From 3cc7780b6fc04318ab08d84f739503989200cf55 Mon Sep 17 00:00:00 2001 From: Stefan Agner Date: Mon, 19 Oct 2015 17:42:23 -0700 Subject: ASoC: fsl_sai: fix Rx synchrounous mode When using the Rx clock for both, transmitter and receiver, the transmitter needs to be set to synchronous with receiver. This reverts 855675f6e6a6 ("ASoC: fsl_sai: Set SYNC bit of TCR2 to Asynchronous Mode"), which, judiging from the commit log, seems to mixed up between the two synchronous modes: The boolean sai->synchronous[TX] is indicating wheather the SAI should work in Rx synchronous mode (sync Tx with Rx), hence if the value is true, the SYNC field of TCR2 needs to be set to 0x1 ("Synchronous with receiver"). Signed-off-by: Stefan Agner Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92c4a85..1f0e5527a2fe 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -454,7 +454,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); -- cgit v1.2.3-58-ga151 From 341604ad839d10314af51669fd454dc0aa2ef288 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 3 Nov 2015 14:24:12 +0000 Subject: ASoC: arizona: fix range of OPCLK_REF The code was able to generate illegal OPCLK_REF values because the reference frequency tables listed all values of SYSCLK instead of valid values for OPCLK_REF clock. The maximum OPCLK_REF clock is 49.152MHz or 45.1584MHz. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8a2221ab3d10..586789597ecd 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -979,24 +979,18 @@ void arizona_init_dvfs(struct arizona_priv *priv) } EXPORT_SYMBOL_GPL(arizona_init_dvfs); -static unsigned int arizona_sysclk_48k_rates[] = { +static unsigned int arizona_opclk_ref_48k_rates[] = { 6144000, 12288000, 24576000, 49152000, - 73728000, - 98304000, - 147456000, }; -static unsigned int arizona_sysclk_44k1_rates[] = { +static unsigned int arizona_opclk_ref_44k1_rates[] = { 5644800, 11289600, 22579200, 45158400, - 67737600, - 90316800, - 135475200, }; static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, @@ -1021,11 +1015,11 @@ static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, } if (refclk % 8000) - rates = arizona_sysclk_44k1_rates; + rates = arizona_opclk_ref_44k1_rates; else - rates = arizona_sysclk_48k_rates; + rates = arizona_opclk_ref_48k_rates; - for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + for (ref = 0; ref < ARRAY_SIZE(arizona_opclk_ref_48k_rates) && rates[ref] <= refclk; ref++) { div = 1; while (rates[ref] / div >= freq && div < 32) { -- cgit v1.2.3-58-ga151 From 41a59cae585678136c28cdcbba9cb2faf27685f5 Mon Sep 17 00:00:00 2001 From: JongHo Kim Date: Tue, 3 Nov 2015 11:06:32 +0900 Subject: ASoC: wm8960: Fix the Input PGA Mute switch Change the xinvert value from 0 to 1 on the "Capture Switch" control WM8960 datasheet is shown as follows: Bit 7 at 00h and 01h register address 1 : Enable Mute, 0 : Disable Mute Signed-off-by: JongHo Kim Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c57411..bbe24275f8c9 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -223,7 +223,7 @@ SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, 6, 1, 0), SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", WM8960_INBMIX1, 4, 7, 0, boost_tlv), -- cgit v1.2.3-58-ga151 From 7099ee85e6af56828c46255f43adb15ed47e67df Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Thu, 5 Nov 2015 19:55:51 +0800 Subject: ASoC: rt5645: Power up the RC clock to make sure the speaker volume adjust properly Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 38 +++++++++++++++++++++++++++++++++++--- 1 file changed, 35 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 28132375e427..672fafd8314a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -245,7 +245,7 @@ struct rt5645_priv { struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; struct snd_soc_jack *btn_jack; - struct delayed_work jack_detect_work; + struct delayed_work jack_detect_work, rcclock_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; struct rt5645_eq_param_s *eq_param; @@ -565,12 +565,33 @@ static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, .put = rt5645_hweq_put \ } +static int rt5645_spk_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret; + + cancel_delayed_work_sync(&rt5645->rcclock_work); + + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PU); + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + queue_delayed_work(system_power_efficient_wq, &rt5645->rcclock_work, + msecs_to_jiffies(200)); + + return ret; +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), - SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL, - RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + SOC_DOUBLE_EXT_TLV("Speaker Playback Volume", RT5645_SPK_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, snd_soc_get_volsw, + rt5645_spk_put_volsw, out_vol_tlv), /* ClassD modulator Speaker Gain Ratio */ SOC_SINGLE_TLV("Speaker ClassD Playback Volume", RT5645_SPO_CLSD_RATIO, @@ -3122,6 +3143,15 @@ static void rt5645_jack_detect_work(struct work_struct *work) SND_JACK_BTN_2 | SND_JACK_BTN_3); } +static void rt5645_rcclock_work(struct work_struct *work) +{ + struct rt5645_priv *rt5645 = + container_of(work, struct rt5645_priv, rcclock_work.work); + + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_PWR_CLK25M_MASK, RT5645_PWR_CLK25M_PD); +} + static irqreturn_t rt5645_irq(int irq, void *data) { struct rt5645_priv *rt5645 = data; @@ -3587,6 +3617,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); + INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, @@ -3621,6 +3652,7 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) free_irq(i2c->irq, rt5645); cancel_delayed_work_sync(&rt5645->jack_detect_work); + cancel_delayed_work_sync(&rt5645->rcclock_work); snd_soc_unregister_codec(&i2c->dev); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); -- cgit v1.2.3-58-ga151 From 474d147ad1ecc98e50a65c9f350fadfcc37a8bb4 Mon Sep 17 00:00:00 2001 From: Adam Sampson Date: Tue, 27 Oct 2015 21:00:45 +0000 Subject: ASoC: sun4i-codec: use consistent names for PA controls The power amplifier for the headphone output is called "the PA" and "the headphone amplifier" in Allwinner's documentation for the A10 and A20. sun4i-codec calls it "PA" in some places and "Pre-Amplifier" (which isn't really accurate) in others, leading to user-visible controls with different names referring to the same device. When this driver implements audio input, it'll also need to expose controls for the line and mic input preamps, so just referring to "the Pre-Amplifier" will be ambiguous. Change it to use "Power Amplifier" consistently for the power amplifier's controls. Signed-off-by: Adam Sampson Acked-by: Maxime Ripard Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-codec.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da168b6..1bb896d78d09 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -2,6 +2,7 @@ * Copyright 2014 Emilio López * Copyright 2014 Jon Smirl * Copyright 2015 Maxime Ripard + * Copyright 2015 Adam Sampson * * Based on the Allwinner SDK driver, released under the GPL. * @@ -404,7 +405,7 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); static const struct snd_kcontrol_new sun4i_codec_widgets[] = { - SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, + SOC_SINGLE_TLV("Power Amplifier Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), }; @@ -452,12 +453,12 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), - /* Pre-Amplifier */ - SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + /* Power Amplifier */ + SND_SOC_DAPM_MIXER("Power Amplifier", SUN4I_CODEC_ADC_ACTL, SUN4I_CODEC_ADC_ACTL_PA_EN, 0, sun4i_codec_pa_mixer_controls, ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), - SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("Power Amplifier Mute", SND_SOC_NOPM, 0, 0, &sun4i_codec_pa_mute), SND_SOC_DAPM_OUTPUT("HP Right"), @@ -480,16 +481,16 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Left Mixer", NULL, "Mixer Enable" }, { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, - /* Pre-Amplifier Mixer Routes */ - { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, - { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, - { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, - { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + /* Power Amplifier Routes */ + { "Power Amplifier", "Mixer Playback Switch", "Left Mixer" }, + { "Power Amplifier", "Mixer Playback Switch", "Right Mixer" }, + { "Power Amplifier", "DAC Playback Switch", "Left DAC" }, + { "Power Amplifier", "DAC Playback Switch", "Right DAC" }, - /* PA -> HP path */ - { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, - { "HP Right", NULL, "Pre-Amplifier Mute" }, - { "HP Left", NULL, "Pre-Amplifier Mute" }, + /* Headphone Output Routes */ + { "Power Amplifier Mute", "Switch", "Power Amplifier" }, + { "HP Right", NULL, "Power Amplifier Mute" }, + { "HP Left", NULL, "Power Amplifier Mute" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { -- cgit v1.2.3-58-ga151 From 021c5d9469960b8c68aa1d1825f7bfd8d61e157d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 5 Nov 2015 23:53:03 +0000 Subject: ASoC: rsnd: fixup SCU_SYS_INT_EN1 address cfcefe0126 ("ASoC: rsnd: add recovery support for under/over flow error on SRC") added SCU_SYS_INT_EN1 address, but it should be 0x1d4, not 0x1c4. This patch fixup it. Fixes: cfcefe0126 ("ASoC: rsnd: add recovery support for under/over flow error on SRC") Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/sh/rcar/gen.