From e5b94083d0996b4c69674c8a5563c2eb272557ba Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Fri, 7 Aug 2015 14:08:15 -0700 Subject: [PATCH 1/7] ASoC: max98090: Fix sequencing when starting additional routes Enforce correct device sequencing when configuring a new audio route when there is an existing active audio route(s). This patch fixed recording noise issue while playback is active. We have some registers which require the device to be in full shutdown or to enter full shutdown before the register settings will take effect. Currently the driver is not shutting down the device when a new audio route is created. If a new audio route is made active while there is already an active audio route, then the required register sequencing is violated. A hardware shutdown toggle when creating a new audio route corrects the sequencing error. The device must remain in hardware shutdown for 40ms to allow the internal hardware core to fully shutdown. Signed-off-by: Fang, Yang A Signed-off-by: Sathyanarayana Nujella Acked-by: Anish Kumar Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 46 ++++++++++++++++++++++++++++++++----- sound/soc/codecs/max98090.h | 1 + 2 files changed, 41 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 78268f0514e9..2a4c2e12972a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -850,6 +850,19 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, return 0; } +static int max98090_shdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (event & SND_SOC_DAPM_POST_PMU) + max98090->shdn_pending = true; + + return 0; + +} + static const char *mic1_mux_text[] = { "IN12", "IN56" }; static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, @@ -1158,9 +1171,11 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, M98090_SDOEN_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICL_SHIFT, 0, NULL, 0), + M98090_DIGMICL_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICR_SHIFT, 0, NULL, 0), + M98090_DIGMICR_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, M98090_AHPF_SHIFT, 0, NULL, 0), @@ -1205,10 +1220,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_adc_mixer_controls[0], ARRAY_SIZE(max98090_right_adc_mixer_controls)), - SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADLEN_SHIFT, 0), - SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADREN_SHIFT, 0), + SND_SOC_DAPM_ADC_E("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), @@ -2536,9 +2553,26 @@ static int max98090_remove(struct snd_soc_codec *codec) return 0; } +static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (max98090->shdn_pending) { + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(40); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + max98090->shdn_pending = false; + } +} + static struct snd_soc_codec_driver soc_codec_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, + .seq_notifier = max98090_seq_notifier, .set_bias_level = max98090_set_bias_level, }; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 21ff743f5af2..bc610d9a9ecb 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1543,6 +1543,7 @@ struct max98090_priv { unsigned int pa2en; unsigned int sidetone; bool master; + bool shdn_pending; }; int max98090_mic_detect(struct snd_soc_codec *codec, From 9b850ca4f1c5acd7fcbbd4b38a2d27132801a8d5 Mon Sep 17 00:00:00 2001 From: John Lin Date: Tue, 11 Aug 2015 14:27:25 +0800 Subject: [PATCH 2/7] ASoC: rt5640: fix line out no sound issue The power for line out was not turned on when line out is enabled. So we add "LOUT amp" widget to turn on the power for line out. Signed-off-by: John Lin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5640.c | 40 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 37 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9bc78e57513d..ff72cd8c236e 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -984,6 +984,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, RT5640_PWR_LM); + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, 0); + break; + + default: + return 0; + } + + return 0; +} + static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1179,13 +1208,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, @@ -1500,8 +1532,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HP R Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HP L Playback"}, {"HPOR", NULL, "HP R Playback"}, - {"LOUTL", NULL, "LOUT MIX"}, - {"LOUTR", NULL, "LOUT MIX"}, + + {"LOUT amp", NULL, "LOUT MIX"}, + {"LOUTL", NULL, "LOUT amp"}, + {"LOUTR", NULL, "LOUT amp"}, }; static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { From 9d8352864907f0ad76124c5b28f65b5a382d7d7c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 14 Aug 2015 17:54:07 +0800 Subject: [PATCH 3/7] ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config Don't set .read_flag_mask for adav803, it's for adav801 only. Fixes: 0c2d69645628 ("ASoC: adav80x: Split SPI and I2C code into different modules") Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/adav80x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 36d842570745..69c63b92e078 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -865,7 +865,6 @@ const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, - .read_flag_mask = 0x01, .max_register = ADAV80X_PLL_OUTE, From 1cf5a330c05ae37a0a98ac7c9800a6f50d5579ec Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Wed, 19 Aug 2015 16:02:24 +0100 Subject: [PATCH 4/7] ASoC: arizona: Fix gain settings of FLL in free-run mode The wrong register was used to set the gain of ref loop, when changing the FLL output on an active FLL. This patch corrects the offset of the gain register. Signed-off-by: Nikesh Oswal Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05eae3e9..4e5d0a96f933 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2058,7 +2058,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, From 14a500fe1396934c6b3ed8f009459a4723da7862 Mon Sep 17 00:00:00 2001 From: Vaishali Thakkar Date: Thu, 20 Aug 2015 22:11:15 +0530 Subject: [PATCH 5/7] ASoC: samsung: Remove redundant arndale_audio_remove There is no use of snd_soc_unregister_card in remove function as devm_snd_soc_register_card in probe function automatically handles it. So, remove use of snd_soc_unregister_card and with this change remove arndale_audio_remove as it is now redundant. Signed-off-by: Vaishali Thakkar Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/samsung/arndale_rt5631.