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[alsa-devel] [PATCH 2/3] ASoC: dt-bindings: rt5682: add button delay device property
by shumingf＠realtek.com 30 Oct '19

30 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The btndet-delay device property could control the HW debounce time. It is easy to adjust the sensitivity of push button. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 312e9a129530..30e927a28369 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -27,6 +27,11 @@ Optional properties: - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,btndet-delay + The debounce delay for push button. + The delay time is realtek,btndet-delay value multiple of 8.192 ms. + If absent, the default is 16. + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -47,4 +52,5 @@ rt5682 { realtek,dmic1-data-pin = <1>; realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; + realtek,btndet-delay = <16>; }; -- 2.23.0

2 2

[alsa-devel] [PATCH 3/3] ASoC: rt1011: improve readability and remove redundant code
by shumingf＠realtek.com 30 Oct '19

30 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> 1. remove !component->card->instantiated statement in kcontrol 2. goto the end of fucntion if "ret = -EINVAL" in rt1011_set_dai_fmt() 3. fix typo 4. add the range check for temperature_calib from device property 5. remove unnecessary tabs using spaces instead Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 72 ++++++++++++++++++--------------------- 1 file changed, 34 insertions(+), 38 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index dcd397a83cb4..10bbdb37d689 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -61,7 +61,6 @@ static const struct reg_sequence init_list[] = { { RT1011_DAC_SET_1, 0xe702 }, { RT1011_DAC_SET_3, 0x2004 }, }; -#define RT1011_INIT_REG_LEN ARRAY_SIZE(init_list) static const struct reg_default rt1011_reg[] = { {0x0000, 0x0000}, @@ -684,7 +683,8 @@ static int rt1011_reg_init(struct snd_soc_component *component) { struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - regmap_multi_reg_write(rt1011->regmap, init_list, RT1011_INIT_REG_LEN); + regmap_multi_reg_write(rt1011->regmap, + init_list, ARRAY_SIZE(init_list)); return 0; } @@ -989,7 +989,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_din_source_enum, RT1011_CROSS_BQ_SET_1, 5, static const char * const rt1011_tdm_data_out_select[] = { "TDM_O_LR", "BQ1", "DVOL", "BQ10", "ALC", "DMIX", "ADC_SRC_LR", - "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", + "ADC_O_LR", "ADC_MONO", "RSPK_BPF_LR", "DMIX_ADD", "ENVELOPE_FS", "SEP_O_GAIN", "ALC_BK_GAIN", "STP_V_C", "DMIX_ABST" }; @@ -1002,7 +1002,7 @@ static SOC_ENUM_SINGLE_DECL(rt1011_tdm2_l_dac1_enum, RT1011_TDM2_SET_4, 12, rt1011_tdm_l_ch_data_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_dat_enum, - RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); + RT1011_ADCDAT_OUT_SOURCE, 0, rt1011_tdm_data_out_select); static SOC_ENUM_SINGLE_DECL(rt1011_tdm1_adc1_loc_enum, RT1011_TDM1_SET_2, 0, rt1011_tdm_l_ch_data_select); @@ -1024,9 +1024,9 @@ static const char * const rt1011_tdm_adc_swap_select[] = { "L/R", "R/L", "L/L", "R/R" }; -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc1_1_enum, RT1011_TDM1_SET_3, 6, rt1011_tdm_adc_swap_select); -static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, +static SOC_ENUM_SINGLE_DECL(rt1011_tdm_adc2_1_enum, RT1011_TDM1_SET_3, 4, rt1011_tdm_adc_swap_select); static void rt1011_reset(struct regmap *regmap) @@ -1092,9 +1092,9 @@ static bool rt1011_validate_bq_drc_coeff(unsigned short reg) { if ((reg == RT1011_DAC_SET_1) | (reg >= RT1011_ADC_SET && reg <= RT1011_ADC_SET_1) | - (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | + (reg == RT1011_ADC_SET_4) | (reg == RT1011_ADC_SET_5) | (reg == RT1011_MIXER_1) | - (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && + (reg == RT1011_A_TIMING_1) | (reg >= RT1011_POWER_7 && reg <= RT1011_POWER_8) | (reg == RT1011_CLASS_D_POS) | (reg == RT1011_ANALOG_CTRL) | (reg >= RT1011_SPK_TEMP_PROTECT_0 && @@ -1163,9 +1163,6 @@ static int rt1011_bq_drc_coeff_put(struct snd_kcontrol *kcontrol, (struct rt1011_bq_drc_params *)ucontrol->value.integer.value; unsigned int i, mode_idx = 0; - if (!component->card->instantiated) - return 0; - if (strstr(ucontrol->id.name, "AdvanceMode Initial Set")) mode_idx = RT1011_ADVMODE_INITIAL_SET; else if (strstr(ucontrol->id.name, "AdvanceMode SEP BQ Coeff")) @@ -1236,9 +1233,6 @@ static int rt1011_r0_cali_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct rt1011_priv *rt1011 = snd_soc_component_get_drvdata(component); - if (!component->card->instantiated) - return 0; - rt1011->cali_done = 0; if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF && ucontrol->value.integer.value[0]) @@ -1284,9 +1278,6 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == rt1011->r0_reg) return 0; - if (!component->card->instantiated) - return 0; - if (ucontrol->value.integer.value[0] == 0) return -EINVAL; @@ -1298,7 +1289,7 @@ static int rt1011_r0_load_mode_put(struct snd_kcontrol *kcontrol, r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); if (rt1011->r0_reg) @@ -1640,6 +1631,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1650,6 +1642,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1666,6 +1659,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: ret = -EINVAL; + goto _set_fmt_err_; } switch (dai->id) { @@ -1683,6 +1677,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = -EINVAL; } +_set_fmt_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } @@ -1778,7 +1773,8 @@ static int rt1011_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", + freq_in); return ret; } @@ -1982,14 +1978,14 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM2_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | - RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); + RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, - RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, - RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); + RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); if (tx_slotnum) snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, @@ -2114,7 +2110,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .remove = rt1011_remove, .suspend = rt1011_suspend, .resume = rt1011_resume, - .set_bias_level = rt1011_set_bias_level, + .set_bias_level = rt1011_set_bias_level, .controls = rt1011_snd_controls, .num_controls = ARRAY_SIZE(rt1011_snd_controls), .dapm_widgets = rt1011_dapm_widgets, @@ -2123,9 +2119,9 @@ static const struct snd_soc_component_driver soc_component_dev_rt1011 = { .num_dapm_routes = ARRAY_SIZE(rt1011_dapm_routes), .set_sysclk = rt1011_set_component_sysclk, .set_pll = rt1011_set_component_pll, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1011_regmap = { @@ -2227,17 +2223,17 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_ADC_OFFSET_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "ADC offset=0x%x\n", dc_offset); + dev_info(dev, "ADC offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G0_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain0 offset=0x%x\n", dc_offset); regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_20_16, &value); dc_offset = value << 16; regmap_read(rt1011->regmap, RT1011_EFUSE_DAC_OFFSET_G1_15_0, &value); dc_offset |= (value & 0xffff); - dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); + dev_info(dev, "Gain1 offset=0x%x\n", dc_offset); if (cali_flag) { @@ -2257,7 +2253,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) while (count < chk_cnt) { msleep(100); regmap_read(rt1011->regmap, - RT1011_INIT_RECIPROCAL_SYN_24_16, &value); + RT1011_INIT_RECIPROCAL_SYN_24_16, &value); r0[count%3] = value << 16; regmap_read(rt1011->regmap, RT1011_INIT_RECIPROCAL_SYN_15_0, &value); @@ -2272,7 +2268,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) break; } if (count > chk_cnt) { - dev_err(dev, "Calibrate R0 Failure\n"); + dev_err(dev, "Calibrate R0 Failure\n"); ret = -EAGAIN; } else { format = 2147483648U; /* 2^24 * 128 */ @@ -2281,7 +2277,7 @@ static int rt1011_calibrate(struct rt1011_priv *rt1011, unsigned char cali_flag) - (r0_integer * 100); rt1011->r0_reg = r0[0]; rt1011->cali_done = 1; - dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(dev, "r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, r0[0]); } } @@ -2346,7 +2342,8 @@ static void rt1011_calibration_work(struct work_struct *work) rt1011_reg_init(component); /* Apply temperature and calibration data from device property */ - if (rt1011->temperature_calib) { + if (rt1011->temperature_calib <= 0xff && + rt1011->temperature_calib > 0) { snd_soc_component_update_bits(component, RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, (rt1011->temperature_calib << 2)); @@ -2359,7 +2356,7 @@ static void rt1011_calibration_work(struct work_struct *work) r0_integer = format / rt1011->r0_reg / 128; r0_factor = ((format / rt1011->r0_reg * 100) / 128) - (r0_integer * 100); - dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", r0_integer, r0_factor, rt1011->r0_reg); rt1011_r0_load(rt1011); @@ -2380,7 +2377,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) } static int rt1011_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { struct rt1011_priv *rt1011; int ret; @@ -2388,7 +2385,7 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, rt1011 = devm_kzalloc(&i2c->dev, sizeof(struct rt1011_priv), GFP_KERNEL); - if (rt1011 == NULL) + if (!rt1011) return -ENOMEM; i2c_set_clientdata(i2c, rt1011); @@ -2425,7 +2422,6 @@ static void rt1011_i2c_shutdown(struct i2c_client *client) rt1011_reset(rt1011->regmap); } - static struct i2c_driver rt1011_i2c_driver = { .driver = { .name = "rt1011", -- 2.23.0