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index f04d17bc6e3d..916b38d54fda 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -231,7 +231,7 @@ static int rsnd_gen2_probe(struct platform_device *pdev, RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc), RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0), - RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1c4), + RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4), RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), -- cgit v1.2.3-58-ga151 From 1bdd593247ee5a74eb58828a4cf18bdc8a5f1baa Mon Sep 17 00:00:00 2001 From: Andreas Dannenberg Date: Mon, 9 Nov 2015 12:19:19 -0600 Subject: ASoC: davinci-mcasp: Fix TDM slot rx/tx mask associations Fixes the associations between the tx_mask and rx_mask and the associated playback / capture streams during setting of the TDM slot. With this patch in place it is now possible for example to only populate tx_mask (leaving rx_mask as 0) for output-only codecs to control the TDM slot(s) the McASP serial port uses for transmit. Before that, this scenario would incorrectly rely on the rx_mask for this. Signed-off-by: Andreas Dannenberg Reviewed-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4495a40a9468..caa0bebcd7f4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -681,8 +681,8 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, } mcasp->tdm_slots = slots; - mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = rx_mask; - mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = tx_mask; + mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; + mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; mcasp->slot_width = slot_width; return davinci_mcasp_set_ch_constraints(mcasp); -- cgit v1.2.3-58-ga151 From fd589a1be20fdd76ef97700dd0185e7a060546dc Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 10 Nov 2015 18:12:42 +0200 Subject: ASoC: dapm: Reset dapm wcache after freeing damp widgets If there is anything in damp->path_source_cache or damp->path_sink_cache, it can not be valid after the widgets have been freed. Without this patch a repeated remove and load of a machine driver may cause NULL pointer reference in dapm_wcache_lookup() when a freed widget, not belonging to any list, is haunting in the wcache. Signed-off-by: Jyri Sarha Reported-by: Felipe Balbi Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 7 +++++++ sound/soc/soc-topology.c | 1 + 3 files changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 7855cfe46b69..95a937eafb79 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -398,6 +398,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w); +void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm); /* dapm events */ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 016eba10b1ec..7d009428934a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2293,6 +2293,12 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) kfree(w); } +void snd_soc_dapm_reset_cache(struct snd_soc_dapm_context *dapm) +{ + dapm->path_sink_cache.widget = NULL; + dapm->path_source_cache.widget = NULL; +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2303,6 +2309,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) continue; snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } static struct snd_soc_dapm_widget *dapm_find_widget( diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 8d7ec80af51b..cce63fe65dd9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1805,6 +1805,7 @@ void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, snd_soc_tplg_widget_remove(w); snd_soc_dapm_free_widget(w); } + snd_soc_dapm_reset_cache(dapm); } EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); -- cgit v1.2.3-58-ga151 From 2f64b6ed44c26eeb3d1bf5428936629cf552eda7 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 10 Nov 2015 14:40:55 +0800 Subject: ASoC: rl6231: avoid using divisible by 3 for DMIC clk Few codecs will meet no DMIC clock output issue when select a divided number which is divisible by 3. To prevent this issue, the patch ignore the numbers when calculating the DMIC clock divider. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rl6231.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index aca479fa7670..18b42925314e 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -80,6 +80,8 @@ int rl6231_calc_dmic_clk(int rate) } for (i = 0; i < ARRAY_SIZE(div); i++) { + if ((div[i] % 3) == 0) + continue; /* find divider that gives DMIC frequency below 3MHz */ if (3000000 * div[i] >= rate) return i; -- cgit v1.2.3-58-ga151 From c22d7666c5c4473cfffe8c40fcf86bd6e16317df Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Mon, 9 Nov 2015 18:01:04 +0800 Subject: ASoC: rt5677: Avoid the pop sound that comes from the filter power The patch changes the type of DACs mixer to AUTODISABLE and add the delay time after power up to avoid the pop sound that comes from the filter power. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 100 ++++++++++++++++++++++++++++------------------ 1 file changed, 61 insertions(+), 39 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index b4cd7e3bf5f8..69d987a9935c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1386,90 +1386,90 @@ static const struct snd_kcontrol_new rt5677_dac_r_mix[] = { }; static const struct snd_kcontrol_new rt5677_sto1_dac_l_mix[] = { - SOC_DAPM_SINGLE("ST L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_ST_DAC1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_L_STO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC2_L_STO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_R_STO_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_sto1_dac_r_mix[] = { - SOC_DAPM_SINGLE("ST