From 4f7931db90a5b812460afc756022b0fa2cfba9ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Nov 2011 07:52:15 +0100 Subject: [PATCH] --- yaml --- r: 276000 b: refs/heads/master c: 61071594f64ed12328046f94716d1d744bddc0a1 h: refs/heads/master v: v3 --- [refs] | 2 +- trunk/MAINTAINERS | 1 + trunk/sound/pci/hda/patch_realtek.c | 24 +++++------ trunk/sound/soc/codecs/adau1373.c | 2 +- trunk/sound/soc/codecs/cs4271.c | 8 ++-- trunk/sound/soc/codecs/rt5631.c | 2 +- trunk/sound/soc/codecs/sgtl5000.c | 2 +- trunk/sound/soc/codecs/sta32x.c | 63 +---------------------------- trunk/sound/soc/codecs/sta32x.h | 1 - trunk/sound/soc/codecs/wm8731.c | 1 - trunk/sound/soc/codecs/wm8753.c | 3 -- trunk/sound/soc/codecs/wm8962.c | 4 +- trunk/sound/soc/codecs/wm8993.c | 2 +- trunk/sound/soc/codecs/wm9081.c | 10 ++--- trunk/sound/soc/codecs/wm9090.c | 6 +-- trunk/sound/soc/codecs/wm_hubs.c | 2 +- trunk/sound/soc/fsl/fsl_ssi.c | 1 - 17 files changed, 33 insertions(+), 101 deletions(-) diff --git a/[refs] b/[refs] index 38d9a7554804..18ec16153afd 100644 --- a/[refs] +++ b/[refs] @@ -1,2 +1,2 @@ --- -refs/heads/master: 77088cc97315e9bc713e335fb082ad26d065a4cf +refs/heads/master: 61071594f64ed12328046f94716d1d744bddc0a1 diff --git a/trunk/MAINTAINERS b/trunk/MAINTAINERS index fd7e441b5ea7..c802e5fa2d11 100644 --- a/trunk/MAINTAINERS +++ b/trunk/MAINTAINERS @@ -5648,6 +5648,7 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS +M: Jassi Brar M: Sangbeom Kim L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported diff --git a/trunk/sound/pci/hda/patch_realtek.c b/trunk/sound/pci/hda/patch_realtek.c index 06c0c12d4fec..cbde019d3d52 100644 --- a/trunk/sound/pci/hda/patch_realtek.c +++ b/trunk/sound/pci/hda/patch_realtek.c @@ -277,6 +277,12 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur) return false; } +static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) +{ + return spec->capsrc_nids ? + spec->capsrc_nids[idx] : spec->adc_nids[idx]; +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -303,8 +309,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, adc_idx = spec->dyn_adc_idx[idx]; } - nid = spec->capsrc_nids ? - spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; + nid = get_capsrc(spec, adc_idx); /* no selection? */ num_conns = snd_hda_get_conn_list(codec, nid, NULL); @@ -1058,8 +1063,7 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) hda_nid_t pin = spec->imux_pins[i]; int c; for (c = 0; c < spec->num_adc_nids; c++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[c] : spec->adc_nids[c]; + hda_nid_t cap = get_capsrc(spec, c); int idx = get_connection_index(codec, cap, pin); if (idx >= 0) { imux->items[i].index = idx; @@ -1969,10 +1973,8 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - const hda_nid_t *nids = spec->capsrc_nids; - if (!nids) - nids = spec->adc_nids; - err = snd_hda_add_nid(codec, kctl, i, nids[i]); + err = snd_hda_add_nid(codec, kctl, i, + get_capsrc(spec, i)); if (err < 0) return err; } @@ -2759,8 +2761,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec) } for (c = 0; c < num_adcs; c++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[c] : spec->adc_nids[c]; + hda_nid_t cap = get_capsrc(spec, c); idx = get_connection_index(codec, cap, pin); if (idx >= 0) { spec->imux_pins[imux->num_items] = pin; @@ -3706,8 +3707,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) if (!pin) return 0; for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[i] : spec->adc_nids[i]; + hda_nid_t cap = get_capsrc(spec, i); int idx; idx = get_connection_index(codec, cap, pin); diff --git a/trunk/sound/soc/codecs/adau1373.c b/trunk/sound/soc/codecs/adau1373.c index 45c63028b40d..1ccf8dd47576 100644 --- a/trunk/sound/soc/codecs/adau1373.c +++ b/trunk/sound/soc/codecs/adau1373.c @@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = { }; static const unsigned int adau1373_bass_tlv[] = { - TLV_DB_RANGE_HEAD(3), + TLV_DB_RANGE_HEAD(4), 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), diff --git a/trunk/sound/soc/codecs/cs4271.c b/trunk/sound/soc/codecs/cs4271.c index 69fde1506fe1..23d1bd5dadda 100644 --- a/trunk/sound/soc/codecs/cs4271.c +++ b/trunk/sound/soc/codecs/cs4271.c @@ -434,8 +434,7 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { int ret; /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); if (ret < 0) return ret; return 0; @@ -502,9 +501,8 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); diff --git a/trunk/sound/soc/codecs/rt5631.c b/trunk/sound/soc/codecs/rt5631.c index 4646e808b90a..27a078cbb6eb 100644 --- a/trunk/sound/soc/codecs/rt5631.c +++ b/trunk/sound/soc/codecs/rt5631.c @@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(6), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), diff --git a/trunk/sound/soc/codecs/sgtl5000.c b/trunk/sound/soc/codecs/sgtl5000.c index bbcf921166f7..d15695d1c273 100644 --- a/trunk/sound/soc/codecs/sgtl5000.c +++ b/trunk/sound/soc/codecs/sgtl5000.c @@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ static const unsigned int mic_gain_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(4), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; diff --git a/trunk/sound/soc/codecs/sta32x.c b/trunk/sound/soc/codecs/sta32x.c index d2f37152f940..bb82408ab8e1 100644 --- a/trunk/sound/soc/codecs/sta32x.c +++ b/trunk/sound/soc/codecs/sta32x.c @@ -76,8 +76,6 @@ struct sta32x_priv { unsigned int mclk; unsigned int format; - - u32 coef_shadow[STA32X_COEF_COUNT]; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -229,7 +227,6 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); int numcoef = kcontrol->private_value >> 16; int index = kcontrol->private_value & 0xffff; unsigned int cfud; @@ -242,11 +239,6 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, snd_soc_write(codec, STA32X_CFUD, cfud); snd_soc_write(codec, STA32X_CFADDR2, index); - for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++) - sta32x->coef_shadow[index + i] = - (ucontrol->value.bytes.data[3 * i] << 16) - | (ucontrol->value.bytes.data[3 * i + 1] << 8) - | (ucontrol->value.bytes.data[3 * i + 2]); for (i = 0; i < 3 * numcoef; i++) snd_soc_write(codec, STA32X_B1CF1 + i, ucontrol->value.bytes.data[i]); @@ -260,48 +252,6 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } -int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) -{ - struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); - unsigned int cfud; - int i; - - /* preserve reserved bits in STA32X_CFUD */ - cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; - - for (i = 0; i < STA32X_COEF_COUNT; i++) { - snd_soc_write(codec, STA32X_CFADDR2, i); - snd_soc_write(codec, STA32X_B1CF1, - (sta32x->coef_shadow[i] >> 16) & 0xff); - snd_soc_write(codec, STA32X_B1CF2, - (sta32x->coef_shadow[i] >> 8) & 0xff); - snd_soc_write(codec, STA32X_B1CF3, - (sta32x->coef_shadow[i]) & 0xff); - /* chip documentation does not say if the bits are - * self-clearing, so do it explicitly */ - snd_soc_write(codec, STA32X_CFUD, cfud); - snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); - } - return 0; -} - -int sta32x_cache_sync(struct snd_soc_codec *codec) -{ - unsigned int mute; - int rc; - - if (!codec->cache_sync) - return 0; - - /* mute during register sync */ - mute = snd_soc_read(codec, STA32X_MMUTE); - snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); - sta32x_sync_coef_shadow(codec); - rc = snd_soc_cache_sync(codec); - snd_soc_write(codec, STA32X_MMUTE, mute); - return rc; -} - #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -711,7 +661,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, return ret; } - sta32x_cache_sync(codec); + snd_soc_cache_sync(codec); } /* Power up to mute */ @@ -840,17 +790,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) STA32X_CxCFG_OM_MASK, 2 << STA32X_CxCFG_OM_SHIFT); - /* initialize coefficient shadow RAM with reset values */ - for (i = 4; i <= 49; i += 5) - sta32x->coef_shadow[i] = 0x400000; - for (i = 50; i <= 54; i++) - sta32x->coef_shadow[i] = 0x7fffff; - sta32x->coef_shadow[55] = 0x5a9df7; - sta32x->coef_shadow[56] = 0x7fffff; - sta32x->coef_shadow[59] = 0x7fffff; - sta32x->coef_shadow[60] = 0x400000; - sta32x->coef_shadow[61] = 0x400000; - sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); diff --git a/trunk/sound/soc/codecs/sta32x.h b/trunk/sound/soc/codecs/sta32x.h index d8e32a6262ee..b97ee5a75667 100644 --- a/trunk/sound/soc/codecs/sta32x.h +++ b/trunk/sound/soc/codecs/sta32x.h @@ -19,7 +19,6 @@ /* STA326 register addresses */ #define STA32X_REGISTER_COUNT 0x2d -#define STA32X_COEF_COUNT 62 #define STA32X_CONFA 