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[alsa-devel] [PATCH] ASoC: rt5677-spi: fixup compile warning
by Kuninori Morimoto 30 Oct '19

30 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch fixup this warning LINUX/sound/soc/codecs/rt5677-spi.c: In function ‘rt5677_spi_pcm_close’: LINUX/sound/soc/codecs/rt5677-spi.c:114:30: warning: unused variable ‘rtd’ [-Wunused-variable] struct snd_soc_pcm_runtime *rtd = substream->private_data; ^~~ Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/rt5677-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index dac1097..3aa3ea7 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -111,7 +111,6 @@ static int rt5677_spi_pcm_close( struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; struct rt5677_dsp *rt5677_dsp = snd_soc_component_get_drvdata(component); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-motu: Correct a typo in the clock proc string
by Takashi Iwai 30 Oct '19

30 Oct '19

Just fix a typo of "S/PDIF" in the clock name string. Fixes: 4638ec6ede08 ("ALSA: firewire-motu: add proc node to show current statuc of clock and packet formats") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/firewire/motu/motu-proc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index b47ba818343f..187f6abd878c 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -16,7 +16,7 @@ static const char *const clock_names[] = { [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT] = "S/PDIF on optical interface", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_A] = "S/PDIF on optical interface A", [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_B] = "S/PDIF on optical interface B", - [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface", + [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PDIF on coaxial interface", [SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface", [SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface", [SND_MOTU_CLOCK_SOURCE_SPH] = "Source packet header", -- 2.16.4

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[alsa-devel] [PATCH 0/6] ALSA: firewire-motu: add support for MOTU UltraLite and source packet header clock mode
by Takashi Sakamoto 30 Oct '19

30 Oct '19

Hi, Devices in MOTU FireWire series uses source packet header (SPH) in IEC 61883-1 for each data block in CIP. The devices have a mode to use the series of SPH for generating sampling clock. This patchset supports the above mechanism. Additionally, this patchset also supports MOTU UltraLite which adopts protocol version 2. Takashi Sakamoto (6): ALSA: firewire-motu: fix wrong spelling for macro ALSA: firewire-motu: print for unknown source of sampling clock ALSA: firewire-motu: detect SPH source of sampling clock ALSA: firewire-motu: minor code refactoring for protocol version 2 ALSA: firewire-motu: code refactoring to handle model specific switch for protocol v2 ALSA: firewire-motu: add support for MOTU UltraLite sound/firewire/motu/motu-pcm.c | 3 +- sound/firewire/motu/motu-proc.c | 2 + sound/firewire/motu/motu-protocol-v2.c | 142 ++++++++++++++----------- sound/firewire/motu/motu-protocol-v3.c | 4 +- sound/firewire/motu/motu.c | 34 ++++-- sound/firewire/motu/motu.h | 6 +- 6 files changed, 113 insertions(+), 78 deletions(-) -- 2.20.1

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[alsa-devel] snd-bebob : from kernel 4.13 to 5.3.19 and .20
by Jean-Paul Argudo 30 Oct '19

30 Oct '19

Hi there, I wrote 1st this here: https://github.com/takaswie/snd-firewire-improve/issues/25 But I think it was the bad place, sorry for that. I think this ML is better? My problem in one line: I upgraded my Ubuntu Disco (kernel 4.13) to Ubuntu Eoan (kernel 5.3.19) and I can report my SaffireLE / Firewire doesnt work anymore, snd-bebob reports "discontinuity of CIP" on dmesg... At startup it lights green ok, but no sound is playable, then the lights turn orange (like it is when it's not working), I hear a "relay sound" (a electric clic of a relay), then, the Saffire LE disapears from the sound menu in Ubuntu sound menu. I can see this in dmesg: [ 19.083583] snd-bebob fw1.0: Detect discontinuity of CIP: 10 50 [ 19.746665] snd-bebob fw1.0: Detect discontinuity of CIP: A0 A8 [ 19.846775] snd-bebob fw1.0: Detect discontinuity of CIP: 70 B0 [ 20.335861] snd-bebob fw1.0: Detect discontinuity of CIP: 90 D0 [ 20.856301] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 21.312770] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 21.796716] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 22.347664] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 22.818247] snd-bebob fw1.0: Detect discontinuity of CIP: A8 E8 [ 23.306079] snd-bebob fw1.0: Detect discontinuity of CIP: 98 D8 [ 29.544652] snd-bebob fw1.0: Detect discontinuity of CIP: 10 50 [ 30.272090] snd-bebob fw1.0: Detect discontinuity of CIP: A8 E8 [ 30.776168] snd-bebob fw1.0: Detect discontinuity of CIP: 00 40 [ 31.268123] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 31.766085] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 32.267903] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 32.767989] snd-bebob fw1.0: Detect discontinuity of CIP: F8 38 [ 33.263808] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 33.757156] snd-bebob fw1.0: Detect discontinuity of CIP: 08 48 [ 34.258860] snd-bebob fw1.0: Detect discontinuity of CIP: 80 C0 [ 34.755571] snd-bebob fw1.0: Detect discontinuity of CIP: 40 80 [ 35.261252] snd-bebob fw1.0: Detect discontinuity of CIP: 60 A0 [ 35.753980] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 36.253547] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 36.752500] snd-bebob fw1.0: Detect discontinuity of CIP: 08 48 [ 37.255203] snd-bebob fw1.0: Detect discontinuity of CIP: B0 F0 [ 277.760550] snd-bebob fw1.0: Detect discontinuity of CIP: 40 80 [ 278.488972] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 278.986290] snd-bebob fw1.0: Detect discontinuity of CIP: E0 20 [ 279.485998] snd-bebob fw1.0: Detect discontinuity of CIP: 18 58 [ 279.983838] snd-bebob fw1.0: Detect discontinuity of CIP: D0 10 [ 280.486776] snd-bebob fw1.0: Detect discontinuity of CIP: C0 00 [ 280.979487] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 281.480952] snd-bebob fw1.0: Detect discontinuity of CIP: 78 B8 [ 281.975413] snd-bebob fw1.0: Detect discontinuity of CIP: 38 78 [ 282.475487] snd-bebob fw1.0: Detect discontinuity of CIP: 60 A0 [ 282.979446] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 283.472403] snd-bebob fw1.0: Detect discontinuity of CIP: 88 C8 [ 283.968484] snd-bebob fw1.0: Detect discontinuity of CIP: 68 A8 [ 284.471585] snd-bebob fw1.0: Detect discontinuity of CIP: 30 70 [ 284.562886] usb 3-4.1: reset high-speed USB device number 5 using xhci_hcd [ 284.965508] snd-bebob fw1.0: Detect discontinuity of CIP: D0 10 [ 285.469348] snd-bebob fw1.0: Detect discontinuity of CIP: 68 A8 [ 285.965174] snd-bebob fw1.0: Detect discontinuity of CIP: 20 60 [ 285.981618] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 [ 290.103982] firewire_core 0000:05:01.0: phy config: new root=ffc1, gap_count=5 I tried update the kernel with eoan-proposed I know run 5.3.20 and I have the same problem (the trace above is made with 5.3.20. First, I hope I report this the best way, really not sure here. If not, please tell me what to bring (files? logs?) and where (on this list? elsewhere?). I have spare time to investigate anything that could help here. Thanks! -- Jean-Paul Argudo

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Re: [alsa-devel] [PATCH 2/2] ASoC: fsl_asrc: Add support for imx8qm
by S.j. Wang 30 Oct '19

30 Oct '19

Hi > > On Tue, Oct 29, 2019 at 05:17:09PM +0800, Shengjiu Wang wrote: > > There are two asrc module in imx8qm, each module has different clock > > configuration, and the DMA type is EDMA. > > > > So in this patch, we define the new clocks, refine the clock map, and > > include struct fsl_asrc_soc_data for different soc usage. > > > > The EDMA channel is fixed with each dma request, one dma request > > corresponding to one dma channel. So we need to request dma channel > > with dma request of asrc module. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 91 +++++++++++++++++++++++++++++------- > > sound/soc/fsl/fsl_asrc.h | 65 +++++++++++++++++++++++++- > > sound/soc/fsl/fsl_asrc_dma.c | 39 ++++++++++++---- > > 3 files changed, 167 insertions(+), 28 deletions(-) > > > diff --git a/sound/soc/fsl/fsl_asrc_dma.c > > b/sound/soc/fsl/fsl_asrc_dma.c index d6146de9acd2..dbb07a486504 > 100644 > > --- a/sound/soc/fsl/fsl_asrc_dma.c > > +++ b/sound/soc/fsl/fsl_asrc_dma.c > > @@ -199,19 +199,40 @@ static int fsl_asrc_dma_hw_params(struct > > snd_soc_component *component, > > > > /* Get DMA request of Back-End */ > > tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); > > - tmp_data = tmp_chan->private; > > - pair->dma_data.dma_request = tmp_data->dma_request; > > - dma_release_channel(tmp_chan); > > + /* tmp_chan may be NULL for it is already allocated by Back-End */ > > + if (tmp_chan) { > > + tmp_data = tmp_chan->private; > > + if (tmp_data) > > + pair->dma_data.dma_request = > > + tmp_data->dma_request; > > If this patch is supposed to add a !tmp_chan case for EDMA, we probably > shouldn't mute the !tmp_data case because dma_request will be NULL, > although the code previously didn't have a check either. I mean we might > need to error-out for !tmp_chan. Or... > is this intentional? > Yes, intentional. May be we can change to if (!asrc_priv->soc->use_edma) { /* Get DMA request of Back-End */ tmp_chan = dma_request_slave_channel(dev_be, tx ? "tx" : "rx"); tmp_data = tmp_chan->private; pair->dma_data.dma_request = tmp_data->dma_request; dma_release_channel(tmp_chan); /* Get DMA request of Front-End */ tmp_chan = fsl_asrc_get_dma_channel(pair, dir); tmp_data = tmp_chan->private; pair->dma_data.dma_request2 = tmp_data->dma_request; pair->dma_data.peripheral_type = tmp_data->peripheral_type; pair->dma_data.priority = tmp_data->priority; dma_release_channel(tmp_chan); } Best regards Wang shengjiu