R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_ST_DAC1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_R_STO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC2_R_STO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_STO1_DAC_MIXER, RT5677_M_DAC1_L_STO_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_mono_dac_l_mix[] = { - SOC_DAPM_SINGLE("ST L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_ST_DAC2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC1_L_MONO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_L_MONO_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_R_MONO_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_mono_dac_r_mix[] = { - SOC_DAPM_SINGLE("ST R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("ST R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_ST_DAC2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC1 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC1_R_MONO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 R Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_R_MONO_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC2 L Switch", RT5677_MONO_DAC_MIXER, RT5677_M_DAC2_L_MONO_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd1_l_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD1_MIXER, RT5677_M_STO_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD1_MIXER, RT5677_M_MONO_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_L_DD1_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_R_DD1_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd1_r_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD1_MIXER, RT5677_M_STO_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD1_MIXER, RT5677_M_MONO_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 R Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 R Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_R_DD1_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC3 L Switch", RT5677_DD1_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC3 L Switch", RT5677_DD1_MIXER, RT5677_M_DAC3_L_DD1_R_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd2_l_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix L Switch", RT5677_DD2_MIXER, RT5677_M_STO_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix L Switch", RT5677_DD2_MIXER, RT5677_M_MONO_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_L_DD2_L_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_R_DD2_L_SFT, 1, 1), }; static const struct snd_kcontrol_new rt5677_dd2_r_mix[] = { - SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Sto DAC Mix R Switch", RT5677_DD2_MIXER, RT5677_M_STO_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("Mono DAC Mix R Switch", RT5677_DD2_MIXER, RT5677_M_MONO_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 R Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 R Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_R_DD2_R_SFT, 1, 1), - SOC_DAPM_SINGLE("DAC4 L Switch", RT5677_DD2_MIXER, + SOC_DAPM_SINGLE_AUTODISABLE("DAC4 L Switch", RT5677_DD2_MIXER, RT5677_M_DAC4_L_DD2_R_SFT, 1, 1), }; @@ -2596,6 +2596,21 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5677_filter_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(50); + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT, 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU | @@ -3072,19 +3087,26 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), + RT5677_PWR_DAC_S1F_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M2F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M2F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M3F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M3F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M4F_L_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, - RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), + RT5677_PWR_DAC_M4F_R_BIT, 0, rt5677_filter_power_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), -- cgit v1.2.3-58-ga151 From 7115cb913d9e2d68583cf76578b32568bc8ea83f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 10 Nov 2015 07:39:17 +0000 Subject: ASoC: rsnd: make sure SRC In Rate feature enablement SRC In Rate convert feature cannot be used if data path is using DVC. This patch judges it, and not allowed to use it in such case. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 261b50217c48..68b439ed22d7 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -923,6 +923,7 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod, struct snd_soc_pcm_runtime *rtd) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_mod *dvc = rsnd_io_to_mod_dvc(io); struct rsnd_src *src = rsnd_mod_to_src(mod); int ret; @@ -936,6 +937,12 @@ static int rsnd_src_pcm_new_gen2(struct rsnd_mod *mod, if (!rsnd_rdai_is_clk_master(rdai)) return 0; + /* + * SRC In doesn't work if DVC was enabled + */ + if (dvc && !rsnd_io_is_play(io)) + return 0; + /* * enable sync convert */ -- cgit v1.2.3-58-ga151 From 7336dcefac4d8f94fa205a668138a6462841acc4 Mon Sep 17 00:00:00 2001 From: John Lin Date: Mon, 16 Nov 2015 14:41:07 +0800 Subject: ASoC: rl6231: fix range of DMIC clock The maximum DMIC clock rate is 3.072 MHz for most DMIC. And it will get better performance in higher clock rate. If we set maximum to 3 MHz in driver, we will get a clock rate which is not even close to 3 MHz. For example, if DMIC clock source is 24.576 MHz, the DMIC clock will be about 1.5 MHz in current code. But it will be 3.072 MHz with this patch. Signed-off-by: John Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rl6231.