARRAY_SIZE(channel_name)) { + snd_BUG(); return "PCM"; + } return channel_name[ch]; } diff --git a/sound/soc/Makefile b/sound/soc/Makefile index bcbf1d00aa85..99f32f7c0692 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,8 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o -snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o -obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o +ifneq ($(CONFIG_SND_SOC_DMAENGINE_PCM),) +snd-soc-core-objs += soc-dmaengine-pcm.o +endif obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd36399..f379b085c392 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -178,6 +178,12 @@ #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -476,7 +482,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -485,6 +491,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -520,6 +536,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index e8f97af75928..00b85cc1b9a3 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -820,10 +820,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"VIBRA DAC", NULL, "Vibra Playback"}, /* ADC -> Stream mapping */ - {"ADC Left", NULL, "Legacy Capture"}, - {"ADC Left", NULL, "Capture"}, - {"ADC Right", NULL, "Legacy Capture"}, - {"ADC Right", NULL, "Capture"}, + {"Legacy Capture" , NULL, "ADC Left"}, + {"Capture", NULL, "ADC Left"}, + {"Legacy Capture", NULL, "ADC Right"}, + {"Capture" , NULL, "ADC Right"}, /* Capture path */ {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index efa93dbb0191..eab64a193989 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1028,7 +1028,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, - WM2200_SPK1R_MUTE_SHIFT, 1, 0), + WM2200_SPK1R_MUTE_SHIFT, 1, 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -2091,6 +2091,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, switch (wm2200->rev) { case 0: + case 1: ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch, ARRAY_SIZE(wm2200_reva_patch)); if (ret != 0) { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2b2dadc54dac..3fddc7ad1127 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1045,6 +1045,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; @@ -1063,6 +1064,10 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: + /* Don't enable timeslot 2 if not in use */ + if (wm8994->channels[0] <= 2) + mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); if ((val & WM8994_AIF1ADCL_SRC) && (val & WM8994_AIF1ADCR_SRC)) @@ -2687,7 +2692,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 4; + bclk_rate = params_rate(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -2708,6 +2713,17 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8994->channels[id] = params_channels(params); + switch (params_channels(params)) { + case 1: + case 2: + bclk_rate *= 2; + break; + default: + bclk_rate *= 4; + break; + } + /* Try to find an appropriate sample rate; look for an exact match. */ for (i = 0; i < ARRAY_SIZE(srs); i++) if (srs[i].rate == params_rate(params)) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f142ec198db3..ccbce5791e95 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -77,6 +77,7 @@ struct wm8994_priv { int sysclk_rate[2]; int mclk[2]; int aifclk[2]; + int channels[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index dc0ee7626626..d8e96b2cd03e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -575,56 +575,53 @@ static struct snd_soc_card ams_delta_audio_card = { }; /* Module init/exit */ -static struct platform_device *ams_delta_audio_platform_device; -static struct platform_device *cx20442_platform_device; - -static int __init ams_delta_module_init(void) +static __devinit int ams_delta_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &ams_delta_audio_card; int ret; - if (!(machine_is_ams_delta())) - return -ENODEV; - - ams_delta_audio_platform_device = - platform_device_alloc("soc-audio", -1); - if (!ams_delta_audio_platform_device) - return -ENOMEM; + card->dev = &pdev->dev; - platform_set_drvdata(ams_delta_audio_platform_device, - &ams_delta_audio_card); - - ret = platform_device_add(ams_delta_audio_platform_device); - if (ret) - goto err; - - /* - * Codec platform device could be registered from elsewhere (board?), - * but I do it here as it makes sense only if used with the card. - */ - cx20442_platform_device = - platform_device_register_simple("cx20442-codec", -1, NULL, 0); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + card->dev = NULL; + return ret; + } return 0; -err: - platform_device_put(ams_delta_audio_platform_device); - return ret; } -late_initcall(ams_delta_module_init); -static void __exit ams_delta_module_exit(void) +static int __devexit ams_delta_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + if (tty_unregister_ldisc(N_V253) != 0) - dev_warn(&ams_delta_audio_platform_device->dev, + dev_warn(&pdev->dev, "failed to unregister V253 line discipline\n"); snd_soc_jack_free_gpios(&ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); - platform_device_unregister(cx20442_platform_device); - platform_device_unregister(ams_delta_audio_platform_device); + snd_soc_unregister_card(card); + card->dev = NULL; + return 0; } -module_exit(ams_delta_module_exit); + +#define DRV_NAME "ams-delta-audio" + +static struct platform_driver ams_delta_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ams_delta_probe, + .remove = __devexit_p(ams_delta_remove), +}; + +module_platform_driver(ams_delta_driver); MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>"); MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 4a73ef3ae12f..a57a4e68dcc6 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -216,7 +216,7 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index c02b001ee4b5..56965bb3275c 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -40,7 +40,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <plat/omap_hwmod.h> #include "omap-mcpdm.h" #include "omap-pcm.h" @@ -260,13 +259,9 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&mcpdm->mutex); if (!dai->active) { - /* Enable watch dog for ES above ES 1.0 to avoid saturation */ - if (omap_rev() != OMAP4430_REV_ES1_0) { - u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); + u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL); - omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, - ctrl | MCPDM_WD_EN); - } + omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN); omap_mcpdm_open_streams(mcpdm); } mutex_unlock(&mcpdm->mutex); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 73ac5463c9e4..e834faf859fd 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -15,13 +15,13 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <linux/platform_data/dma-mmp_tdma.h> #include <linux/platform_data/mmp_audio.h> #include <sound/pxa2xx-lib.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <mach/sram.h> #include <sound/dmaengine_pcm.h> struct mmp_dma_data { diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 5dc10dfc0d42..b0d46d63d55e 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -212,7 +212,7 @@ static struct snd_soc_dai_link bells_dai_wm5102[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF @@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = { { .name = "Sub", .stream_name = "Sub", - .cpu_dai_name = "wm5102-aif3", + .cpu_dai_name = "wm5110-aif3", .codec_dai_name = "wm9081-hifi", .codec_name = "wm9081.1-006c", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5328ae5539f1..9d7f30774a44 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -20,6 +20,7 @@ #include <linux/sh_dma.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/workqueue.h> #include <sound/soc.h> #include <sound/sh_fsi.h> @@ -223,7 +224,7 @@ struct fsi_stream { */ struct dma_chan *chan; struct sh_dmae_slave slave; /* see fsi_handler_init() */ - struct tasklet_struct tasklet; + struct work_struct work; dma_addr_t dma; }; @@ -1085,9 +1086,9 @@ static void fsi_dma_complete(void *data) snd_pcm_period_elapsed(io->substream); } -static void fsi_dma_do_tasklet(unsigned long data) +static void fsi_dma_do_work(struct work_struct *work) { - struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_stream *io = container_of(work, struct fsi_stream, work); struct fsi_priv *fsi = fsi_stream_to_priv(io); struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; @@ -1129,7 +1130,7 @@ static void fsi_dma_do_tasklet(unsigned long data) * FIXME * * In DMAEngine case, codec and FSI cannot be started simultaneously - * since FSI is using tasklet. + * since FSI is using the scheduler work queue. * Therefore, in capture case, probably FSI FIFO will have got * overflow error in this point. * in that case, DMA cannot start transfer until error was cleared. @@ -1153,7 +1154,7 @@ static bool fsi_dma_filter(struct dma_chan *chan, void *param) static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_schedule(&io->tasklet); + schedule_work(&io->work); return 0; } @@ -1195,14 +1196,14 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev return fsi_stream_probe(fsi, dev); } - tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); + INIT_WORK(&io->work, fsi_dma_do_work); return 0; } static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) { - tasklet_kill(&io->tasklet); + cancel_work_sync(&io->work); fsi_stream_stop(fsi, io); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa0fd8ddae90..1ab5fe04bfcc 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,7 +22,7 @@ /** * snd_soc_jack_new - Create a new jack - * @card: ASoC card + * @codec: ASoC codec * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack @@ -133,12 +133,13 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); /** * snd_soc_jack_get_type - Based on the mic bias value, this function returns - * the type of jack from the zones delcared in the jack type + * the type of jack from the zones declared in the jack type * + * @jack: ASoC jack * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in * * Based on the mic bias value passed, this function helps identify - * the type of jack from the already delcared jack zones + * the type of jack from the already declared jack zones */ int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) { |