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[alsa-devel] [PATCH v1] ALSA: usb-audio: sound: usb: usb true/false for bool return type
by Saurav Girepunje 30 Oct '19

30 Oct '19

Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje <saurav.girepunje(a)gmail.com> --- Changes in v1: - Update the patch subject suggested by Takashi Iwai <tiwai(a)suse.de> - Drop "!!" but keep the ternary operator in last return statement so that function will return always true/false. sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 72e9bdf76115..8d5538063598 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -167,21 +167,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -193,10 +193,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, -- 2.20.1

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[alsa-devel] axfer: break transmission of audio data frame when transmission backend queues no PCM event within timeout
by GitHub pull_request - opened 30 Oct '19

30 Oct '19

alsa-project/alsa-utils pull request #31 was opened from takaswie: Axfer internally has the waiter abstraction for polling system call. This is used without explicit timeout for IRQ-based scheduling model. As a result, axfer process is corrupted to be blocked due to a call of polling system call when transmission backend has a bug to queue PCM event in I/O thread or hardware IRQ context. This patchset implements error reporting for such buggy backend. When call of polling system call returns with timeout, axfer reports it as a bug. The timeout is equivalent to the size of buffer. Precisely it should be the size of period, however the larger size is used to stabilize runtime for existent PCM plugins. ``` Takashi Sakamoto (6): axfer: return the number of file descriptors for I/O events from select(2) waiter axfer: fix to return error code when a call of select(2) fails axfer: return ETIMEDOUT when no event occurs after waiter expiration axfer: handle -ETIMEDOUT before handle mmap I/O operation axfer: code refactoring for a helper function to wait for avail buffer space axfer: handle -ETIMEDOUT before non-block I/O operation axfer/waiter-select.c | 4 +-- axfer/xfer-libasound-irq-mmap.c | 19 +++++++++- axfer/xfer-libasound-irq-rw.c | 62 +++++++++++++++++++-------------- axfer/xfer-libasound.c | 21 +++++++---- 4 files changed, 70 insertions(+), 36 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/31 Patch URL : https://github.com/alsa-project/alsa-utils/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] [PATCH 1/2] ASoC: dt-bindings: fsl_asrc: add compatible string for imx8qm
by Shengjiu Wang 30 Oct '19

30 Oct '19

In order to support the two asrc modules in imx8qm, we need to add compatible string "fsl,imx8qm-asrc0" and "fsl,imx8qm-asrc1" Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,asrc.txt | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,asrc.txt b/Documentation/devicetree/bindings/sound/fsl,asrc.txt index 1d4d9f938689..cd2bd3daa7e1 100644 --- a/Documentation/devicetree/bindings/sound/fsl,asrc.txt +++ b/Documentation/devicetree/bindings/sound/fsl,asrc.txt @@ -8,7 +8,8 @@ three substreams within totally 10 channels. Required properties: - - compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc". + - compatible : Contains "fsl,imx35-asrc", "fsl,imx53-asrc", + "fsl,imx8qm-asrc0" or "fsl,imx8qm-asrc1". - reg : Offset and length of the register set for the device. -- 2.21.0

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[alsa-devel] Adding support for Android's Wired Audio Headset Specification v1.1
by Mads Lønsethagen 30 Oct '19

30 Oct '19

Posting this to alsa-devel without being actually sure it belongs here, so please do tell if it belongs elsewhere. But here goes: If you have e.g. a Bose QC25 headset (wired, not bluetooth) in Linux, you can't get the volume buttons nor the play/pause button on the headphone cable itself to work. As far as I know, these buttons works in both Android and Windows 10, so it would be nice if they would work with plain Linux too. This StackExchange thread discusses how it works: https://electronics.stackexchange.com/questions/242883/how-do-media-buttons… As one poster on StackExchange mentions, Android has a defined standard on these kind of buttons: https://source.android.com/devices/accessories/headset/plug-headset-spec Since these kind of button presses must be decoded from impedance changes in the 3.5mm jack connector, does it belong in alsa somewhere? Bose QC25 buttons works on Intel HDA cards on new Dell XPS laptops, so I'm guessing it should be possible within snd-hda-intel... - Mads

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[alsa-devel] [PATCH v8 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 29 Oct '19

29 Oct '19

Hey, here's the 8th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Intended to go in via Mark's ASoC tree. v8 changes: - rebased on top of Mark's for-next (50484b6a523a) - addressed Takashi's review comment to patch 5 - addressed two sparse warnings in patch 3 reported by kbuild test robot <lkp(a)intel.com> - fixed itnull.cocci warning in patch 3 reported by Julia Lawall <julia.lawall(a)lip6.fr> and kbuild test robot <lkp(a)intel.com> Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.c | 85 +++++++++++++ sound/soc/intel/boards/hda_dsp_common.h | 32 +++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 6 + sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 19 files changed, 367 insertions(+), 31 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.c create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH 00/26] ASoC: SOF: enable S0ix support for Intel platforms
by Pierre-Louis Bossart 29 Oct '19

29 Oct '19

In S0ix (aka S0 Idle), the audio DSP can enter the D0i3 substate and keep some of the pipelines and processing active for e.g. Hotwording and other acoustic event detection while optimizing power consumption. This patchset mainly adds flags and state machines to track which streams may remain active (tagged as such in the topology file), and keep them active when the ALSA core tries to suspend everything. The Intel specific parts are limited to the changes in the IPC, where instead of using an SRAM window, the IPC commands needed to enter and exit D0i3 are sent in a compressed form using IPC registers themselves. This patchset is made of 26 patches, which is rather large, but each change is rather small and contained. The code was initially reviewed on GitHub, with 165+ comments provided and no regressions found [1]. The S0ix mode was tested on a CometLake platform. Future improvements will consist in an additional idle detection so that the D0i3 mode can be entered even on devices where S0ix is not enabled. [1] https://github.com/thesofproject/linux/pull/1337 Keyon Jie (26): ASoC: SOF: add a field to store the current D0 substate of DSP ASoC: SOF: reset default d0_substate at probe() and resume() ASoC: SOF: add set_power_state() to dsp_ops for power state update ASoC: SOF: Intel: hda-dsp: Add helper for setting DSP D0ix substate ASoC: SOF: Intel: CNL: add set_power_state() ops ASoC: SOF: Intel: APL: add set_power_state() ops ASoC: SOF: add flag to snd_sof_pcm_stream for D0i3 compatible stream ASoC: SOF: token: add tokens for PCM compatible with D0i3 substate ASoC: SOF: topology: parse and store d0i3_compatible flag ASoC: SOF: Intel: hda-dsp: align the comments for D0I3C update ASoC: SOF: Intel: HDA: use macro for register polling retry count ASoC: SOF: PM: rename sof_send_pm_ipc to sof_send_pm_ctx_ipc ASoC: SOF: ipc: introduce message for DSP power gating ASoC: SOF: Intel: hda-ipc: Don't read mailbox for PM_GATE reply ASoC: SOF: Intel: HDA: add cAVS specific compact IPC header file ASoC: SOF: configure D0ix IPC flags in set_power_state ASoC: SOF: PM: add helpers for setting D0 substate for ADSP ASoC: SOF: Intel: CNL: add support for sending compact IPC ASoC: SOF: add a flag to indicate the system suspend target ASoC: SOF: add a flag suspend_ignored for sof stream ASoC: SOF: PM: implement prepare/complete callbacks ASoC: SOF: ignore suspend/resume for D0ix compatible streams ASoC: SOF: Intel: hda-dsp: implement suspend/resume for S0ix<->S0 transition ASoC: SOF: return -ENOTSUPP if D0I3 is not supported ASoC: SOF: PM: Add support for DSP D0i3 state when entering S0ix ASoC: SOF: pci: Add prepare/complete PM callbacks include/sound/sof/header.h | 1 + include/sound/sof/pm.h | 8 +++ include/uapi/sound/sof/tokens.h | 4 ++ sound/soc/sof/core.c | 3 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 43 +++++++++++++-- sound/soc/sof/intel/hda-dsp.c | 91 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 6 ++- sound/soc/sof/intel/hda-ipc.h | 51 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 12 +++++ sound/soc/sof/ops.h | 10 ++++ sound/soc/sof/pcm.c | 31 +++++++++++ sound/soc/sof/pm.c | 95 +++++++++++++++++++++++++++++++-- sound/soc/sof/sof-pci-dev.c | 2 + sound/soc/sof/sof-priv.h | 23 ++++++++ sound/soc/sof/topology.c | 43 ++++++++++++++- 16 files changed, 413 insertions(+), 11 deletions(-) create mode 100644 sound/soc/sof/intel/hda-ipc.h -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: bxt_da7219_max98357a: update dai_link platform name
by mac.chiang＠intel.com 29 Oct '19