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 18b42925314e..1dc68ab08a17 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -82,8 +82,8 @@ int rl6231_calc_dmic_clk(int rate) for (i = 0; i < ARRAY_SIZE(div); i++) { if ((div[i] % 3) == 0) continue; - /* find divider that gives DMIC frequency below 3MHz */ - if (3000000 * div[i] >= rate) + /* find divider that gives DMIC frequency below 3.072MHz */ + if (3072000 * div[i] >= rate) return i; } -- cgit v1.2.3-58-ga151 From 91ed37e45c485533997e8a7c1efd2ca39b441b60 Mon Sep 17 00:00:00 2001 From: John Lin Date: Mon, 16 Nov 2015 13:55:35 +0800 Subject: ASoC: rt5645: Increase the delay time to imporve the HP pop noise Unmuting headphone has pop noise in particular hardware design. So we extend the delay time in headphone unmuting sequence to avoid pop. Signed-off-by: John Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 672fafd8314a..fa8b5dfa673e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1519,7 +1519,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on) regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); - msleep(40); + msleep(70); rt5645->hp_on = true; } else { /* depop parameters */ -- cgit v1.2.3-58-ga151 From 0e18d457b31e98e68f6918e41c85ad3b736c4789 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Fri, 13 Nov 2015 18:15:56 +0100 Subject: ASoC: fix rockchip 64-bit build warning The rk_spdif_probe uses the device match data as a token to identify a particular device, but accidentally casts a pointer to 'int', which is not portable, as gcc points out in this warning on arm64: rockchip_spdif.c: In function 'rk_spdif_probe': rockchip_spdif.c:283:6: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] This changes the logic to compare two pointer values instead, using the same cast that was used for initializing the value in the first place. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index a38a3029062c..ac72ff5055bb 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -280,7 +280,7 @@ static int rk_spdif_probe(struct platform_device *pdev) int ret; match = of_match_node(rk_spdif_match, np); - if ((int) match->data == RK_SPDIF_RK3288) { + if (match->data == (void *)RK_SPDIF_RK3288) { struct regmap *grf; grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); -- cgit v1.2.3-58-ga151 From e9f96bc53c1b959859599cb30ce6fd4fbb4448c2 Mon Sep 17 00:00:00 2001 From: Sachin Pandhare Date: Tue, 10 Nov 2015 23:38:02 +0530 Subject: ASoC: wm8962: correct addresses for HPF_C_0/1 From datasheet: R17408 (4400h) HPF_C_1 R17409 (4401h) HPF_C_0 17048 -> 17408 (0x4400) 17049 -> 17409 (0x4401) Signed-off-by: Sachin Pandhare Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 39ebd7bf4f53..a7e79784fc16 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -365,8 +365,8 @@ static const struct reg_default wm8962_reg[] = { { 16924, 0x0059 }, /* R16924 - HDBASS_PG_1 */ { 16925, 0x999A }, /* R16925 - HDBASS_PG_0 */ - { 17048, 0x0083 }, /* R17408 - HPF_C_1 */ - { 17049, 0x98AD }, /* R17409 - HPF_C_0 */ + { 17408, 0x0083 }, /* R17408 - HPF_C_1 */ + { 17409, 0x98AD }, /* R17409 - HPF_C_0 */ { 17920, 0x007F }, /* R17920 - ADCL_RETUNE_C1_1 */ { 17921, 0xFFFF }, /* R17921 - ADCL_RETUNE_C1_0 */ -- cgit v1.2.3-58-ga151 From 0580bcc91d0aee7367c001955234d71b0b337b41 Mon Sep 17 00:00:00 2001 From: John Lin Date: Wed, 11 Nov 2015 15:25:28 +0800 Subject: ASoC: rt5645: Add struct dmi_system_id "Google Edgar" for Chrome OS Add platform specific data for Edgar project. Signed-off-by: John Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index fa8b5dfa673e..647b594ad04e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3378,6 +3378,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Reks"), }, }, + { + .ident = "Google Edgar", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"), + }, + }, { } }; -- cgit v1.2.3-58-ga151 From 4454a8378be5809c2b830531bb4c4712b5e46bef Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Mon, 9 Nov 2015 12:56:00 -0800 Subject: ASoC: nau8825: add pm function This patch adds pm function and fixes following issues 1.i2c timeout after resume, after resume we saw interrupt handler is called prior to i2c controller is resumed.This causes i2c timeout 2.no audio after resume Signed-off-by: Fang, Yang A Signed-off-by: Yong Zhi Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7fc7b4e3f444..c1b87c5800b1 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1271,6 +1271,36 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_suspend(struct device *dev) +{ + struct i2c_client *client = to_i2c_client(dev); + struct nau8825 *nau8825 = dev_get_drvdata(dev); + + disable_irq(client->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct device *dev) +{ + struct i2c_client *client = to_i2c_client(dev); + struct nau8825 *nau8825 = dev_get_drvdata(dev); + + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + enable_irq(client->irq); + + return 0; +} +#endif + +static const struct dev_pm_ops nau8825_pm = { + SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) +}; + static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1297,6 +1327,7 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, -- cgit v1.2.3-58-ga151 From e71bf05554c9015bef8df3ffc386ccb37b153858 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Nov 2015 16:30:17 +0800 Subject: ASoC: rt5670: fix wrong bit def for pll src The bit allocation for PLL source is 0x80 [13:11] instead of [12:11] Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index dc2b46236c5c..3f1b0f1df809 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -973,12 +973,12 @@ #define RT5670_SCLK_SRC_MCLK (0x0 << 14) #define RT5670_SCLK_SRC_PLL1 (0x1 << 14) #define RT5670_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ -#define RT5670_PLL1_SRC_MASK (0x3 << 12) -#define RT5670_PLL1_SRC_SFT 12 -#define RT5670_PLL1_SRC_MCLK (0x0 << 12) -#define RT5670_PLL1_SRC_BCLK1 (0x1 << 12) -#define RT5670_PLL1_SRC_BCLK2 (0x2 << 12) -#define RT5670_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5670_PLL1_SRC_MASK (0x7 << 11) +#define RT5670_PLL1_SRC_SFT 11 +#define RT5670_PLL1_SRC_MCLK (0x0 << 11) +#define RT5670_PLL1_SRC_BCLK1 (0x1 << 11) +#define RT5670_PLL1_SRC_BCLK2 (0x2 << 11) +#define RT5670_PLL1_SRC_BCLK3 (0x3 << 11) #define RT5670_PLL1_PD_MASK (0x1 << 3) #define RT5670_PLL1_PD_SFT 3 #define RT5670_PLL1_PD_1 (0x0 << 3) -- cgit v1.2.3-58-ga151 From f4be978b9611c94f20cdc8cee540ef1a52f8875c Mon Sep 17 00:00:00 2001 From: Omair M Abdullah Date: Mon, 9 Nov 2015 23:20:01 +0530 Subject: ASoC: topology: fix info callback for TLV byte control topology core used wrong callback for TLV bytes control, it should be snd_soc_bytes_info_ext and not snd_soc_bytes_info Signed-off-by: Omair M Abdullah Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 8d7ec80af51b..50f21ed00cfa 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -531,7 +531,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, /* TLV bytes controls need standard kcontrol info handler, * TLV callback and extended put/get handlers. */ - k->info = snd_soc_bytes_info; + k->info = snd_soc_bytes_info_ext; k->tlv.c = snd_soc_bytes_tlv_callback; ext_ops = tplg->bytes_ext_ops; -- cgit v1.2.3-58-ga151 From 4b6295b238cf0fe0841675816be0998345d5990a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 18 Nov 2015 19:11:46 +0530 Subject: ASoC: Intel: Skylake: Add I2C depends for SKL machine The i2c is dependency for the i2c codec drivers, so machine should depend on i2c. WIthout this we get build failures if I2C is not selected sound/soc/codecs/rl6347a.c: In function 'rl6347a_hw_write': >> sound/soc/codecs/rl6347a.c:66:8: error: implicit declaration of function >> 'i2c_master_send' [-Werror=implicit-function-declaration] ret = i2c_master_send(client, data, 4); ^ sound/soc/codecs/rl6347a.c: In function 'rl6347a_hw_read': >> sound/soc/codecs/rl6347a.c:114:8: error: implicit declaration of function >> 'i2c_transfer' [-Werror=implicit-function-declaration] ret = i2c_transfer(client->adapter, xfer, 2); Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7b778ab85f8b..d430ef5a4f38 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -144,7 +144,7 @@ config SND_SOC_INTEL_SKYLAKE config SND_SOC_INTEL_SKL_RT286_MACH tristate "ASoC Audio driver for SKL with RT286 I2S mode" - depends on X86 && ACPI + depends on X86 && ACPI && I2C select SND_SOC_INTEL_SST select SND_SOC_INTEL_SKYLAKE select SND_SOC_RT286 -- cgit v1.2.3-58-ga151 From cd3ed08a86e8b5022f107aa72a1929b6417c1f42 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 19 Nov 2015 14:54:07 +0100 Subject: ASoC: sti: remove wrong error message Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_reader.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index f791239a3087..819eeafdf6b4 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -346,7 +346,6 @@ int uni_reader_init(struct platform_device *pdev, reader->hw = &uni_reader_pcm_hw; reader->dai_ops = &uni_reader_dai_ops; - dev_err(reader->dev, "%s: enter\n", __func__); ret = uni_reader_parse_dt(pdev, reader); if (ret < 0) { dev_err(reader->dev, "Failed to parse DeviceTree"); -- cgit v1.2.3-58-ga151 From f9f51973d3a8559731a228e91ac29792b43046a5 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 19 Nov 2015 14:54:08 +0100 Subject: ASoC: sti: rename ST proprietary DT properties "st," prefix has been added for ST proprietary DT properties. Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_player.c | 6 +++--- sound/soc/sti/uniperif_reader.