[alsa-devel] [PATCH 01/10] ASoC: Fix register cache sizes for Wolfson codecs
The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5acf43a..7788053 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -534,7 +534,7 @@ static int wm8731_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8731_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1f11ad2..1ae670a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -798,7 +798,7 @@ static int wm8750_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(wm8750_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c32e632..285c5ea 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,7 @@ static int wm8753_init(struct snd_soc_device *socdev) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = sizeof(wm8753_reg); + codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
if (codec->reg_cache == NULL)
The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com Cc: Philipp Zabel philipp.zabel@gmail.com --- sound/soc/codecs/uda1380.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index cb50486..6d5335b 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -655,8 +655,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM; - codec->reg_cache_size = sizeof(uda1380_reg); - codec->reg_cache_step = 2; + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache_step = 1; uda1380_reset(codec);
/* register pcms */
ASoC codec drivers frequently set the register cache size using sizeof() rather than ARRAY_SIZE(). For tlv320aicx either is correct since the registers are 8 bit but update to use ARRAY_SIZE() for clarity.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com Cc: Jarkko Nikula jarkko.nikula@nokia.com --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index dc8a38d..528c26a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1021,7 +1021,7 @@ static int aic3x_init(struct snd_soc_device *socdev) codec->set_bias_level = aic3x_set_bias_level; codec->dai = &aic3x_dai; codec->num_dai = 1; - codec->reg_cache_size = sizeof(aic3x_reg); + codec->reg_cache_size = ARRAY_SIZE(aic3x_reg); codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL); if (codec->reg_cache == NULL) return -ENOMEM;
Ensure that DAIs are prototyped in the codec drivers that define them.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/ac97.c | 1 + sound/soc/codecs/wm9712.c | 1 + 2 files changed, 2 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 7bf2081..300d007 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -21,6 +21,7 @@ #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/soc.h> +#include "ac97.h"
#define AC97_VERSION "0.6"
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index d9789f1..28ac66f 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,6 +22,7 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include "wm9712.h"
#define WM9712_VERSION "0.4"
This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/soc.h | 6 ++++-- sound/soc/davinci/davinci-i2s.c | 6 ++++-- sound/soc/pxa/pxa2xx-ac97.c | 3 ++- sound/soc/s3c24xx/s3c2412-i2s.c | 3 ++- sound/soc/s3c24xx/s3c2443-ac97.c | 6 ++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 3 ++- sound/soc/soc-core.c | 6 +++--- 7 files changed, 21 insertions(+), 12 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h index 9fa2093..56d2224 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -358,8 +358,10 @@ struct snd_soc_cpu_dai { unsigned char type;
/* DAI callbacks */ - int (*probe)(struct platform_device *pdev); - void (*remove)(struct platform_device *pdev); + int (*probe)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai); int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); int (*resume)(struct platform_device *pdev, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c421774..c3b545c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -295,7 +295,8 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; }
-static int davinci_i2s_probe(struct platform_device *pdev) +static int davinci_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; @@ -356,7 +357,8 @@ err_release_region: return ret; }
-static void davinci_i2s_remove(struct platform_device *pdev) +static void davinci_i2s_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 97ec2d9..cb94795 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #define pxa2xx_ac97_resume NULL #endif
-static int pxa2xx_ac97_probe(struct platform_device *pdev) +static int pxa2xx_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret;
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd..c463a82 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
-static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__);
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 0eed140..533565b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -209,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, };
-static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; u32 ac_glbctrl; @@ -260,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; }
-static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4c52f79..42e96b5 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -377,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
-static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f594ab8..c96a618 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -768,7 +768,7 @@ static int soc_probe(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev); + ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) goto cpu_dai_err; } @@ -798,7 +798,7 @@ cpu_dai_err: for (i--; i >= 0; i--) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); }
if (machine->remove) @@ -827,7 +827,7 @@ static int soc_remove(struct platform_device *pdev) for (i = 0; i < machine->num_links; i++) { struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) - cpu_dai->remove(pdev); + cpu_dai->remove(pdev, cpu_dai); }
if (machine->remove)
Reported-by: Rodolfo Giometti giometti@enneenne.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 4f516a5..aba3301 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -876,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, gpio |= 0x0018; break; case SND_SOC_DAIFMT_CBS_CFS: - reg |= 0x0200; + reg |= 0x2000; gpio |= 0x001a; break; case SND_SOC_DAIFMT_CBS_CFM:
From: Mike Montour mail@mmontour.net
Signed-off-by: Mike Montour mail@mmontour.net Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/wm8753.c | 75 ++++++++++++++++++++++++++++++--------------- 1 files changed, 50 insertions(+), 25 deletions(-)
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 285c5ea..00b4811 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -260,28 +260,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, return 1; }
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); +static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + /* 0000000 - 0101111 = "Analogue mute" */ + 0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0), + 48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0), +}; +static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0); +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
static const struct snd_kcontrol_new wm8753_snd_controls[] = { -SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), - -SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0), - -SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), -SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), - -SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), - -SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), -SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), -SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), - -SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), - -SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), -SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), +SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0, + adc_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, + 127, 0, out_tlv), + +SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv), + +SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, + 1, mix_tlv), +SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, + 7, 1, mix_tlv), +SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, + 1, voice_mix_tlv), + +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, + 1, 0), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, + 1, 0), + +SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv), +SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1, + mix_tlv), +SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1, + voice_mix_tlv), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
SOC_ENUM("Bass Boost", wm8753_enum[0]), @@ -291,10 +313,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1), SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
-SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv), -SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv), +SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, + rec_mix_tlv), +SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, + rec_mix_tlv),
-SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), +SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0, + pga_tlv), SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
@@ -326,8 +351,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]), SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), SOC_ENUM("Playback Phase", wm8753_enum[10]),
-SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), -SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), +SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv), +SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
From: Mike Montour mail@mmontour.net
Signed-off-by: Mike Montour mail@mmontour.net Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/soc.h | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h index 56d2224..1f5c621 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -100,6 +100,15 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } +#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmask, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmask, xinvert) } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \
From: Mike Montour mail@mmontour.net
Signed-off-by: Mike Montour mail@mmontour.net Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 files changed, 10 insertions(+), 6 deletions(-)
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c1a0161..3485123 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/tlv.h>
#include <asm/mach-types.h> #include <asm/hardware/scoop.h> @@ -474,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), };
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
From: Jarkko Nikula jarkko.nikula@nokia.com
Signed-off-by: Jarkko Nikula jarkko.nikula@nokia.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/omap/n810.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c32487b..c168a64 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -265,7 +265,7 @@ static int __init n810_soc_init(void) int err; struct device *dev;
- if (!machine_is_nokia_n810()) + if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) return -ENODEV;
n810_snd_device = platform_device_alloc("soc-audio", -1);
On Wed, 11 Jun 2008 13:47:08 +0100 "ext Mark Brown" broonie@opensource.wolfsonmicro.com wrote:
ASoC codec drivers frequently set the register cache size using sizeof () rather than ARRAY_SIZE(). For tlv320aicx either is correct since the registers are 8 bit but update to use ARRAY_SIZE() for clarity.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com Cc: Jarkko Nikula jarkko.nikula@nokia.com
sound/soc/codecs/tlv320aic3x.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)
Acked-by: Jarkko Nikula jarkko.nikula@nokia.com
At Wed, 11 Jun 2008 13:47:06 +0100, Mark Brown wrote:
The register cache size is used by the codec_reg sysfs file which works in terms of the register cache access functions rather than in terms of raw access to the cache so the size specified needs to be in terms of the number of elements.
Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
Thanks, all 10 patches taken to my git tree as now.
Takashi
participants (3)
-
Jarkko Nikula -
Mark Brown -
Takashi Iwai