29 Oct '19

From: Mac Chiang <mac.chiang(a)intel.com> redefine soc platform name "0000:00:1f.3" if cml core Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5..38b7dbe 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -40,6 +40,13 @@ struct bxt_card_private { struct list_head hdmi_pcm_list; }; +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + enum { BXT_DPCM_AUDIO_PB = 0, BXT_DPCM_AUDIO_CP, @@ -708,6 +715,7 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].name = "SSP0-Codec"; broxton_dais[i].cpus->dai_name = "SSP0 Pin"; } + broxton_dais[i].platforms = platform_component; } } -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: intel - fix the card names
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Those strings are exposed to the user space as the card name thus used in the GUIs. The common standard is to avoid '_' here. The worst case is 'sof-skl_hda_card' string. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a..d14ca327c684 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -623,7 +623,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 */ static struct snd_soc_card kabylake_audio_card = { - .name = "kbl_r5514_5663_max", + .name = "kbl-r5514-5663-max", .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 1778acdc367c..e8d676c192f6 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -90,7 +90,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) } static struct snd_soc_card hda_soc_card = { - .name = "skl_hda_card", + .name = "hda-dsp", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, .dapm_widgets = skl_hda_widgets, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..320a9b9273db 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -370,7 +370,7 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { - .name = "sof_rt5682", + .name = "rt5682", /* the sof- prefix is added by the core */ .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

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[alsa-devel] [PATCH] ASoC: SOF - remove the dead code (skylake/kabylake)
by Jaroslav Kysela 29 Oct '19

29 Oct '19

Appearently the CONFIG_SND_SOC_SOF_KABYLAKE and CONFIG_SND_SOC_SOF_SKYLAKE options are not present in Kconfig and 'struct snd_sof_dsp_ops sof_skl_ops' is not declared in the code, too. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.h | 1 - sound/soc/sof/sof-pci-dev.c | 44 ------------------------------------- 2 files changed, 45 deletions(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 23e430d3e056..f521c48ec2eb 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -596,7 +596,6 @@ extern struct snd_soc_dai_driver skl_dai[]; */ extern const struct snd_sof_dsp_ops sof_apl_ops; extern const struct snd_sof_dsp_ops sof_cnl_ops; -extern const struct snd_sof_dsp_ops sof_skl_ops; extern const struct sof_intel_dsp_desc apl_chip_info; extern const struct sof_intel_dsp_desc cnl_chip_info; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3a9e0e2a150d..bca3fb791a6b 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -168,42 +168,6 @@ static const struct sof_dev_desc icl_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) -static const struct sof_dev_desc skl_desc = { - .machines = snd_soc_acpi_intel_skl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-skl.ri", - .nocodec_tplg_filename = "sof-skl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) -static const struct sof_dev_desc kbl_desc = { - .machines = snd_soc_acpi_intel_kbl_machines, - .resindex_lpe_base = 0, - .resindex_pcicfg_base = -1, - .resindex_imr_base = -1, - .irqindex_host_ipc = -1, - .resindex_dma_base = -1, - .chip_info = &skl_chip_info, - .default_fw_path = "intel/sof", - .default_tplg_path = "intel/sof-tplg", - .nocodec_fw_filename = "sof-kbl.ri", - .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", - .ops = &sof_skl_ops, - .arch_ops = &sof_xtensa_arch_ops -}; -#endif - #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, @@ -407,14 +371,6 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa348), .driver_data = (unsigned long)&cfl_desc}, #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) - { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = (unsigned long)&kbl_desc}, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) - { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = (unsigned long)&skl_desc}, -#endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) { PCI_DEVICE(0x8086, 0x34C8), .driver_data = (unsigned long)&icl_desc}, -- 2.20.1

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[alsa-devel] [ANNOUNCE] Draft schedule for 2019 Linux Audio miniconference
by Mark Brown 29 Oct '19

29 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. The agenda is looking pretty full now, what we've got so far is: - Use case management enhancements (Curtis) - DSP framework integration (Liam) - SOF integration with ACPI and DT (Daniel) - SOF support for non-modular drivers (Daniel) - Gapless compressed playback (Patrick) - Soundwire status (Liam?) - Multi-CPU support - Mixing with HDA/I2S in a single system - PCM issues (Patrick) - PCM device volume control - Attaching timestamps to buffers - Virtualization (Liam) - User experiences (Jerome) - Unified graph (Patrick) - Componentisation status/plans (Me) If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

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[alsa-devel] [PATCH] usb: clock.c : usb true/false for bool return type
by Saurav Girepunje 29 Oct '19

29 Oct '19

Use true/false for bool type return in uac_clock_source_is_valid(). Signed-off-by: Saurav Girepunje <saurav.girepunje(a)gmail.com> --- sound/usb/clock.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 6b8c14f9b5d4..8b8ab83fac0d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -165,21 +165,21 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = le32_to_cpu(cs_desc->bmControls); } else { /* UAC_VERSION_1/2 */ struct uac_clock_source_descriptor *cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, source_id); if (!cs_desc) - return 0; + return false; bmControls = cs_desc->bmControls; } /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) - return 1; + return true; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -191,10 +191,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, dev_warn(&dev->dev, "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return 0; + return false; } - return !!data; + return !!data ? true : false; } static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, -- 2.20.1

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[alsa-devel] [PATCH 00/21] Enable BACO for power savings (v2)
by Alex Deucher 29 Oct '19

29 Oct '19

This patch set enables BACO (Bus Active Chip Off) for power savings on VI+ asics. Similar to PowerXpress and Hybrid Graphics (PX/HG) laptops, we can disable GPUs at runtime when they are not in use is they support BACO. The runtime pm code in amdgpu was originally developed for PX/HG laptops, so it was pretty entangled with the vga_switcheroo and ACPI code. Since the GPU contains an audio codec for HDMI/DP, there is some interaction with the hda driver. I am by no means an expert on alsa, so any advice on those patches is much appreciated. I had to enable runtime pm on the hda device to allow the GPU to enter runtime pm because they are linked. These patches depend on the clean patches and CI/VI BACO patches I sent out earlier today. The full tree can be found here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=baco TODO: - Turn off runtime pm when KFD is active - Make sure audio still works - Make sure PX/HG still works v2: - Split whitespace changes from baco callback addition - Fix pci ref counting in px/hg handling in hda driver - Handle CONFIG_ACPI properly - Split adding new hda pci ids from adding runpm flag Alex Deucher (21): drm/amdgpu: add asic callback for BACO support drm/amdgpu/soc15: move struct definition around to align with other soc15 asics drm/amdgpu: add supports_baco callback for soc15 asics. (v2) drm/amdgpu: add supports_baco callback for SI asics. drm/amdgpu: add supports_baco callback for CIK asics. drm/amdgpu: add supports_baco callback for VI asics. drm/amdgpu: add supports_baco callback for NV asics. drm/amdgpu: add a amdgpu_device_supports_baco helper drm/amdgpu: rename amdgpu_device_is_px to amdgpu_device_supports_boco (v2) drm/amdgpu: add additional boco checks to runtime suspend/resume (v2) drm/amdgpu: split swSMU baco_reset into enter and exit drm/amdgpu: add helpers for baco entry and exit drm/amdgpu: add baco support to runtime suspend/resume drm/amdgpu: start to disentangle boco from runtime pm drm/amdgpu: disentangle runtime pm and vga_switcheroo drm/amdgpu: enable runtime pm on BACO capable boards if runpm=1 drm/amdgpu/runpm: enable runpm on baco capable VI+ asics ALSA: hda/hdmi - fix vgaswitcheroo detection for AMD ALSA: hda/hdmi - Add new pci ids for AMD GPU display audio ALSA: hda/hdmi - enable runtime pm for newer AMD display audio ALSA: hda/hdmi - enable automatic runtime pm for AMD HDMI codecs by default drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 +- drivers/gpu/drm/amd/amdgpu/amdgpu_device.c | 106 ++++++++++++++++-- drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 63 ++++++----- drivers/gpu/drm/amd/amdgpu/amdgpu_kms.c | 20 +++- drivers/gpu/drm/amd/amdgpu/cik.c | 18 +++ drivers/gpu/drm/amd/amdgpu/nv.c | 18 ++- drivers/gpu/drm/amd/amdgpu/si.c | 6 + drivers/gpu/drm/amd/amdgpu/soc15.c | 36 +++++- drivers/gpu/drm/amd/amdgpu/vi.c | 22 ++++ drivers/gpu/drm/amd/powerplay/amdgpu_smu.c | 20 +++- drivers/gpu/drm/amd/powerplay/arcturus_ppt.c | 3 +- .../gpu/drm/amd/powerplay/inc/amdgpu_smu.h | 6 +- drivers/gpu/drm/amd/powerplay/inc/smu_v11_0.h | 3 +- drivers/gpu/drm/amd/powerplay/navi10_ppt.c | 3 +- drivers/gpu/drm/amd/powerplay/smu_v11_0.c | 9 +- drivers/gpu/drm/amd/powerplay/vega20_ppt.c | 3 +- sound/pci/hda/hda_intel.c | 84 +++++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 18 files changed, 371 insertions(+), 62 deletions(-) -- 2.23.0