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index 8bf2e2c4bafb..9e371eb3e4fa 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -116,15 +116,6 @@ static int arndale_audio_probe(struct platform_device *pdev) return ret; } -static int arndale_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -139,7 +130,6 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, - .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); From e134cb2041359c927b735b81568373624ea26baf Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Fri, 14 Aug 2015 11:22:16 +0800 Subject: [PATCH 6/7] ASoC: wm8994: fix add dynamic path error If there don't exist dynamic sink or source widget, it will failed to add dynamic path. "AIF3ADCDAT" is snd_soc_dapm_aif_out, can't be dynamic sink widget. So change the audio route to fix this issue. Signed-off-by: Zidan Wang Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 962e1d31a629..2ccbb322df77 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1942,14 +1942,16 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" }, - { "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" }, - { "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, - { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC1R" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2L" }, + { "AIF3ADC Mux", "AIF1ADCDAT", "AIF1ADC2R" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, + { "AIF3ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACL" }, + { "AIF3ADC Mux", "AIF2DACDAT", "AIF2DACR" }, + + { "AIF3ADCDAT", NULL, "AIF3ADC Mux" }, /* Loopback */ { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, From 0e7659712836ca59b4735bc5cc94de38698a5e01 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 25 Aug 2015 12:43:48 +0100 Subject: [PATCH 7/7] ASoC: arizona: Poll for FLL clock OK rather than use interrupts The extcon driver takes the DAPM mutex from within the interrupt thread in several places, which makes it possible to get into a situation where the interrupt thread is blocked waiting on the DAPM mutex whilst a DAPM sequence is running which is attempting to configure the FLL. In this case the FLL completion can't be completed as as the IRQ handler is ONE_SHOT, which cause the FLL lock to use the full time out (250mS) and report that the process timed out. It is not really practical to make the extcon driver not take the DAPM mutex from within the interrupt thread, at least not without extensive modification. So this patch fixes the issue by switching the wait for the FLL lock to polling. A few fast polls are done first as the FLL should lock quickly for a good quality reference clock, (indeed it hits on the first poll on my system) and it will poll every 20mS after that until it times out. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/arizona.c | 47 +++++++++++++++----------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 19 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4e5d0a96f933..4180827a8480 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1756,17 +1756,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_clock_ok(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "clock OK\n"); - - complete(&fll->ok); - - return IRQ_HANDLED; -} - static struct { unsigned int min; unsigned int max; @@ -2048,10 +2037,11 @@ static int arizona_is_enabled_fll(struct arizona_fll *fll) static int arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; - unsigned long time_left; bool use_sync = false; int already_enabled = arizona_is_enabled_fll(fll); struct arizona_fll_cfg cfg; + int i; + unsigned int val; if (already_enabled < 0) return already_enabled; @@ -2110,9 +2100,6 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (!already_enabled) pm_runtime_get(arizona->dev); - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); if (use_sync) @@ -2124,10 +2111,24 @@ static int arizona_enable_fll(struct arizona_fll *fll) regmap_update_bits_async(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); - time_left = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (time_left == 0) + arizona_fll_dbg(fll, "Waiting for FLL lock...\n"); + val = 0; + for (i = 0; i < 15; i++) { + if (i < 5) + usleep_range(200, 400); + else + msleep(20); + + regmap_read(arizona->regmap, + ARIZONA_INTERRUPT_RAW_STATUS_5, + &val); + if (val & (ARIZONA_FLL1_CLOCK_OK_STS << (fll->id - 1))) + break; + } + if (i == 15) arizona_fll_warn(fll, "Timed out waiting for lock\n"); + else + arizona_fll_dbg(fll, "FLL locked (%d polls)\n", i); return 0; } @@ -2212,11 +2213,8 @@ EXPORT_SYMBOL_GPL(arizona_set_fll); int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { - int ret; unsigned int val; - init_completion(&fll->ok); - fll->id = id; fll->base = base; fll->arizona = arizona; @@ -2238,13 +2236,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, - arizona_fll_clock_ok, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", - id, ret); - } - regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, 0); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 43deb0462309..36867d05e0bb 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -242,7 +242,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion ok; unsigned int fout; int sync_src;