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 843f037a317d..1e19a7c6b7e8 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,7 +989,7 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - if (of_property_read_u32(pnode, "version", &player->ver) || + if (of_property_read_u32(pnode, "st,version", &player->ver) || player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; @@ -998,13 +998,13 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + if (of_property_read_u32(pnode, "st,uniperiph-id", &info->id)) { dev_err(dev, "uniperipheral id not defined"); return -EINVAL; } /* Read the device mode property */ - if (of_property_read_string(pnode, "mode", &mode)) { + if (of_property_read_string(pnode, "st,mode", &mode)) { dev_err(dev, "uniperipheral mode not defined"); return -EINVAL; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 819eeafdf6b4..8a0eb2050169 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - if (of_property_read_u32(node, "version", &reader->ver) || + if (of_property_read_u32(node, "st,version", &reader->ver) || reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(&pdev->dev, "Unknown uniperipheral version "); return -EINVAL; -- cgit v1.2.3-58-ga151 From 36a65e2072625556191c6c616d65ed4f67f4f0d0 Mon Sep 17 00:00:00 2001 From: Moise Gergaud Date: Thu, 19 Nov 2015 14:54:09 +0100 Subject: ASoC: sti: set player private data Set substream player private data. substream player private data is used in uni_player_irq_handler to lock, stop & unlock the stream when interrupt indicates underflow/overflow. If not set, then segmentation fault occurs. Signed-off-by: Moise Gergaud Acked-by: Arnaud Pouliquen Signed-off-by: Mark Brown --- sound/soc/sti/uniperif_player.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 1e19a7c6b7e8..5c2bc53f0a9b 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -669,6 +669,7 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + player->substream = substream; player->clk_adj = 0; @@ -950,6 +951,8 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, if (player->state != UNIPERIF_STATE_STOPPED) /* Stop the player */ uni_player_stop(player); + + player->substream = NULL; } static int uni_player_parse_dt_clk_glue(struct platform_device *pdev, -- cgit v1.2.3-58-ga151 From 3f58b7039c70f1d0a19157c7bf97ef69d445565f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 20 Nov 2015 12:25:59 +0800 Subject: ASoC: rt5645: Add dmi_system_id "Google Wizpig" Add platform specific data for Wizpig project. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 647b594ad04e..5af90234d453 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3385,6 +3385,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Edgar"), }, }, + { + .ident = "Google Wizpig", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"), + }, + }, { } }; -- cgit v1.2.3-58-ga151 From 113b0b20fc123bc522ded68ea710d789b0415ebe Mon Sep 17 00:00:00 2001 From: John Keeping Date: Fri, 20 Nov 2015 11:42:22 +0000 Subject: ASoC: es8328: Fix shifts for mixer switches These are all off by one; the playback and bypass switches are the top two bits of the registers, which are at shifts 7 and 6 not 8 and 7. Signed-off-by: John Keeping Signed-off-by: Mark Brown --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337dc17c..84f5eb07a91b 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -205,18 +205,18 @@ static const struct snd_kcontrol_new es8328_right_line_controls = /* Left Mixer */ static const struct snd_kcontrol_new es8328_left_mixer_controls[] = { - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0), - SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 6, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 6, 1, 0), }; /* Right Mixer */ static const struct snd_kcontrol_new es8328_right_mixer_controls[] = { - SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0), - SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0), - SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0), - SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 7, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 6, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 7, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 6, 1, 0), }; static const char * const es8328_pga_sel[] = { -- cgit v1.2.3-58-ga151 From 0ad7d3a04b2a1a43fa71eb89f754527b082213ad Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 23 Nov 2015 12:51:53 +0200 Subject: ASoC: davinci-mcasp: Fix master capture only mode When McASP is used as TX/RX synchronous (TX side generating clocks for RX side also) and only capture is used we need to configure the number of TX slots in order McASP to be able to generate the Frame sync. Fixes: 9273de1940d9e ("ASoC: davinci-mcasp: Add set_tdm_slots() support") Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index caa0bebcd7f4..c1c9c2e3525b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -908,6 +908,14 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRMOD(total_slots), FSRMOD(0x1FF)); + /* + * If McASP is set to be TX/RX synchronous and the playback is + * not running already we need to configure the TX slots in + * order to have correct FSX on the bus + */ + if (mcasp_is_synchronous(mcasp) && !mcasp->channels) + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(total_slots), FSXMOD(0x1FF)); } return 0; -- cgit v1.2.3-58-ga151 From 87b5ed8ecb9fe05a696e1c0b53c7a49ea66432c1 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee Date: Mon, 23 Nov 2015 16:38:48 +0530 Subject: ASoC: Intel: Skylake: fix memory leak We have requested the firmware but missed releasing