2 25

[alsa-devel] axfer: call of polling system call never return when driver fails to queue no period-wakeup event due to the lack of hw IRQ context
by GitHub issues - edited 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was edited from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] axfer: call of polling system call never return when driver fails to queue any events due to the lack of hw IRQ context
by GitHub issues - opened 29 Oct '19

29 Oct '19

alsa-project/alsa-utils issue #30 was opened from takaswie: Axfer calls polling system call(i.e. poll(2), epoll(7), select(7)) without explicit timeout when running for IRQ-based scheduling model. In a case that driver achieves to control sound hardware for periodical hardware IRQ, the call can return due to any event queued by the driver in hardware IRQ context. However, in a case that driver fails to control it, the call cannot return. Especially, this situation easily occurs when developers work to support hardware newly. It's better to care for the situation. How to reproduce: 1. prepare sound hardware which driver fail to control for periodical hardware IRQ. 2. execute axfer for the driver/hardware with options to: * use IRQ-based scheduling model: `--sched-model=irq` * use polling system call: without `--test-nowait` 3. the running process blocks as a call of polling system call (defaultly poll(2) by snd_pcm_wait()) Issue URL : https://github.com/alsa-project/alsa-utils/issues/30 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v8 0/6] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 29 Oct '19

29 Oct '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c 6fa5963c37a2 ASoC: hdmi-codec: Add an op to set callback function for plug event has been merged to upstream. Changes from v7 to v8: 1. rockchip_max98090: Allow three different use cases: max98090-only: For backward compatibility where DTS does not specify HDMI node. HDMI-only: For HDMI-only board like veyron-mickey. max98090 + HDMI: For other veyron boards. Pass different compatible string to specify the use case. 2. Add more maintainers to cc-list for new device property reviewing. Cheng-Yi Chiang (6): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Support usage with and without HDMI ASoC: rockchip_max98090: Optionally support HDMI use case ASoC: rockchip_max98090: Add HDMI jack support ARM: dts: rockchip: Add HDMI support to rk3288-veyron-analog-audio ARM: dts: rockchip: Add HDMI audio support to rk3288-veyron-mickey.dts .../bindings/sound/rockchip-max98090.txt | 38 +- .../boot/dts/rk3288-veyron-analog-audio.dtsi | 3 +- arch/arm/boot/dts/rk3288-veyron-mickey.dts | 7 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 +- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 392 +++++++++++++++--- 8 files changed, 425 insertions(+), 74 deletions(-) -- 2.24.0.rc0.303.g954a862665-goog

3 12

[alsa-devel] [PATCH 0/2] Update adau7118 dt-bindings
by Nuno Sá 29 Oct '19

29 Oct '19

This is a follow-up patch as requested by Rob. The dt-bindings patch for this device was applied before Rob's review. Here is the link to the review: https://lore.kernel.org/alsa-devel/20191010210850.GA20184@bogus/ Nuno Sá (2): dt-bindings: asoc: adau7118: Cleanup ASOC: adau7118: Change regulators id .../bindings/sound/adi,adau7118.yaml | 25 ++++++++----------- sound/soc/codecs/adau7118.c | 4 +-- 2 files changed, 12 insertions(+), 17 deletions(-) -- 2.23.0

4 5

[alsa-devel] [PATCH] Revert "ALSA: hda: Flush interrupts on disabling"
by Takashi Iwai 28 Oct '19

28 Oct '19

This reverts commit caa8422d01e983782548648e125fd617cadcec3f. It turned out that this commit caused a regression at shutdown / reboot, as the synchronize_irq() calls seems blocking the whole shutdown. Also another part of the change about shuffling the call order looks suspicious; the azx_stop_chip() call disables the CORB / RIRB while the others may still need the CORB/RIRB update. Since the original commit itself was a cargo-fix, let's revert the whole patch. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205333 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_controller.c | 2 -- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d3999e7b0705..7e7be8e4dcf9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - synchronize_irq(bus->irq); - /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a815bc811799..cf53fbd872ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1348,9 +1348,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { - azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); + azx_stop_chip(chip); } if (bus->irq >= 0) -- 2.16.4

3 4

[alsa-devel] [PATCH v7 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 28 Oct '19

28 Oct '19

Hey, here's the 7th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v7 changes: - Only change since v6 is moving inlined logic from hda_dsp_common.h to hda_dsp_common.c and related changes to machine driver make rules. This is to address Takashi's feedback. - Rebased on top of Mark's tree (agreed with Takashi and Pierre to submit via Mark's ASoC tree). Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/Makefile | 10 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.c | 85 +++++++++++++ sound/soc/intel/boards/hda_dsp_common.h | 32 +++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 19 files changed, 370 insertions(+), 32 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.c create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

3 15

[alsa-devel] [PATCH] ASoC: intel - fix the card names
by Jaroslav Kysela 28 Oct '19

28 Oct '19

Those strings are exposed to the user space as the card name thus used in the GUIs. The common standard is to avoid '_' here. The worst case is 'sof-skl_hda_card' string. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a..d14ca327c684 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -623,7 +623,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 */ static struct snd_soc_card kabylake_audio_card = { - .name = "kbl_r5514_5663_max", + .name = "kbl-r5514-5663-max", .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 1778acdc367c..5261f7bf0b3b 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -90,7 +90,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) } static struct snd_soc_card hda_soc_card = { - .name = "skl_hda_card", + .name = "skl-hda", .owner = THIS_MODULE, .dai_link = skl_hda_be_dai_links, .dapm_widgets = skl_hda_widgets, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..fc4638b8e03b 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -370,7 +370,7 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { - .name = "sof_rt5682", + .name = "sof-rt5682", .owner = THIS_MODULE, .controls = sof_controls, .num_controls = ARRAY_SIZE(sof_controls), -- 2.20.1

2 2

[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in __snd_usbmidi_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of __snd_usbmidi_create() there is a memory leak caused by incorrect goto destination. Go to free_midi if snd_usbmidi_create_endpoints_midiman() or snd_usbmidi_create_endpoints() fail. Fixes: 731209cc0417 ("ALSA: usb-midi: Use common error handling code in __snd_usbmidi_create()") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b737f0ec77d0..22db37fbfbbd 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -2476,7 +2476,7 @@ int __snd_usbmidi_create(struct snd_card *card, else err = snd_usbmidi_create_endpoints(umidi, endpoints); if (err < 0) - goto exit; + goto free_midi; usb_autopm_get_interface_no_resume(umidi->iface); -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: SOF: ipc: Fix memory leak in sof_set_get_large_ctrl_data
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_set_get_large_ctrl_data() there is a memory leak in case an error. Release partdata if sof_get_ctrl_copy_params() fails. Fixes: 54d198d5019d ("ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/ipc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b2f359d2f7e5..086eeeab8679 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -572,8 +572,10 @@ static int sof_set_get_large_ctrl_data(struct snd_sof_dev *sdev, else err = sof_get_ctrl_copy_params(cdata->type, partdata, cdata, sparams); - if (err < 0) + if (err < 0) { + kfree(partdata); return err; + } msg_bytes = sparams->msg_bytes; pl_size = sparams->pl_size; -- 2.17.1

3 3

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda-stream: fix the CONFIG_ prefix missing
by Pierre-Louis Bossart 28 Oct '19

28 Oct '19

From: Keyon Jie <yang.jie(a)linux.intel.com> We are missing the 'CONFIG_' prefix when using the kernel configure item SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1, here correct them. Fixes: 43b2ab9009b13b ('ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture') Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 450f9c55785f..29ab43281670 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -190,7 +190,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) * Workaround to address a known issue with host DMA that results * in xruns during pause/release in capture scenarios. */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, @@ -228,7 +228,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) spin_unlock_irq(&bus->reg_lock); /* Enable DMI L1 entry if there are no capture streams open */ - if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) if (!active_capture_stream) snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: SOF: Fix memory leak in sof_dfsentry_write
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of sof_dfsentry_write() memory allocated for string is leaked in case of an error. Go to error handling path if the d_name.name is not valid. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/soc/sof/debug.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54cd431faab7..5529e8eeca46 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -152,8 +152,10 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, */ dentry = file->f_path.dentry; if (strcmp(dentry->d_name.name, "ipc_flood_count") && - strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) - return -EINVAL; + strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) { + ret = -EINVAL; + goto out; + } if (!strcmp(dentry->d_name.name, "ipc_flood_duration_ms")) flood_duration_test = true; -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: eve: implement set_bias_level function for rt5514
by Brent Lu 28 Oct '19

28 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). This rt5514 codec requires to control mclk directly in the set_bias_level function. Implement this function in machine driver to control the ssp1_mclk clock explicitly could fix this issue. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 50 ++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..b546de8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -653,6 +653,55 @@ static struct snd_soc_dai_link kabylake_dais[] = { }, }; +static int kabylake_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + if (!component || strcmp(component->name, RT5514_DEV_NAME)) + return 0; + + if (IS_ERR(priv->mclk)) + return 0; + + /* + * It's required to control mclk directly in the set_bias_level + * function for rt5514 codec or the recording function could + * break. + */ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_ON) { + dev_dbg(card->dev, "Disable mclk"); + clk_disable_unprepare(priv->mclk); + } else { + dev_dbg(card->dev, "Enable mclk"); + ret = clk_set_rate(priv->mclk, 24000000); + if (ret) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", + ret); + + /* mclk is already enabled in FW */ + ret = 0; + } + } + break; + default: + break; + } + + return ret; +} + static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); @@ -692,6 +741,7 @@ static struct snd_soc_card kabylake_audio_card = { .owner = THIS_MODULE, .dai_link = kabylake_dais, .num_links = ARRAY_SIZE(kabylake_dais), + .set_bias_level = kabylake_set_bias_level, .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, -- 2.7.4

