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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 From 86f799b82f5c011404ddef54600bc5e99b7e0cf2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 14 Nov 2015 17:46:31 +0100 Subject: ALSA: hda - Fix noise on Dell Latitude E6440 Dell Latitude E6440 (1028:05bd) needs the same fixup as applied to other Latitude E7xxx models for the click noise due to the recent power-saving changes. Bugzilla: http://bugzilla.opensuse.org/show_bug.cgi?id=954876 Cc: # v4.1+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f7b065f9ac4..081ef9bdf0df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5182,6 +5182,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), + SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), -- cgit v1.2.3-70-g09d2 From 5d5563b14fe34021b690eb3edc54abcc876e417c Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 14 Nov 2015 16:42:04 +0900 Subject: ALSA: dice: fix detection of Loud devices Commit a471fcde8c2c ("ALSA: dice: fix detection of Weiss devices") adds a quirk of Weiss models. According to users' reports, Loud models also have the similar quirk. They have 0x10 in the category field. This commit adds support for Mackie Onyx Blackbird and Onyx-i series. As long as I know, Dice-based models produced by Focusrite/Alesis/PreSonus/M-Audio/TC Electronic have default value (0x04) in their category field, thus it may be reasonable to add a condition statement for Loud models, instead of removing the check of category value. Reported-by: Rouge Etienne Reported-by: Etilem Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/firewire/dice/dice.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 5d99436dfcae..0cda05c72f50 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -12,9 +12,11 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); #define OUI_WEISS 0x001c6a +#define OUI_LOUD 0x000ff2 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 +#define LOUD_CATEGORY_ID 0x10 static int dice_interface_check(struct fw_unit *unit) { @@ -57,6 +59,8 @@ static int dice_interface_check(struct fw_unit *unit) } if (vendor == OUI_WEISS) category = WEISS_CATEGORY_ID; + else if (vendor == OUI_LOUD) + category = LOUD_CATEGORY_ID; else category = DICE_CATEGORY_ID; if (device->config_rom[3] != ((vendor << 8) | category) || -- cgit v1.2.3-70-g09d2 From 98d362becb6621bebdda7ed0eac7ad7ec6c37898 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 15 Nov 2015 22:37:44 +0100 Subject: ALSA: usb-audio: add packet size quirk for the Medeli DD305 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 7661616f3636..68e9f9a83fde 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1341,6 +1341,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, * Various chips declare a packet size larger than 4 bytes, but * do not actually work with larger packets: */ + case USB_ID(0x0a67, 0x5011): /* Medeli DD305 */ case USB_ID(0x0a92, 0x1020): /* ESI M4U */ case USB_ID(0x1430, 0x474b): /* RedOctane GH MIDI INTERFACE */ case USB_ID(0x15ca, 0x0101): /* Textech USB Midi Cable */ -- cgit v1.2.3-70-g09d2 From 1ca8b201309d842642f221db7f02f71c0af5be2d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 15 Nov 2015 22:38:29 +0100 Subject: ALSA: usb-audio: prevent CH345 multiport output SysEx corruption The CH345 USB MIDI chip has two output ports. However, they are multiplexed through one pin, and the number of ports cannot be reduced even for hardware that implements only one connector, so for those devices, data sent to either port ends up on the same hardware output. This becomes a problem when both ports are used at the same time, as longer MIDI commands (such as SysEx messages) are likely to be interrupted by messages from the other port, and thus to get lost. It would not be possible for the driver to detect how many ports the device actually has, except that in practice, _all_ devices built with the CH345 have only one port. So we can just ignore the device's descriptors, and hardcode one output port. Signed-off-by: Clemens Ladisch Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 3 +++ sound/usb/quirks-table.h | 11 +++++++++++ sound/usb/quirks.c | 1 + sound/usb/usbaudio.h | 1 + 4 files changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 68e9f9a83fde..010094abf752 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2377,6 +2377,9 @@ int snd_usbmidi_create(struct snd_card *card, if (err < 0) break; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + break; + case QUIRK_MIDI_CH345: err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; default: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1a1e2e4df35e..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2829,6 +2829,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), .idProduct = 0x1020, }, +/* QinHeng devices */ +{ + USB_DEVICE(0x1a86, 0x752d), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "QinHeng", + .product_name = "CH345", + .ifnum = 1, + .type = QUIRK_MIDI_CH345 + } +}, + /* KeithMcMillen Stringport */ { USB_DEVICE(0x1f38, 0x0001), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5ca80e7d30cd..7016ad898187 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -538,6 +538,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_MIDI_FTDI] = create_any_midi_quirk, + [QUIRK_MIDI_CH345] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 15a12715bd05..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -95,6 +95,7 @@ enum quirk_type { QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_MIDI_FTDI, + QUIRK_MIDI_CH345, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UAXX, -- cgit v1.2.3-70-g09d2 From a91e627e3f0ed820b11d86cdc04df38f65f33a70 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 15 Nov 2015 22:39:08 +0100 Subject: ALSA: usb-audio: work around CH345 input SysEx corruption One of the many faults of the QinHeng CH345 USB MIDI interface chip is that it does not handle received SysEx messages correctly -- every second event packet has a wrong code index number, which is the one from the last seen message, instead of 4. For example, the two messages "FE F0 01 02 03 04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event packets: correct: CH345: 0F FE 00 00 0F FE 00 00 04 F0 01 02 04 F0 01 02 04 03 04 05 0F 03 04 05 04 06 07 08 04 06 07 08 04 09 0A 0B 0F 09 0A 0B 04 0C 0D 0E 04 0C 0D 0E 05 F7 00 00 05 F7 00 00 A class-compliant driver must interpret an event packet with CIN 15 as having a single data byte, so the other two bytes would be ignored. The message received by the host would then be missing two bytes out of six; in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7". These corrupted SysEx event packages contain only data bytes, while the CH345 uses event packets with a correct CIN value only for messages with a status byte, so it is possible to distinguish between these two cases by checking for the presence of this status byte. (Other bugs in the CH345's input handling, such as the corruption resulting from running status, cannot be worked around.) Signed-off-by: Clemens Ladisch Cc: stable@vger.kernel.org Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 010094abf752..5b4c58c3e2c5 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -174,6 +174,8 @@ struct snd_usb_midi_in_endpoint { u8 running_status_length; } ports[0x10]; u8 seen_f5; + bool in_sysex; + u8 last_cin; u8 error_resubmit; int current_port; }; @@ -467,6 +469,39 @@ static void snd_usbmidi_maudio_broken_running_status_input( } } +/* + * QinHeng CH345 is buggy: every second packet inside a SysEx has not CIN 4 + * but the previously seen CIN, but still with three data bytes. + */ +static void ch345_broken_sysex_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int i, cin, length; + + for (i = 0; i + 3 < buffer_length; i += 4) { + if (buffer[i] == 0 && i > 0) + break; + cin = buffer[i] & 0x0f; + if (ep->in_sysex && + cin == ep->last_cin && + (buffer[i + 1 + (cin == 0x6)] & 0x80) == 0) + cin = 0x4; +#if 0 + if (buffer[i + 1] == 0x90) { + /* + * Either a corrupted running status or a real note-on + * message; impossible to detect reliably. + */ + } +#endif + length = snd_usbmidi_cin_length[cin]; + snd_usbmidi_input_data(ep, 0, &buffer[i + 1], length); + ep->in_sysex = cin == 0x4; + if (!ep->in_sysex) + ep->last_cin = cin; + } +} + /* * CME protocol: like the standard protocol, but SysEx commands are sent as a * single USB packet preceded by a 0x0F byte. @@ -660,6 +695,12 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { .output_packet = snd_usbmidi_output_standard_packet, }; +static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { + .input = ch345_broken_sysex_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + /* * AKAI MPD16 protocol: * @@ -2380,6 +2421,7 @@ int snd_usbmidi_create(struct snd_card *card, err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_CH345: + umidi->usb_protocol_ops = &snd_usbmidi_ch345_broken_sysex_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; default: -- cgit v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 From c87693da69f979f8a4370e7bc6115dd0898d8501 Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 19 Nov 2015 23:25:12 +0800 Subject: ALSA: hda - add PCI IDs for Intel Broxton Add HD Audio Device PCI ID for the Intel Broxton platform. It is an HDA Intel PCH controller. Signed-off-by: Lu, Han Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4d2cbe2ca141..10fa7432e678 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -312,6 +312,10 @@ enum { (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ AZX_DCAPS_I915_POWERWELL) +#define AZX_DCAPS_INTEL_BROXTON \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ @@ -2125,6 +2129,9 @@ static const struct pci_device_id azx_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Broxton-P(Apollolake) */ + { PCI_DEVICE(0x8086, 0x5a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, -- cgit v1.2.3-70-g09d2 From ff9d8859e2d4e47c2d5537e309b49cf1f2ed1ddc Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Thu, 19 Nov 2015 23:25:13 +0800 Subject: ALSA: hda - apply SKL display power request/release patch to BXT For SKL, only the HDMI codec is in the display power well while the HD-A controller isn't. So the codec flag 'link_power_control' is set to request/release the display power via bus link_power ops. For BXT, the power well design is the same as SKL, so the patch should be applied to BXT too. Signed-off-by: Lu, Han Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 60cd9e700909..bdb6f226d006 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2378,7 +2378,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) * can cover the codec power request, and so need not set this flag. * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec)) + if (is_valleyview_plus(codec) || is_skylake(codec) || + is_broxton(codec)) codec->core.link_power_control = 1; if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { -- cgit v1.2.3-70-g09d2 From b9c2fa52135d49a931c56ed2bfc17d61f771b412 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Nov 2015 16:39:50 +0100 Subject: ALSA: hda - Add fixup for Acer Aspire One Cloudbook 14 For making the speakers on Acer Aspire One Cloudbook 14 to work, we need the as same quirk as for another Chromebook. This patch adds the corresponding fixup entry. Reported-by: Patrick Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 081ef9bdf0df..53f6a0261ec9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5180,6 +5180,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- cgit v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 From 196543d54574f50e3fd04df4e3048181e006a9da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Nov 2015 14:46:35 +0100 Subject: ALSA: hda - Apply HP headphone fixups more generically It turned out that many HP laptops suffer from the same problem as fixed in commit [c932b98c1e47: ALSA: hda - Apply pin fixup for HP ProBook 6550b]. But, it's tiresome to list up all such PCI SSIDs, as there are really lots of HP machines. Instead, we do a bit more clever, try to check the supposedly dock and built-in headphone pins, and apply the fixup when both seem valid. This rule can be applied generically to all models using the same quirk, so we'll fix all in a shot. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=107491 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 45 +++++++++++++++++++++++++++--------------- 1 file changed, 29 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 826122d8acee..2c7c5eb8b1e9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3110,6 +3110,29 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, spec->gpio_led = 0x08; } +static bool is_hp_output(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin); + + /* count line-out, too, as BIOS sets often so */ + return get_defcfg_connect(pin_cfg) != AC_JACK_PORT_NONE && + (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || + get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT); +} + +static void fixup_hp_headphone(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, pin); + + /* It was changed in the BIOS to just satisfy MS DTM. + * Lets turn it back into slaved HP + */ + pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) | + (AC_JACK_HP_OUT << AC_DEFCFG_DEVICE_SHIFT); + pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC | AC_DEFCFG_SEQUENCE))) | + 0x1f; + snd_hda_codec_set_pincfg(codec, pin, pin_cfg); +} static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3119,22 +3142,12 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; - if (hp_blike_system(codec->core.subsystem_id)) { - unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); - if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || - get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || - get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) { - /* It was changed in the BIOS to just satisfy MS DTM. - * Lets turn it back into slaved HP - */ - pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) - | (AC_JACK_HP_OUT << - AC_DEFCFG_DEVICE_SHIFT); - pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC - | AC_DEFCFG_SEQUENCE))) - | 0x1f; - snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg); - } + /* when both output A and F are assigned, these are supposedly + * dock and built-in headphones; fix both pin configs + */ + if (is_hp_output(codec, 0x0a) && is_hp_output(codec, 0x0f)) { + fixup_hp_headphone(codec, 0x0a); + fixup_hp_headphone(codec, 0x0f); } if (find_mute_led_cfg(codec, 1)) -- cgit v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 From 8c69729b4439bbda88c3073df7243f755cc418ed Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 24 Nov 2015 11:08:18 +0800 Subject: ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3 We have a machine Dell XPS 13 with the codec alc256, after resume back from S3, the headphone has noise when play sound. Through comparing with the coeff vaule before and after S3, we found restoring a coeff register will help remove noise. BugLink: https://bugs.launchpad.net/bugs/1519168 Cc: Kailang Yang Cc: Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53f6a0261ec9..e4f80dc4704f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4587,6 +4587,7 @@ enum { ALC292_FIXUP_DISABLE_AAMIX, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5167,6 +5168,17 @@ static const struct hda_fixup alc269_fixups[] = { {} } }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x36}, + {0x20, AC_VERB_SET_PROC_COEF, 0x1737}, + {} + }, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5206,6 +5218,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3-70-g09d2 From 0c25ad80408e95e0a4fbaf0056950206e95f726f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Nov 2015 20:02:12 +0100 Subject: ALSA: hda - Fix noise on Gigabyte Z170X mobo Gigabyte Z710X mobo with ALC1150 codec gets significant noises from the analog loopback routes even if their inputs are all muted. Simply kill the aamix for fixing it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=108301 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e4f80dc4704f..9bedf7c85e29 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1759,6 +1759,7 @@ enum { ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, + ALC882_FIXUP_DISABLE_AAMIX, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1920,6 +1921,8 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action); +static void alc_fixup_disable_aamix(struct hda_codec *codec, + const struct hda_fixup *fix, int action); static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { @@ -2151,6 +2154,10 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC882_FIXUP_DISABLE_AAMIX] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2218,6 +2225,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), + SND_PCI_QUIRK(0x1458, 0xa182, "Gigabyte Z170X-UD3", ALC882_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- cgit v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 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') 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 v1.2.3-70-g09d2 From 8ae743e82f0b86f3b860c27fc2c8f574cf959fd0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Nov 2015 14:23:00 +0100 Subject: ALSA: hda - Skip ELD notification during system suspend The recent addition of ELD notifier for Intel HDMI/DP codec may lead the bad codec connection found as kernel messages like below: Suspending console(s) (use no_console_suspend to debug) hdmi_present_sense: snd_hda_codec_hdmi hdaudioC0D2: HDMI status: Codec=2 Pin=6 Presence_Detect=1 ELD_Valid=1 snd_hda_intel 0000:00:1f.3: spurious response 0x0:0x2, last cmd=0x206f2e08 snd_hda_intel 0000:00:1f.3: spurious response 0x0:0x2, last cmd=0x206f2e08 .... snd_hda_codec_hdmi hdaudioC0D2: HDMI: ELD buf size is 0, force 128 snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to polling mode: last cmd=0x206f2f00 snd_hda_intel 0000:00:1f.3: No response from codec, disabling MSI: last cmd=0x206f2f00 snd_hda_intel 0000:00:1f.3: azx_get_response timeout, switching to single_cmd mode: last cmd=0x206f2f00 azx_single_wait_for_response: 42 callbacks suppressed This seems appearing when the sound driver went to suspend before i915 driver. Then i915 driver disables HDMI/DP audio bit and calls the registered notifier, and the HDA codec tries to handle it as a hot(un)plug. But since the driver is already in the suspended state, it fails miserably. As this is a sort of spurious wakeup, it can be ignored safely, as long as it's delivered during the system suspend. OTOH, if a notification comes during the runtime suspend, the situation is different: we need to wake up. But during the system suspend, such a notification can't be the reason for a wakeup. This patch addresses it by a simple check of the current sound card status. The skipped notification doesn't matter because the HDA driver will check the plugged status forcibly at the resume in return. Then, why the card status, not a runtime PM status or else? The HDA controller driver is supposed to set the card status to D3 at the system suspend but not at the runtime suspend. So we can see it as a flag that is set only for the system suspend. Admittedly, it's a bit ugly, but it should work well for now. Reported-and-tested-by: "Zhang, Xiong Y" Fixes: 25adc137c546 ('ALSA: hda - Wake the codec up on pin/ELD notify events') Cc: # v4.3+ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bdb6f226d006..4b6fb668c91c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2352,6 +2352,12 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) struct hda_codec *codec = audio_ptr; int pin_nid = port + 0x04; + /* skip notification during system suspend (but not in runtime PM); + * the state will be updated at resume + */ + if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + return; + check_presence_and_report(codec, pin_nid); } -- cgit v1.2.3-70-g09d2 From 3b7e2a7d9d6220dad2950d8166f9495ef4c69b2e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Nov 2015 17:34:12 +0100 Subject: ALSA: hda - Correct codec names for 14f1:50f1 and 14f1:50f3 The numbers aren't always linear, just like in the real world. Correct to the right numbers stated in the datasheet (although we can't trust the datasheet as well). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c8b8ef5246a6..89d9e8ccea51 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -972,9 +972,9 @@ static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f1, "CX21722", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto), - HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f150f3, "CX21724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto), -- cgit v1.2.3-70-g09d2 From bcdda2ec28c31fe62d5d85c8b603f202a2c85fac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Nov 2015 17:36:02 +0100 Subject: ALSA: hda - Add Conexant CX8200 (14f1:2008) codec entry It's supposed to be equivalent with CX20724. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 89d9e8ccea51..ef198903c0c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -955,6 +955,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), -- cgit v1.2.3-70-g09d2 From a74a821624c0c75388a193337babd17a8c02c740 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 4 Dec 2015 16:44:24 +0100 Subject: ALSA: rme96: Fix unexpected volume reset after rate changes rme96 driver needs to reset DAC depending on the sample rate, and this results in resetting to the max volume suddenly. It's because of the missing call of snd_rme96_apply_dac_volume(). However, calling this function right after the DAC reset still may not work, and we need some delay before this call. Since the DAC reset and the procedure after that are performed in the spinlock, we delay the DAC volume restore at the end after the spinlock. Reported-and-tested-by: Sylvain LABOISNE Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme96.c | 41 ++++++++++++++++++++++++++--------------- 1 file changed, 26 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 714df906249e..41c31db65039 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -741,10 +741,11 @@ snd_rme96_playback_setrate(struct rme96 *rme96, { /* change to/from double-speed: reset the DAC (if available) */ snd_rme96_reset_dac(rme96); + return 1; /* need to restore volume */ } else { writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); + return 0; } - return 0; } static int @@ -980,6 +981,7 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err, rate, dummy; + bool apply_dac_volume = false; runtime->dma_area = (void __force *)(rme96->iobase + RME96_IO_PLAY_BUFFER); @@ -993,24 +995,26 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, { /* slave clock */ if ((int)params_rate(params) != rate) { - spin_unlock_irq(&rme96->lock); - return -EIO; - } - } else if ((err = snd_rme96_playback_setrate(rme96, params_rate(params))) < 0) { - spin_unlock_irq(&rme96->lock); - return err; - } - if ((err = snd_rme96_playback_setformat(rme96, params_format(params))) < 0) { - spin_unlock_irq(&rme96->lock); - return err; + err = -EIO; + goto error; + } + } else { + err = snd_rme96_playback_setrate(rme96, params_rate(params)); + if (err < 0) + goto error; + apply_dac_volume = err > 0; /* need to restore volume later? */ } + + err = snd_rme96_playback_setformat(rme96, params_format(params)); + if (err < 0) + goto error; snd_rme96_setframelog(rme96, params_channels(params), 1); if (rme96->capture_periodsize != 0) { if (params_period_size(params) << rme96->playback_frlog != rme96->capture_periodsize) { - spin_unlock_irq(&rme96->lock); - return -EBUSY; + err = -EBUSY; + goto error; } } rme96->playback_periodsize = @@ -1021,9 +1025,16 @@ snd_rme96_playback_hw_params(struct snd_pcm_substream *substream, rme96->wcreg &= ~(RME96_WCR_PRO | RME96_WCR_DOLBY | RME96_WCR_EMP); writel(rme96->wcreg |= rme96->wcreg_spdif_stream, rme96->iobase + RME96_IO_CONTROL_REGISTER); } + + err = 0; + error: spin_unlock_irq(&rme96->lock); - - return 0; + if (apply_dac_volume) { + usleep_range(3000, 10000); + snd_rme96_apply_dac_volume(rme96); + } + + return err; } static int -- cgit v1.2.3-70-g09d2 From 7c23b7c1996597dd9d60bb282fb5fa1be6ebd18b Mon Sep 17 00:00:00 2001 From: "Lu, Han" Date: Mon, 7 Dec 2015 15:59:13 +0800 Subject: ALSA: hda - Fix playback noise with 24/32 bit sample size on BXT In BXT-P A0, HD-Audio DMA requests is later than expected, and makes an audio stream sensitive to system latencies when 24/32 bits are playing. Adjusting threshold of DMA fifo to force the DMA request sooner to improve latency tolerance at the expense of power. v2: move Intel specific code to hda_intel.c Signed-off-by: Lu, Han Signed-off-by: Takashi Iwai --- include/sound/hda_register.h | 3 +++ sound/pci/hda/hda_intel.c | 23 +++++++++++++++++++++++ 2 files changed, 26 insertions(+) (limited to 'sound') diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 2ae8812d7b1a..94dc6a9772e0 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -93,6 +93,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_HSW_EM4 0x100c #define AZX_REG_HSW_EM5 0x1010 +/* Skylake/Broxton display HD-A controller Extended Mode registers */ +#define AZX_REG_SKL_EM4L 0x1040 + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 963f82430938..bff5c8b329d1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -355,6 +355,8 @@ enum { ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) +#define IS_BROXTON(pci) ((pci)->device == 0x5a98) + static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", [AZX_DRIVER_PCH] = "HDA Intel PCH", @@ -506,15 +508,36 @@ static void azx_init_pci(struct azx *chip) } } +/* + * In BXT-P A0, HD-Audio DMA requests is later than expected, + * and makes an audio stream sensitive to system latencies when + * 24/32 bits are playing. + * Adjusting threshold of DMA fifo to force the DMA request + * sooner to improve latency tolerance at the expense of power. + */ +static void bxt_reduce_dma_latency(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, SKL_EM4L); + val &= (0x3 << 20); + azx_writel(chip, SKL_EM4L, val); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); + struct pci_dev *pci = chip->pci; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); azx_init_chip(chip, full_reset); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, false); + + /* reduce dma latency to avoid noise */ + if (IS_BROXTON(pci)) + bxt_reduce_dma_latency(chip); } /* calculate runtime delay from LPIB */ -- cgit v1.2.3-70-g09d2 From 02f6ff90400d055f08b0ba0b5f0707630b6faed7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 7 Dec 2015 11:29:31 +0100 Subject: ALSA: hda - Add inverted dmic for Packard Bell DOTS On the internal mic of the Packard Bell DOTS, one channel has an inverted signal. Add a quirk to fix this up. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523232 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedf7c85e29..ebc53620a6f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6409,6 +6409,7 @@ static const struct hda_fixup alc662_fixups[] = { static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), -- cgit v1.2.3-70-g09d2 From 23adc192b862b69ad80a40bd5206e337f41264ac Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Tue, 8 Dec 2015 12:27:18 +0800 Subject: ALSA: hda - Fixing speaker noise on the two latest thinkpad models We have two latest thinkpad laptop models which are all based on the Intel skylake platforms, and all of them have the codec alc293 on them. When the machines boot to the desktop, an greeting dialogue shows up with the notification sound. But on these two models, there is noise with the notification sound. We have 3 SKUs for each of the models, all of them have this problem. So far, this problem is only specific to these two thinkpad models, we did not find this problem on the old thinkpad models with the codec alc293 or alc292. A workaround for this problem is disabling the aamix. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1523517 Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ebc53620a6f2..49b1d366f82a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4596,6 +4596,7 @@ enum { ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC293_FIXUP_LENOVO_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5187,6 +5188,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC293_FIXUP_LENOVO_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5334,6 +5341,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5343,6 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- cgit v1.2.3-70-g09d2 From 9a811230481243f384b8036c6a558bfdbd961f78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 9 Dec 2015 15:17:43 +0100 Subject: ALSA: hda - Fix noise problems on Thinkpad T440s Lenovo Thinkpad T440s suffers from constant background noises, and it seems to be a generic hardware issue on this model: https://forums.lenovo.com/t5/ThinkPad-T400-T500-and-newer-T/T440s-speaker-noise/td-p/1339883 As the noise comes from the analog loopback path, disabling the path is the easy workaround. Also, the machine gives significant cracking noises at PM suspend. A workaround found by trial-and-error is to disable the shutup callback currently used for ALC269-variant. This patch addresses these noise issues by introducing a new fixup chain. Although the same workaround might be applicable to other Thinkpad models, it's applied only to T440s (17aa:220c) in this patch, so far, just to be safe (you chicken!). As a compromise, a new model option string "tp440" is provided now, though, so that owners of other Thinkpad models can test it more easily. Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=958504 Reported-and-tested-by: Tim Hardeck Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 49b1d366f82a..8dd2ac13b3af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4204,6 +4204,18 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, } } +/* additional fixup for Thinkpad T440s noise problem */ +static void alc_fixup_tpt440(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->shutup = alc_no_shutup; /* reduce click noise */ + spec->gen.mixer_nid = 0; /* reduce background noise */ + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4578,6 +4590,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPT440, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5051,6 +5064,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPT440] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5332,7 +5351,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), @@ -5432,6 +5451,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-dac-wcaps"}, {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, + {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, {} }; -- cgit v1.2.3-70-g09d2 From 5328e1ea87fb2b5cf695115df4325c1913209e97 Mon Sep 17 00:00:00 2001 From: Gabriele Martino Date: Wed, 9 Dec 2015 17:05:58 +0100 Subject: ALSA: hda/ca0132 - quirk for Alienware 17 2015 The Alienware 17 (2015) has the same card and pin configuration of the Alienware 15, so the same quirks must be applied. Signed-off-by: Gabriele Martino Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index f8a12ca477f1..4ef2259f88ca 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -778,7 +778,8 @@ static const struct hda_pintbl alienware_pincfgs[] = { }; static const struct snd_pci_quirk ca0132_quirks[] = { - SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), {} }; -- cgit v1.2.3-70-g09d2 From 84ebac4d04d25ac5c1b1dc3ae621fd465eb38f4e Mon Sep 17 00:00:00 2001 From: John Keeping Date: Wed, 9 Dec 2015 11:38:13 +0000 Subject: ASoC: es8328: Fix deemphasis values This is using completely the wrong mask and value when updating the register. Since the correct values are already defined in the header, switch to using a table with explicit constants rather than shifting the array index. Signed-off-by: John Keeping Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + 2 files changed, 18 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 969e337dc17c..c4c64e21963e 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; +static const struct { + int rate; + unsigned int val; +} deemph_settings[] = { + { 0, ES8328_DACCONTROL6_DEEMPH_OFF }, + { 32000, ES8328_DACCONTROL6_DEEMPH_32k }, + { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k }, + { 48000, ES8328_DACCONTROL6_DEEMPH_48k }, +}; static int es8328_set_deemph(struct snd_soc_codec *codec) { @@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec) * rate. */ if (es8328->deemph) { - best = 1; - for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - es8328->playback_fs) < - abs(deemph_settings[best] - es8328->playback_fs)) + best = 0; + for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i].rate - es8328->playback_fs) < + abs(deemph_settings[best].rate - es8328->playback_fs)) best = i; } - val = best << 1; + val = deemph_settings[best].val; } else { - val = 0; + val = ES8328_DACCONTROL6_DEEMPH_OFF; } dev_dbg(codec->dev, "Set deemphasis %d\n", val); - return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, + ES8328_DACCONTROL6_DEEMPH_MASK, val); } static int es8328_get_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index cb36afe10c0e..156c748c89c7 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL6_CLICKFREE (1 << 3) #define ES8328_DACCONTROL6_DAC_INVR (1 << 4) #define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6) #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) -- cgit v1.2.3-70-g09d2 From e2a0c9fa80227be5ee017b5476638829dd41cb39 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 11 Dec 2015 13:06:24 +0200 Subject: ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx The condition for checking for XDAT being cleared was not correct. Fixes: 36bcecd0a73eb ("ASoC: davinci-mcasp: Correct TX start sequence") Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4495a40a9468..41235d3867c4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & - ~XRDATA) && (cnt < 100000)) + while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) && + (cnt < 100000)) cnt++; /* Release TX state machine */ -- cgit v1.2.3-70-g09d2 From 352d52e2442f42539c76d8a13d795ccab7079b26 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') diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index c4c64e21963e..afa6c5db9dcc 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -214,18 +214,18 @@ 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 v1.2.3-70-g09d2 From 1ea5998afe903384ddc16391d4c023cd4c867bea Mon Sep 17 00:00:00 2001 From: Mans Rullgard Date: Fri, 11 Dec 2015 11:27:08 +0000 Subject: ASoC: wm8974: set cache type for regmap Attempting to use this codec driver triggers a BUG() in regcache_sync() since no cache type is set. The register map of this device is fairly small and has few holes so a flat cache is suitable. Signed-off-by: Mans Rullgard Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8974.