it. Signed-off-by: Sudip Mukherjee Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a7854c8fc523..ffea427aeca8 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1240,6 +1240,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) */ ret = snd_soc_tplg_component_load(&platform->component, &skl_tplg_ops, fw, 0); + release_firmware(fw); if (ret < 0) { dev_err(bus->dev, "tplg component load failed%d\n", ret); return -EINVAL; -- cgit v1.2.3-58-ga151 From ab07eaedb7ada83cc6341894dff9cd54f1af7f8b Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 24 Nov 2015 23:21:10 +0100 Subject: ASoC: fsl: clarify ac97 dependency A new randconfig build failure shows that the fsl-asoc-card module must not be built-in when the AC97 driver is a loadable module: sound/built-in.o: In function `fsl_asoc_card_late_probe': :(.text+0x571d8): undefined reference to `snd_ac97_update_bits' I couldn't come up with a nice solution, so this adds another dependency on "X || !X", which is the Kconfig way of saying that we have an optional dependency on something that might be a loadable module. Fixes: 50760cad9de9 ("ASoC: fsl-asoc-card: add AC'97 support") Signed-off-by: Arnd Bergmann Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19c302b0d763..14dfdee05fd5 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -283,6 +283,8 @@ config SND_SOC_IMX_MC13783 config SND_SOC_FSL_ASOC_CARD tristate "Generic ASoC Sound Card with ASRC support" depends on OF && I2C + # enforce SND_SOC_FSL_ASOC_CARD=m if SND_AC97_CODEC=m: + depends on SND_AC97_CODEC || SND_AC97_CODEC=n select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_ESAI -- cgit v1.2.3-58-ga151 From 18a9d7486ad28d68920128720514f9555a4c1869 Mon Sep 17 00:00:00 2001 From: Sjoerd Simons Date: Wed, 25 Nov 2015 09:54:11 +0100 Subject: ASoC: rockchip: Fix incorrect VDW value for 24 bit Correct valid data word register value for 24 bit data width. The bit value should be 10 (aka 0x2), not 0x10. This fixes playback of 24 bit audio. Signed-off-by: Sjoerd Simons Reviewed-by: Caesar Wang Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_spdif.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 07f86a21046a..921b4095fb92 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -28,9 +28,9 @@ #define SPDIF_CFGR_VDW(x) (x << SPDIF_CFGR_VDW_SHIFT) #define SDPIF_CFGR_VDW_MASK (0xf << SPDIF_CFGR_VDW_SHIFT) -#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x00) -#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x01) -#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x10) +#define SPDIF_CFGR_VDW_16 SPDIF_CFGR_VDW(0x0) +#define SPDIF_CFGR_VDW_20 SPDIF_CFGR_VDW(0x1) +#define SPDIF_CFGR_VDW_24 SPDIF_CFGR_VDW(0x2) /* * DMACR -- cgit v1.2.3-58-ga151 From 6b3cecd11539178978e1f54fe1363c39fe0db045 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 24 Nov 2015 10:55:29 +0800 Subject: ASoC: rt5645: Add dmi_system_id "Google Terra" Add platform specific data for Terra project. Signed-off-by: Luke_Yin@asus.com Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 5af90234d453..ef76940f9dcb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3392,6 +3392,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Wizpig"), }, }, + { + .ident = "Google Terra", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Terra"), + }, + }, { } }; -- cgit v1.2.3-58-ga151 From 9a11ef7ff00e08825ac970a6bda56a3ea8ab0321 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 23 Nov 2015 17:37:54 -0800 Subject: ASoC: fix kernel-doc warnings in sound/soc/soc-ops.c Fix kernel-doc warnings in soc-ops.c: ..//sound/soc/soc-ops.c:415: warning: No description found for parameter 'ucontrol' ..//sound/soc/soc-ops.c:415: warning: Excess function parameter 'uinfo' description in 'snd_soc_put_volsw_sx' Signed-off-by: Randy Dunlap Cc: Liam Girdwood Cc: Mark Brown Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index ecd38e52285a..2f67ba6d7a8f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -404,7 +404,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * -- cgit v1.2.3-58-ga151 From 1a7aaa58ec7aaa389cd6b200809908ec472d316b Mon Sep 17 00:00:00 2001 From: Jeeja KP Date: Mon, 23 Nov 2015 21:22:31 +0530 Subject: ASoC: core: Change power state before rechecking endpoint For DAPM resume, we should first change the power state of the card and then recheck the endpoints. This ensures the dapm is resumed first and then userspace can resume the streams. Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 24b096066a07..a1305f827a98 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -795,12 +795,12 @@ static void soc_resume_deferred(struct work_struct *work) dev_dbg(card->dev, "ASoC: resume work completed\n"); - /* userspace can access us now we are back as we were before */ - snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); - /* Recheck all endpoints too, their state is affected by suspend */ dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); + + /* userspace can access us now we are back as we were before */ + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); } /* powers up audio subsystem after a suspend */ -- cgit v1.2.3-58-ga151