4 6

[alsa-devel] [PATCH -next] ASoC: sunxi: sun4i-codec: remove unneeded semicolon
by YueHaibing 28 Oct '19

28 Oct '19

remove unneeded semicolon. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sunxi/sun4i-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5..34f3e0b 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1442,7 +1442,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1480,7 +1480,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) @@ -1518,7 +1518,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) if (!aux_dev.dlc.of_node) { dev_err(dev, "Can't find analog controls for codec.\n"); return ERR_PTR(-EINVAL); - }; + } card->dai_link = sun4i_codec_create_link(dev, &card->num_links); if (!card->dai_link) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: fsl: fsl_dma: fix build failure
by Michael Ellerman 28 Oct '19

28 Oct '19

Commit 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") removed fsl_dma_ops but left a usage, leading to a build error for some configs, eg. mpc85xx_defconfig: sound/soc/fsl/fsl_dma.c: In function ‘fsl_soc_dma_probe’: sound/soc/fsl/fsl_dma.c:905:18: error: ‘fsl_dma_ops’ undeclared (first use in this function) dma->dai.ops = &fsl_dma_ops; ^~~~~~~~~~~ Remove the usage to fix the build. Fixes: 4ac85de9977e ("ASoC: fsl: fsl_dma: remove snd_pcm_ops") Signed-off-by: Michael Ellerman <mpe(a)ellerman.id.au> --- sound/soc/fsl/fsl_dma.c | 1 - 1 file changed, 1 deletion(-) This breakage is only in linux-next. diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a092726510d4..2868c4f97cb2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -901,7 +901,6 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) } dma->dai.name = DRV_NAME; - dma->dai.ops = &fsl_dma_ops; dma->dai.open = fsl_dma_open; dma->dai.close = fsl_dma_close; dma->dai.ioctl = snd_soc_pcm_lib_ioctl; -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: rockchip: rockchip_max98090: Enable SHDN to fix headset detection
by Cheng-Yi Chiang 28 Oct '19

28 Oct '19

max98090 spec states that chip needs to be in turned-on state to supply mic bias. Enable SHDN dapm widget along with MICBIAS widget to actually turn on mic bias for proper headset button detection. This is similar to cht_ti_jack_event in sound/soc/intel/boards/cht_bsw_max98090_ti.c. Note that due to ts3a227e reports the jack event right away before the notifier is registered, if headset is plugged on boot, headset button will not get detected until headset is unplugged and plugged. This is still an issue to be fixed. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/rockchip/rockchip_max98090.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 0097df1fae66..e80b09143b63 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -66,10 +66,13 @@ static int rk_jack_event(struct notifier_block *nb, unsigned long event, struct snd_soc_jack *jack = (struct snd_soc_jack *)data; struct snd_soc_dapm_context *dapm = &jack->card->dapm; - if (event & SND_JACK_MICROPHONE) + if (event & SND_JACK_MICROPHONE) { snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - else + snd_soc_dapm_force_enable_pin(dapm, "SHDN"); + } else { snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SHDN"); + } snd_soc_dapm_sync(dapm); -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH] ASoC: ti: sdma-pcm: Add back the flags parameter for non standard dma names
by Peter Ujfalusi 28 Oct '19

28 Oct '19

When non standard names are used it is possible that one of the directions are not provided, thus the flags needs to be present to tell the core that we have half duplex setup. Fixes: 642aafea8889 ("ASoC: ti: remove compat dma probing") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/sdma-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index a236350beb10..2b0bc234e1b6 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -62,7 +62,7 @@ int sdma_pcm_platform_register(struct device *dev, config->chan_names[0] = txdmachan; config->chan_names[1] = rxdmachan; - return devm_snd_dmaengine_pcm_register(dev, config, 0); + return devm_snd_dmaengine_pcm_register(dev, config, flags); } EXPORT_SYMBOL_GPL(sdma_pcm_platform_register); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: rsnd: core.c: add WARN_ON() on rsnd_channel_normalization()
by Kuninori Morimoto 28 Oct '19

28 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> chan > 8 or chan < 0 shouldn't happen. This patch uses WARN_ON() for such case. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9242a0..295faad 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -302,7 +302,7 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, int rsnd_channel_normalization(int chan) { - if ((chan > 8) || (chan < 0)) + if (WARN_ON((chan > 8) || (chan < 0))) return 0; /* TDM Extend Mode needs 8ch */ -- 2.7.4

2 1

[alsa-devel] [PATCH V3] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 28 Oct '19

28 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitation in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 - refine comments Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 45 ++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..a3cfceea7d2f 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,15 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * The ideal ratio configuration can work with a flexible clock rate setting. + * Using IDEAL_RATIO_RATE gives a faster converting speed but overloads ASRC. + * For a regular audio playback, the clock rate should not be slower than an + * clock rate aligning with the output sample rate; For a use case requiring + * faster conversion, set use_ideal_rate to have the faster speed. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +275,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,27 +359,42 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + + /* + * The divider range is [1, 1024], defined by the hardware. For non- + * ideal ratio configuration, clock rate has to be strictly aligned + * with the sample rate. For ideal ratio configuration, clock rates + * only result in different converting speeds. So remainder does not + * matter, as long as we keep the divider within its valid range. + */ + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; } + div[IN] = min_t(u32, 1024, div[IN]); + clk = asrc_priv->asrck_clk[clk_index[OUT]]; - - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + /* Output divider has the same limitation as the input one */ + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +583,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V3] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 28 Oct '19

28 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v3 -refine comments Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..453392a9cd68 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock between hw_reset() and trigger() * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect hw_reset and trigger */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_korg1212_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_korg1212_create() the allocated memory for korg1212 is leaked in cases of error. Release korg1212 via snd_korg1212_free() if either of these calls fail: snd_korg1212_downloadDSPCode(), snd_pcm_new(), or snd_ctl_add(). Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/korg1212/korg1212.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 0d81eac0a478..e976e857d915 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2367,8 +2367,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, mdelay(CARD_BOOT_DELAY_IN_MS); - if (snd_korg1212_downloadDSPCode(korg1212)) + if (snd_korg1212_downloadDSPCode(korg1212)) { + snd_korg1212_free(korg1212); return -EBUSY; + } K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " "PlayDataPhy = %08x L[%08x]\n" @@ -2383,8 +2385,11 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); - if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) + err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm); + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } korg1212->pcm->private_data = korg1212; korg1212->pcm->private_free = snd_korg1212_free_pcm; @@ -2398,8 +2403,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, for (i = 0; i < ARRAY_SIZE(snd_korg1212_controls); i++) { err = snd_ctl_add(korg1212->card, snd_ctl_new1(&snd_korg1212_controls[i], korg1212)); - if (err < 0) + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } } snd_korg1212_proc_init(korg1212); -- 2.17.1

3 7

[alsa-devel] KASAN: slab-out-of-bounds Read in build_audio_procunit
by syzbot 28 Oct '19

28 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=103893cf600000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=0620f79a1978b1133fd7 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16fd6a40e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1033215f600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+0620f79a1978b1133fd7(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: slab-out-of-bounds in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: slab-out-of-bounds in build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 Read of size 1 at addr ffff8881d537c7b7 by task kworker/0:2/101 CPU: 0 PID: 101 Comm: kworker/0:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 parse_audio_processing_unit sound/usb/mixer.c:2468 [inline] parse_audio_unit+0x17e9/0x36f0 sound/usb/mixer.c:2757 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 101: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x314/0x3050 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 17: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 get_port_status drivers/usb/core/hub.c:569 [inline] hub_ext_port_status+0x125/0x460 drivers/usb/core/hub.c:586 hub_port_status drivers/usb/core/hub.c:608 [inline] hub_activate+0x497/0x1570 drivers/usb/core/hub.c:1111 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d537c700 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 183 bytes inside of 192-byte region [ffff8881d537c700, ffff8881d537c7c0) The buggy address belongs to the page: page:ffffea000754df00 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000754d680 0000000600000006 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d537c680: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d537c700: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > ffff8881d537c780: 00 00 00 02 fc fc fc fc fc fc fc fc fc fc fc fc ^ ffff8881d537c800: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8881d537c880: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 3

[alsa-devel] [PATCH] ASoC: change 'HDMI/DP, pcm=' to 'HDMI/DP, pcm=' Jack control names
by Jaroslav Kysela 28 Oct '19