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 0a60677397b3..4c29bd2ae75c 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -574,6 +574,7 @@ static const struct regmap_config wm8974_regmap = { .max_register = WM8974_MONOMIX, .reg_defaults = wm8974_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults), + .cache_type = REGCACHE_FLAT, }; static int wm8974_probe(struct snd_soc_codec *codec) -- cgit v1.2.3-70-g09d2 From 5042f936c6810d0e7153cc9e1794c6998590a930 Mon Sep 17 00:00:00 2001 From: Sjoerd Simons Date: Fri, 11 Dec 2015 09:45:33 +0100 Subject: ASoC: rockchip: spdif: Set transmit data level to 16 samples Explicitly set the transmit data level on the transceiver to 16 samples rather then the default 0. This matches both the level set in the vendor kernel and the (seemingly very similar) i2s engine. This fixes audio glitches when playing back at 192k rate. At the same time, fix a trivial typo in the TDL mask definition Signed-off-by: Sjoerd Simons Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index a38a3029062c..bb09a071a320 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -152,8 +152,10 @@ static int rk_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_update_bits(spdif->regmap, SPDIF_DMACR, - SPDIF_DMACR_TDE_ENABLE, - SPDIF_DMACR_TDE_ENABLE); + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL_MASK, + SPDIF_DMACR_TDE_ENABLE | + SPDIF_DMACR_TDL(16)); if (ret != 0) return ret; diff --git a/sound/soc/rockchip/rockchip_spdif.h b/sound/soc/rockchip/rockchip_spdif.h index 07f86a21046a..9c24dbccf7ee 100644 --- a/sound/soc/rockchip/rockchip_spdif.h +++ b/sound/soc/rockchip/rockchip_spdif.h @@ -42,7 +42,7 @@ #define SPDIF_DMACR_TDL_SHIFT 0 #define SPDIF_DMACR_TDL(x) ((x) << SPDIF_DMACR_TDL_SHIFT) -#define SPDIF_DMACR_TDL_MASK (0x1f << SDPIF_DMACR_TDL_SHIFT) +#define SPDIF_DMACR_TDL_MASK (0x1f << SPDIF_DMACR_TDL_SHIFT) /* * XFER -- cgit v1.2.3-70-g09d2 From 42e3121d90f42e57f6dbd6083dff2f57b3ec7daa Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sun, 13 Dec 2015 20:49:58 +0200 Subject: ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and would cause software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. Commit 2d1cb7f658fb ("ALSA: usb-audio: add dB range mapping for some devices") added a dB range mapping for it with range 0..50 dB. However, the actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of approx. -53...0 dB. Replace the previous quirk with a custom dB mapping based on some basic output measurements, using a 10-item range TLV (which will still fit in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. v2: incorporated Takashi Iwai's suggestion for the quirk application method Signed-off-by: Anssi Hannula Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 2 ++ sound/usb/mixer_maps.c | 12 ------------ sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ sound/usb/mixer_quirks.h | 4 ++++ 4 files changed, 43 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f494dced3c11..4f85757009b3 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1354,6 +1354,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } } + snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl); + range = (cval->max - cval->min) / cval->res; /* * Are there devices with volume range more than 255? I use a bit more diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 6a803eff87f7..ddca6547399b 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -348,13 +348,6 @@ static struct usbmix_name_map bose_companion5_map[] = { { 0 } /* terminator */ }; -/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ -static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; -static struct usbmix_name_map dragonfly_1_2_map[] = { - { 7, NULL, .dB = &dragonfly_1_2_dB }, - { 0 } /* terminator */ -}; - /* * Control map entries */ @@ -470,11 +463,6 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x1020), .map = bose_companion5_map, }, - { - /* Dragonfly DAC 1.2 */ - .id = USB_ID(0x21b4, 0x0081), - .map = dragonfly_1_2_map, - }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index fe91184ce832..0ce888dceed0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include #include #include +#include #include "usbaudio.h" #include "mixer.h" @@ -1825,3 +1826,39 @@ void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, } } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + /* Approximation using 10 ranges based on output measurement on hw v1.2. + * This seems close to the cubic mapping e.g. alsamixer uses. */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + kctl->tlv.p = scale; + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; +} + +void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, int unitid, + struct snd_kcontrol *kctl) +{ + switch (mixer->chip->usb_id) { + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + if (unitid == 7 && cval->min == 0 && cval->max == 50) + snd_dragonfly_quirk_db_scale(mixer, kctl); + break; + } +} + diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h index bdbfab093816..177c329cd4dd 100644 --- a/sound/usb/mixer_quirks.h +++ b/sound/usb/mixer_quirks.h @@ -9,5 +9,9 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, int unitid); +void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, + struct usb_mixer_elem_info *cval, int unitid, + struct snd_kcontrol *kctl); + #endif /* SND_USB_MIXER_QUIRKS_H */ -- cgit v1.2.3-70-g09d2 From 12a6116e66695a728bcb9616416c508ce9c051a1 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Sun, 13 Dec 2015 20:49:59 +0200 Subject: ALSA: usb-audio: Add sample rate inquiry quirk for AudioQuest DragonFly Avoid getting sample rate on AudioQuest DragonFly as it is unsupported and causes noisy "cannot get freq at ep 0x1" messages when playback starts. Signed-off-by: Anssi Hannula Cc: Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7016ad898187..b6c0c8e3b450 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1125,6 +1125,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } return false; -- cgit v1.2.3-70-g09d2 From c04017ea81dc1eccae87be7ac7b82b2972f9931f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 15 Dec 2015 14:44:03 +0100 Subject: ALSA: hda - Fix headphone mic input on a few Dell ALC293 machines These laptops support both headphone, headset and mic modes for the 3.5mm jack. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1526330 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8dd2ac13b3af..b745a721c363 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4606,6 +4606,7 @@ enum { ALC288_FIXUP_DISABLE_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, + ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, @@ -5169,6 +5170,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE }, + [ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC292_FIXUP_DELL_E7X] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, @@ -5247,11 +5254,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), - SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- cgit v1.2.3-70-g09d2 From 157f0b7f6c0cc0bc88647390006e959e267a0143 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Dec 2015 23:30:43 +0100 Subject: ALSA: hda - Apply click noise workaround for Thinkpads generically It seems that a workaround for Thinkpad T440s crackling noise can be applied generically to all Thinkpad models: namely, disabling the default alc269 shutup callback. This patch moves it to the existing alc_fixup_tpt440_dock() while also replacing the rest code with another existing alc_fixup_disable_aamix(). It resulted in a good code reduction. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439 Reported-and-tested-by: Benjamin Poirier Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b745a721c363..531065eaac1b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4198,24 +4198,13 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->shutup = alc_no_shutup; /* reduce click noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); } } -/* additional fixup for Thinkpad T440s noise problem */ -static void alc_fixup_tpt440(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct alc_spec *spec = codec->spec; - - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->shutup = alc_no_shutup; /* reduce click noise */ - spec->gen.mixer_nid = 0; /* reduce background noise */ - } -} - static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5067,7 +5056,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC292_FIXUP_TPT440] = { .type = HDA_FIXUP_FUNC, - .v.func = alc_fixup_tpt440, + .v.func = alc_fixup_disable_aamix, .chained = true, .chain_id = ALC292_FIXUP_TPT440_DOCK, }, -- cgit v1.2.3-70-g09d2 From 70a0976b0c0d90f4246d7e63359d796ec82b87d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 Dec 2015 14:59:58 +0100 Subject: ALSA: hda - Set codec to D3 at reboot/shutdown on Thinkpads Lenovo Thinkpads with Realtek codecs may still have some loud crackling noises at reboot/shutdown even though a few previous fixes have been applied. It's because the previous fix (disabling the default shutup callback) takes effect only at transition of the codec power state. Meanwhile, at reboot or shutdown, we don't take down the codec power as default, thus it triggers the same problem unless the codec is powered down casually by runtime PM. This patch tries to address the issue. It gives two things: - implement the separate reboot_notify hook to struct alc_spec, and call it optionally if defined. - turn off the codec to D3 for Thinkpad models via this new callback Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439 Reported-and-tested-by: Benjamin Poirier Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 531065eaac1b..5a79c7a2187a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -111,6 +111,7 @@ struct alc_spec { void (*power_hook)(struct hda_codec *codec); #endif void (*shutup)(struct hda_codec *codec); + void (*reboot_notify)(struct hda_codec *codec); int init_amp; int codec_variant; /* flag for other variants */ @@ -773,6 +774,25 @@ static inline void alc_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc_reboot_notify(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec && spec->reboot_notify) + spec->reboot_notify(codec); + else + alc_shutup(codec); +} + +/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */ +static void alc_d3_at_reboot(struct hda_codec *codec) +{ + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); +} + #define alc_free snd_hda_gen_free #ifdef CONFIG_PM @@ -818,7 +838,7 @@ static const struct hda_codec_ops alc_patch_ops = { .suspend = alc_suspend, .check_power_status = snd_hda_gen_check_power_status, #endif - .reboot_notify = alc_shutup, + .reboot_notify = alc_reboot_notify, }; @@ -4199,6 +4219,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->shutup = alc_no_shutup; /* reduce click noise */ + spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); -- cgit v1.2.3-70-g09d2 From b6903c0ed9f0bcbbe88f67f7ed43d1721cbc6235 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Dec 2015 12:20:20 +0100 Subject: ALSA: hda - Add a fixup for Thinkpad X1 Carbon 2nd Apply the same fixup for Thinkpad with dock to Thinkpad X1 Carbon 2nd, too. This reduces the annoying loud cracking noise problem, as well as the support of missing docking port. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=958439 Reported-and-tested-by: Benjamin Poirier Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5a79c7a2187a..6c268dad143f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5375,6 +5375,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), -- cgit v1.2.3-70-g09d2 From c803cc2dcd722e08020c1ba63bb5ceece4a19fdb Mon Sep 17 00:00:00 2001 From: Jean-Michel Hautbois Date: Thu, 17 Dec 2015 11:07:23 +0100 Subject: ASoC: sgtl5000: fix VAG power up timing When power up, a "pop" is heard on line-in and mic-in. An analysis of the PCM shows it lasts ~400ms and looks like a filter response. VAG power up should be delayed by 400ms as VAG power down is. Signed-off-by: Jean-Michel Hautbois Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f540f82b1f27..08b40460663c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -189,6 +189,7 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + msleep(400); break; case SND_SOC_DAPM_PRE_PMD: -- cgit v1.2.3-70-g09d2 From 3e6db33aaf1d42a30339f831ec4850570d6cc7a3 Mon Sep 17 00:00:00 2001 From: Xiong Zhang Date: Fri, 18 Dec 2015 13:29:18 +0800 Subject: ALSA: hda - Set SKL+ hda controller power at freeze() and thaw() It takes three minutes to enter into hibernation on some OEM SKL machines and we see many codec spurious response after thaw() opertion. This is because HDA is still in D0 state after freeze() call and pci_pm_freeze/pci_pm_freeze_noirq() don't set D3 hot in pci_bus driver. It seems bios still access HDA when system enter into freeze state, HDA will receive codec response interrupt immediately after thaw() call. Because of this unexpected interrupt, HDA enter into a abnormal state and slow down the system enter into hibernation. In this patch, we put HDA into D3 hot state in azx_freeze_noirq() and put HDA into D0 state in azx_thaw_noirq(). V2: Only apply this fix to SKL+ Fix compile error when CONFIG_PM_SLEEP isn't defined [Yet another fix for CONFIG_PM_SLEEP ifdef and the additional comment by tiwai] Signed-off-by: Xiong Zhang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bff5c8b329d1..3b3658297070 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -954,6 +954,36 @@ static int azx_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP || SUPPORT_VGA_SWITCHEROO */ +#ifdef CONFIG_PM_SLEEP +/* put codec down to D3 at hibernation for Intel SKL+; + * otherwise BIOS may still access the codec and screw up the driver + */ +#define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) +#define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) + +static int azx_freeze_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (IS_SKL_PLUS(pci)) + pci_set_power_state(pci, PCI_D3hot); + + return 0; +} + +static int azx_thaw_noirq(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + + if (IS_SKL_PLUS(pci)) + pci_set_power_state(pci, PCI_D0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + #ifdef CONFIG_PM static int azx_runtime_suspend(struct device *dev) { @@ -1063,6 +1093,10 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) +#ifdef CONFIG_PM_SLEEP + .freeze_noirq = azx_freeze_noirq, + .thaw_noirq = azx_thaw_noirq, +#endif SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; -- cgit v1.2.3-70-g09d2 From 3e3f8bd569558acefdfaae273d71f7a29b8c0b4f Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Fri, 18 Dec 2015 16:53:41 +0800 Subject: ASoC: fsl_sai: fix no frame clk in master mode After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a4435f5e3be9..a31f0ba527eb 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -504,6 +504,24 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* + * For sai master mode, after several open/close sai, + * there will be no frame clock, and can't recover + * anymore. Add software reset to fix this issue. + * This is a hardware bug, and will be fix in the + * next sai version. + */ + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- cgit v1.2.3-70-g09d2 From 9f660a1c43890c2cdd1f423fd73654e7ca08fe56 Mon Sep 17 00:00:00 2001 From: Mario Kleiner Date: Tue, 22 Dec 2015 00:45:43 +0100 Subject: ALSA: hda/realtek - Fix silent headphone output on MacPro 4,1 (v2) Without this patch, internal speaker and line-out work, but front headphone output jack stays silent on the Mac Pro 4,1. This code path also gets executed on the MacPro 5,1 due to identical codec SSID, but i don't know if it has any positive or adverse effects there or not. (v2) Implement feedback from Takashi Iwai: Reuse alc889_fixup_mbp_vref and just add a new nid 0x19 for the MacPro 4,1. Signed-off-by: Mario Kleiner Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6c268dad143f..fe96428aa403 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1775,6 +1775,7 @@ enum { ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, ALC889_FIXUP_MP11_VREF, + ALC889_FIXUP_MP41_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1863,7 +1864,7 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x14, 0x15 }; + static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 }; int i; if (action != HDA_FIXUP_ACT_INIT) @@ -2153,6 +2154,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC885_FIXUP_MACPRO_GPIO, }, + [ALC889_FIXUP_MP41_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic, @@ -2235,7 +2242,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 4,1/5,1", ALC889_FIXUP_MP41_VREF), SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), -- cgit v1.2.3-70-g09d2 From fb203adc28a3717c252bde0f068b3ebd2206994b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 5 Jan 2016 17:16:03 +0530 Subject: ASoC: Intel: Skylake: Revert previous broken fix memory leak fix This reverts commit 87b5ed8ecb9fe05a696e1c0b53c7a49ea66432c1 ("ASoC: Intel: Skylake: fix memory leak") as it causes regression on Skylake devices The SKL drivers can be deferred probe. The topology file based widgets can have references to topology file so this can't be freed until card is fully created, so revert this patch for now [ 66.682767] BUG: unable to handle kernel paging request at ffffc900001363fc [ 66.690735] IP: [] strnlen+0xd/0x40 [ 66.696509] PGD 16e035067 PUD 16e036067 PMD 16e038067 PTE 0 [ 66.702925] Oops: 0000 [#1] PREEMPT SMP [ 66.768390] CPU: 3 PID: 57 Comm: kworker/u16:3 Tainted: G O 4.4.0-rc7-skl #62 [ 66.778869] Hardware name: Intel Corporation Skylake Client platform [ 66.793201] Workqueue: deferwq deferred_probe_work_func [ 66.799173] task: ffff88008b700f40 ti: ffff88008b704000 task.ti: ffff88008b704000 [ 66.807692] RIP: 0010:[] [] strnlen+0xd/0x40 [ 66.816243] RSP: 0018:ffff88008b707878 EFLAGS: 00010286 [ 66.822293] RAX: ffffffff80e60a82 RBX: 000000000000000e RCX: fffffffffffffffe [ 66.830406] RDX: ffffc900001363fc RSI: ffffffffffffffff RDI: ffffc900001363fc [ 66.838520] RBP: ffff88008b707878 R08: 000000000000ffff R09: 000000000000ffff [ 66.846649] R10: 0000000000000001 R11: ffffffffa01c6368 R12: ffffc900001363fc [ 66.854765] R13: 0000000000000000 R14: 00000000ffffffff R15: 0000000000000000 [ 66.862910] FS: 0000000000000000(0000) GS:ffff88016ecc0000(0000) knlGS:0000000000000000 [ 66.872150] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 66.878696] CR2: ffffc900001363fc CR3: 0000000002c09000 CR4: 00000000003406e0 [ 66.886820] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 66.894938] DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 [ 66.903052] Stack: [ 66.905346] ffff88008b7078b0 ffffffff806cb1db 000000000000000e 0000000000000000 [ 66.913854] ffff88008b707928 ffffffffa00d1050 ffffffffa00d104e ffff88008b707918 [ 66.922353] ffffffff806ccbd6 ffff88008b707948 0000000000000046 ffff88008b707940 [ 66.930855] Call Trace: [ 66.933646] [] string.isra.4+0x3b/0xd0 [ 66.939793] [] vsnprintf+0x116/0x540 [ 66.945742] [] kvasprintf+0x40/0x80 [ 66.951591] [] kasprintf+0x40/0x50 [ 66.957359] [] dapm_create_or_share_kcontrol+0x1cf/0x300 [snd_soc_core] [ 66.966771] [] ? __kmalloc+0x16e/0x2a0 [ 66.972931] [] snd_soc_dapm_new_widgets+0x41b/0x4b0 [snd_soc_core] [ 66.981857] [] ? snd_soc_dapm_add_routes+0xb0/0xd0 [snd_soc_core] [ 67.007828] [] soc_probe_component+0x23d/0x360 [snd_soc_core] [ 67.016244] [] ? mutex_unlock+0x9/0x10 [ 67.022405] [] snd_soc_instantiate_card+0x47f/0xd10 [snd_soc_core] [ 67.031329] [] ? debug_mutex_init+0x32/0x40 [ 67.037973] [] snd_soc_register_card+0x1d2/0x2b0 [snd_soc_core] [ 67.046619] [] devm_snd_soc_register_card+0x44/0x80 [snd_soc_core] [ 67.055539] [] skylake_audio_probe+0x1b/0x20 [snd_soc_skl_rt286] [ 67.064292] [] platform_drv_probe+0x37/0x90 Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ffea427aeca8..a7854c8fc523 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1240,7 +1240,6 @@ 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 v1.2.3-70-g09d2 From d8018361b58bb7b9a2a657104e54c33c2ef1439d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 5 Jan 2016 17:16:04 +0530 Subject: ASoC: Intel: Skylake: Fix the memory leak This provide the fix for firmware memory by freeing the pointer in driver remove where it is safe to do so Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-topology.c | 2 ++ sound/soc/intel/skylake/skl.c | 4 ++++ sound/soc/intel/skylake/skl.h | 2 ++ 3 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a7854c8fc523..ad4d0f82603e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1248,5 +1248,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) skl->resource.max_mcps = SKL_MAX_MCPS; skl->resource.max_mem = SKL_FW_MAX_MEM; + skl->tplg = fw; + return 0; } diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 5319529aedf7..caa69c4598a6 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include "skl.h" @@ -520,6 +521,9 @@ static void skl_remove(struct pci_dev *pci) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); + if (skl->tplg) + release_firmware(skl->tplg); + if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); pci_dev_put(pci); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index dd2e79ae45a8..a0709e344d44 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -68,6 +68,8 @@ struct skl { struct skl_dsp_resource resource; struct list_head ppl_list; struct list_head dapm_path_list; + + const struct firmware *tplg; }; #define skl_to_ebus(s) (&(s)->ebus) -- cgit v1.2.3-70-g09d2