0x00 #define STA32X_CONFB 0x01 diff --git a/trunk/sound/soc/codecs/wm8731.c b/trunk/sound/soc/codecs/wm8731.c index a7c9ae17fc7e..7e5ec03f6f8d 100644 --- a/trunk/sound/soc/codecs/wm8731.c +++ b/trunk/sound/soc/codecs/wm8731.c @@ -453,7 +453,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - codec->cache_sync = 1; break; } codec->dapm.bias_level = level; diff --git a/trunk/sound/soc/codecs/wm8753.c b/trunk/sound/soc/codecs/wm8753.c index 3a629d0d690e..a9504710bb69 100644 --- a/trunk/sound/soc/codecs/wm8753.c +++ b/trunk/sound/soc/codecs/wm8753.c @@ -190,9 +190,6 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; - if (wm8753->dai_func == ucontrol->value.integer.value[0]) - return 0; - if (codec->active) return -EBUSY; diff --git a/trunk/sound/soc/codecs/wm8962.c b/trunk/sound/soc/codecs/wm8962.c index 53edd9a8c758..91d3c6dbeba3 100644 --- a/trunk/sound/soc/codecs/wm8962.c +++ b/trunk/sound/soc/codecs/wm8962.c @@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0); static const unsigned int mixinpga_tlv[] = { - TLV_DB_RANGE_HEAD(5), + TLV_DB_RANGE_HEAD(7), 0, 1, TLV_DB_SCALE_ITEM(0, 600, 0), 2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0), 3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0), @@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0); static const unsigned int classd_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(7), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/trunk/sound/soc/codecs/wm8993.c b/trunk/sound/soc/codecs/wm8993.c index d1a142f48b09..eec8e1435116 100644 --- a/trunk/sound/soc/codecs/wm8993.c +++ b/trunk/sound/soc/codecs/wm8993.c @@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); static const unsigned int drc_max_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(4), 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), }; diff --git a/trunk/sound/soc/codecs/wm9081.c b/trunk/sound/soc/codecs/wm9081.c index 4a398c3bfe84..3cd35a02c28c 100644 --- a/trunk/sound/soc/codecs/wm9081.c +++ b/trunk/sound/soc/codecs/wm9081.c @@ -807,6 +807,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, mdelay(100); /* Normal bias enable & soft start off */ + reg |= WM9081_BIAS_ENA; reg &= ~WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -817,7 +818,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -829,15 +830,14 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Startup bias source and disable bias */ + /* Startup bias source */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC; - reg &= ~WM9081_BIAS_ENA; snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); - /* Disable VMID with soft ramping */ + /* Disable VMID and biases with soft ramping */ reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; + reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); reg |= WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); diff --git a/trunk/sound/soc/codecs/wm9090.c b/trunk/sound/soc/codecs/wm9090.c index f94c06057c64..2b5252c9e377 100644 --- a/trunk/sound/soc/codecs/wm9090.c +++ b/trunk/sound/soc/codecs/wm9090.c @@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) } static const unsigned int in_tlv[] = { - TLV_DB_RANGE_HEAD(3), + TLV_DB_RANGE_HEAD(6), 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0), 4, 6, TLV_DB_SCALE_ITEM(600, 600, 0), }; static const unsigned int mix_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(4), 0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0), 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0), }; static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(7), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/trunk/sound/soc/codecs/wm_hubs.c b/trunk/sound/soc/codecs/wm_hubs.c index 48e61e912400..84f33d4ea2cd 100644 --- a/trunk/sound/soc/codecs/wm_hubs.c +++ b/trunk/sound/soc/codecs/wm_hubs.c @@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(2), + TLV_DB_RANGE_HEAD(7), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; diff --git a/trunk/sound/soc/fsl/fsl_ssi.c b/trunk/sound/soc/fsl/fsl_ssi.c index 83c4bd5b2dd7..0268cf989736 100644 --- a/trunk/sound/soc/fsl/fsl_ssi.c +++ b/trunk/sound/soc/fsl/fsl_ssi.c @@ -694,7 +694,6 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; - sysfs_attr_init(&dev_attr->attr); dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show;