28 Oct '19

There is an inconsistency in the names for the HDMI/DP Jack control names between some ASoC drivers and the HDA HDMI driver which introduced this naming in 2011. There might be an impact for the user space (UCM). I will fix the UCM configurations when this patch is applied. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Documentation/sound/designs/control-names.rst | 5 +++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 13 files changed, 15 insertions(+), 14 deletions(-) diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst index 7fedd0f33cd9..a5f6e3842df8 100644 --- a/Documentation/sound/designs/control-names.rst +++ b/Documentation/sound/designs/control-names.rst @@ -50,7 +50,7 @@ Internal internal SOURCE ~~~~~~ -=================== ================================================= +=================== =================================================== Master Master Mono Hardware Master @@ -92,7 +92,8 @@ SPDIF output only SPDIF In Digital In HDMI/DP either HDMI or DisplayPort -=================== ================================================= +HDMI/DP,pcm=<num> either HDMI or DisplayPort with the PCM device link +=================== =================================================== Exceptions (deprecated) ----------------------- diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5f9d11..b49028c6fd10 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -618,7 +618,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index adf416a49b48..caaf2f2af287 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -530,7 +530,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index bd2d371f2acd..5474f28e0c31 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -548,7 +548,7 @@ static int glk_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &geminilake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..367415efed9b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -548,7 +548,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 829f95fc4179..b30c2148d1f4 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -977,7 +977,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &kabylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 74fe1f3a5479..0965d1f02b2c 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -470,7 +470,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7cefda341fbf..9f420e978459 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -888,7 +888,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 58409b6e476e..f717b1d179bd 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -139,7 +139,7 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &pcm->hdmi_jack, NULL, 0); diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ce8efbeed12..b009e2df25d3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -607,7 +607,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 1a7ac8bdf543..0a6e650dc698 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -648,7 +648,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 231349a47cc9..56ee4b55dce9 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -489,7 +489,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..862a7e35cb5c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -277,7 +277,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &sof_hdmi[i], NULL, 0); -- 2.20.1

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[alsa-devel] [PATCH] ALSA: hda: Allow non-Intel device probe gracefully
by Takashi Iwai 28 Oct '19

28 Oct '19

The recent addition of snd_intel_dsp_driver_probe() check caused a spurious kernel warning when the driver is loaded for a non-Intel hardware due to snd_BUG_ON(). Moreover, for such a hardware, we should always return SND_INTEL_DSP_DRIVER_ANY, not check the dsp_driver option at all. This patch fixes these issues for non-Intel devices. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/intel-dsp-config.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 0b2a7201a46a..be1df80ed013 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,13 +307,13 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; - if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) - return dsp_driver; - /* Intel vendor only */ - if (snd_BUG_ON(pci->vendor != 0x8086)) + if (pci->vendor != 0x8086) return SND_INTEL_DSP_DRIVER_ANY; + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + /* * detect DSP by checking class/subclass/prog-id information * class=04 subclass 03 prog-if 00: no DSP, use legacy driver -- 2.16.4

1 0

[alsa-devel] Questions about max98090 codec driver
by Tzung-Bi Shih 28 Oct '19

28 Oct '19

Hi, I am studying an odd issue of max98090 codec on Baytrail-based chromebook. The issue is: when user playback and capture simultaneously, it seems the PLL never get locked if msleep(10) between the SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). The playback stream becomes silent and the console keeps printing "PLL unlocked". But, if comment out the msleep(10) between the SHDN off and on, the issue fixed. I am trying to find the reason but facing further more questions and may need your inputs. 1. The commit b8a3ee820f7b ("ASoC: max98090: Add recovery for PLL lock failure") enables ULK interrupts (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) when PCM stream starting. If max98090 claims its PLL is unlocked, max98090_pll_work() get scheduled to workaround it by SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). I feel it is weird to sleep in max98090_pll_work(). Especially, at this line https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.… (it makes less sense to "wait" in another thread). Note that, the threaded IRQF_ONESHOT handler and max98090_pll_work() are in 2 different threads. I guess the original intention is: - disable ULK interrupt in IRQ handler (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) - schedule max98090_pll_work() to workaround it - wait 10ms to give PLL chance to lock - enable ULK interrupt again If max98090 claims its PLL is unlocked again, repeat the above by receiving another ULK interrupt. Unfortunately, the odd issue seems not be fixed by my rough implementation of these. 2. According to the datasheet page 164 table 90 (https://datasheets.maximintegrated.com/en/ds/MAX98090.pdf), there are some registers should only be adjusted when SHDN==0. But I fail to find max98090.c tries to set SHDN to 0 and restore it afterwards when writing to these registers. For example, https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…. I am wondering if it would bring any side effects because the datasheet states "Changing these settings during normal operation (SHDN=1) can compromise device stability and performance specifications." 3. By searching some history data, I found a previous version did not have the msleep(10) between the SHDN off and on (https://crrev.com/c/191740, click the file name in the middle of the window to see the diff. Pardon me, I do not find another public repository for this). I am curious if anyone of you still remember why the upstream version contains the msleep(10). I am also curious if anyone of your environment works well with the upstream version max98090.c.

4 5

Re: [alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by S.j. Wang 28 Oct '19

28 Oct '19

Hi > > On Fri, Oct 25, 2019 at 03:13:53PM +0800, Shengjiu Wang wrote: > > xrun may happen at the end of stream, the > > trigger->fsl_esai_trigger_stop maybe called in the middle of > > fsl_esai_hw_reset, this may cause esai in wrong state after stop, and > > there may be endless xrun interrupt. > > > > This issue may also happen with trigger->fsl_esai_trigger_start. > > > > So Add spin lock to lock those functions. > > > > Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after > > xrun") > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > Some small comments inline. Once they are addressed, please add: > > Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > > Thanks > > > --- > > Change in v2 > > -add lock for fsl_esai_trigger_start. > > > > sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ > > 1 file changed, 12 insertions(+) > > > > diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index > > 37b14c48b537..9b28e2af26e4 100644 > > --- a/sound/soc/fsl/fsl_esai.c > > +++ b/sound/soc/fsl/fsl_esai.c > > @@ -33,6 +33,7 @@ > > * @fsysclk: system clock source to derive HCK, SCK and FS > > * @spbaclk: SPBA clock (optional, depending on SoC design) > > * @task: tasklet to handle the reset operation > > + * @lock: spin lock to handle reset and stop behavior > > Should be "between hw_reset() and trigger()" now. > > > * @fifo_depth: depth of tx/rx FIFO > > * @slot_width: width of each DAI slot > > * @slots: number of slots > > @@ -56,6 +57,7 @@ struct fsl_esai { > > struct clk *fsysclk; > > struct clk *spbaclk; > > struct tasklet_struct task; > > + spinlock_t lock; /* Protect reset and stop */ > > We can drop the comments here since you add it to the top. Here is required by the checkpatch.pl, so still is needed. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_ali_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_ali_create() the memory allocated for codec is leaked in case of an error. Release codec if snd_ali_chip_init() fails. Fixes: f9ab2b1c3ab5 ("[ALSA] ali5451 - Code clean up, irq handler fix") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6e28e381c21a..3ed07a18be4d 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2179,6 +2179,7 @@ static int snd_ali_create(struct snd_card *card, err = snd_ali_chip_init(codec); if (err < 0) { dev_err(card->dev, "ali create: chip init error.\n"); + snd_ali_free(codec); return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: lx6464es: Fix memory leaks in snd_lx6464es_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_lx6464es_create() it there are memory leaks when error happens. Go to error path if any of these calls fail: lx_init_dsp(), lx_pcm_create(), lx_proc_create(), snd_ctl_add(). Fixes: 02bec4904508 ("ALSA: lx6464es - driver for the digigram lx6464es interface") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/lx6464es/lx6464es.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index fe10714380f2..7c6e8f4ef826 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1020,25 +1020,26 @@ static int snd_lx6464es_create(struct snd_card *card, err = lx_init_dsp(chip); if (err < 0) { dev_err(card->dev, "error during DSP initialization\n"); - return err; + goto cleanup; } err = lx_pcm_create(chip); if (err < 0) - return err; + goto cleanup; err = lx_proc_create(card, chip); if (err < 0) - return err; + goto cleanup; err = snd_ctl_add(card, snd_ctl_new1(&lx_control_playback_switch, chip)); if (err < 0) - return err; + goto cleanup; *rchip = chip; return 0; +cleanup: device_new_failed: free_irq(pci->irq, chip); -- 2.17.1

3 2

[alsa-devel] [PATCH -next] ASoC: SOF: select SND_INTEL_DSP_CONFIG in SND_SOC_SOF_PCI
by Mao Wenan 28 Oct '19

28 Oct '19

When SND_SOC_SOF_PCI=y, and SND_INTEL_DSP_CONFIG=m, below errors can be seen: sound/soc/sof/sof-pci-dev.o: In function `sof_pci_probe': sof-pci-dev.c:(.text+0xb9): undefined reference to `snd_intel_dsp_driver_probe' After commit 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code"), sof_pci_probe() will call snd_intel_dsp_driver_probe(), so it should select SND_INTEL_DSP_CONFIG in Kconfig SND_SOC_SOF_PCI. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 56a3ab6..a9b2be2 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_SOF_PCI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_PCI if SND_SOC_SOF_INTEL_TOPLEVEL + select SND_INTEL_DSP_CONFIG help This adds support for PCI enumeration. This option is required to enable Intel Skylake+ devices -- 2.7.4

2 1

[alsa-devel] [PATCH] Fix JACK plugin buffer sizes
by John McKay 27 Oct '19

27 Oct '19

I was having a conversation on the Archlinux forums[1] about using the alsa-plugins JACK plugin. While one of the bugs that we found has already been fixed in the git alsa-lib, there is another bug relating to buffer sizes. Currently snd_pcm_hw_params_set_buffer_size_near() doesn't really work when you are using this plugin unless you happen to request a buffer size that almost exactly matches a whole multiple of periods that the JACK is asking for. After some debugging, I discovered the reason is that some plugins, notably the 'plug' plugin that most everybody is going to use to change frequency/format directly attempt to manipulate the buffer size. This causes the code in pcm_ioplug.c to fail because it cannot find a number of periods/period time that can fit the buffer. The included patch stops this by specifying the list of acceptable buffer sizes as well, so that when other plugins try to change the buffer it always picks one that will work. It does have some problems in the JACK plugin now offers a number of periods/sizes that don't exist but in practice asking for huge buffers just seems to give you a more sane value. [1] https://bbs.archlinux.org/viewtopic.php?id=250116 --- jack/pcm_jack.c 2019-05-10 06:57:24.000000000 +0000 +++ jack/pcm_jack.c 2019-10-27 03:43:36.053734268 +0000 @@ -428,6 +428,7 @@ unsigned int format = SND_PCM_FORMAT_FLOAT; unsigned int rate = jack_get_sample_rate(jack->client); unsigned int psize_list[MAX_PERIODS_MULTIPLE]; + unsigned int bsize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * jack->num_ports); @@ -441,6 +442,9 @@ for (i = 1; i <= ARRAY_SIZE(psize_list); i++) psize_list[i-1] = jack_buffer_bytes * i; + for (i = 1; i <= ARRAY_SIZE(bsize_list); i++) + bsize_list[i-1] = jack_buffer_bytes * (i + 1); + jack->sample_bits = snd_pcm_format_physical_width(format); if ((err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list)) < 0 || @@ -452,6 +456,8 @@ rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, ARRAY_SIZE(psize_list), psize_list)) < 0 || + (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bsize_list), bsize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err;

1 0

[alsa-devel] [PATCH] ALSA: bebob: Fix prototype of helper function to return negative value
by Takashi Sakamoto 26 Oct '19

26 Oct '19

A helper function of ALSA bebob driver returns negative value in a function which has a prototype to return unsigned value. This commit fixes it by changing the prototype. Fixes: eb7b3a056cd8 ("ALSA: bebob: Add commands and connections/streams management" Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 7ac0d9f495c4..f7f0db5aa811 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; -- 2.20.1

2 1

[alsa-devel] [PATCH v2 00/12] Add Samus Hotwording for RT5677
by Curtis Malainey 26 Oct '19

26 Oct '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Changes since v1: * Addressed Mark's and Kuninori's comments * Adjusted DAI Links * Squashed related patches * Removed all kcontrol writes * Addressed race condition with dapm * Added firmware boot check * Moved all clocking to driver API * Moved DAPM routes that spanned drivers to machine driver Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Ben Zhang <benzh(a)chromium.org> Ben Zhang (8): ASoC: rt5677: Add a PCM device for streaming hotword via SPI ASoC: rt5677: Load firmware via SPI using delayed work ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Disable irq at suspend Curtis Malainey (4): ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM ASoC: rt5677: Set ADC clock to use PLL and enable ASRC ASoC: rt5677: Wait for DSP to boot before loading firmware sound/soc/codecs/rt5677-spi.c | 397 ++++++++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 1 + sound/soc/codecs/rt5677.c | 447 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 11 +- sound/soc/intel/boards/bdw-rt5677.c | 51 ++++ 5 files changed, 834 insertions(+), 73 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

4 19

[alsa-devel] [PATCH V2] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 26 Oct '19

26 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..89cf333154c7 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,11 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * use_ideal_rate = true is need by some case which need higher performance. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +271,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,8 +355,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; @@ -360,18 +366,29 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + /* + * Output rate should be align with the out samplerate. If set too + * high output rate, there will be lots of Overload. + * But some case need higher performance, then we can use + * IDEAL_RATIO_RATE specifically for such case. + */ + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +577,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 26 Oct '19

26 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..9b28e2af26e4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v2] ASoC: ti: omap-mcpdm: Add support for pdmclk clock handling
by Peter Ujfalusi 25 Oct '19

25 Oct '19

McPDM module receives it's functional clock from external source. This clock is the pdmclk provided by the twl6040 audio IC. If the clock is not available all register accesses to McPDM fails and the module is not operational. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Hi, this has been lurking in my next-wip branch for some time... It might not needed anymore as I have not heard about issues with McPDM for a while. It was dropped back then because some core parts were not picked in time and this caused some issues, but can not find the exact reason Regards, Peter sound/soc/ti/omap-mcpdm.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index b8c8290265c7..2c536947b69f 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -17,6 +17,7 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -42,6 +43,7 @@ struct omap_mcpdm { int irq; struct pm_qos_request pm_qos_req; int latency[2]; + struct clk *pdmclk; struct mutex mutex; @@ -416,6 +418,10 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; + pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -454,6 +460,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) if (pm_qos_request_active(&mcpdm->pm_qos_req)) pm_qos_remove_request(&mcpdm->pm_qos_req); + clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -473,12 +480,19 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } + clk_disable_unprepare(mcpdm->pdmclk); + return 0; } static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) @@ -573,6 +587,15 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; + mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); + if (IS_ERR(mcpdm->pdmclk)) { + if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", + PTR_ERR(mcpdm->pdmclk)); + mcpdm->pdmclk = NULL; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] sof-skl_hda: move hdmi as separate use case
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #6 was opened from juimonen: Move hdmi definitions as separate use case and conf file. This will enable Pulseaudio to create separate profile for hdmi and make routing much more fluent than with single profile. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] sof-skl_hda: fix headphone jack name
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #5 was opened from juimonen: Change "Headphone Jack" to "Headphone Mic Jack". Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 25 Oct '19

25 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. Thus far for the agenda we have: - Use case management enhancements (Curtis) - DSP framework integration (Liam?) - Soundwire status (Pierre?) - Componentisation status/plans (Me) As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

8 19

[alsa-devel] [PATCH v5 1/2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 25 Oct '19

25 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 5 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 322 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..601e617918b8 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,6 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <linux/bits.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/intel-dsp-config.h> +#include <sound/intel-nhlt.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a916963822e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2021,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2038,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

3 6

[alsa-devel] [PATCH -next] ALSA: intel_hdmi: Remove dev_err() on platform_get_irq() failure
by YueHaibing 25 Oct '19

25 Oct '19

platform_get_irq() will call dev_err() itself on failure, so there is no need for the driver to also do this. This is detected by coccinelle. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/x86/intel_hdmi_audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 5fd4e32..cd389d2 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1708,10 +1708,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* get resources */ irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Could not get irq resource: %d\n", irq); + if (irq < 0) return irq; - } res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res_mmio) { -- 2.7.4

2 1

[alsa-devel] [PATCH][next] ASoC: rt5677: Add missing null check for failed allocation of rt5677_dsp
by Colin King 25 Oct '19

25 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> The allocation of rt5677_dsp can potentially fail and return null, so add a null check and return -ENOMEM on a memory allocation failure. Addresses-Coverity: ("Dereference null return") Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5677-spi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..3a17643fcd9f 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -376,6 +376,8 @@ static int rt5677_spi_pcm_probe(struct snd_soc_component *component) rt5677_dsp = devm_kzalloc(component->dev, sizeof(*rt5677_dsp), GFP_KERNEL); + if (!rt5677_dsp) + return -ENOMEM; rt5677_dsp->dev = &g_spi->dev; mutex_init(&rt5677_dsp->dma_lock); INIT_DELAYED_WORK(&rt5677_dsp->copy_work, rt5677_spi_copy_work); -- 2.20.1

3 2

[alsa-devel] [PATCH trivial] ASoC: Spelling s/configr/configur/
by Geert Uytterhoeven 25 Oct '19

25 Oct '19

Fix misspellings of "configuration" and "configure". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- include/sound/wm8904.h | 2 +- sound/soc/codecs/cx2072x.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 14074405f5012f9a..88ac1870510ec063 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -120,7 +120,7 @@ * DRC configurations are specified with a label and a set of register * values to write (the enable bits will be ignored). At runtime an * enumerated control will be presented for each DRC block allowing - * the user to choose the configration to use. + * the user to choose the configuration to use. * * Configurations may be generated by hand or by using the DRC control * panel provided by the WISCE - see http://www.wolfsonmicro.com/wisce/ diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index 1c1ba7bea4d81969..2ad00ed21bec6c59 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1507,7 +1507,7 @@ static int cx2072x_probe(struct snd_soc_component *codec) regmap_multi_reg_write(cx2072x->regmap, cx2072x_reg_init, ARRAY_SIZE(cx2072x_reg_init)); - /* configre PortC as input device */ + /* configure PortC as input device */ regmap_update_bits(cx2072x->regmap, CX2072X_PORTC_PIN_CTRL, 0x20, 0x20); -- 2.17.1

2 1

[alsa-devel] [PATCH trivial] ASoC: pxa: poodle: Spelling s/enpoints/endpoints/, s/connetion/connection/
by Geert Uytterhoeven 25 Oct '19

25 Oct '19

Fix misspelling of "endpoints" and "connection". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/pxa/poodle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 48d5c2252b10081d..59ef04d0467a622d 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -56,7 +56,7 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); } - /* set the enpoints to their new connetion states */ + /* set the endpoints to their new connection states */ if (poodle_spk_func == POODLE_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: rt5677-spi: fix sparse warnings
by Curtis Malainey 25 Oct '19

25 Oct '19

Fix bugs reported by kbuild test robot Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..958871b342ea 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -145,7 +145,7 @@ static int rt5677_spi_hw_free( snd_soc_component_get_drvdata(component); mutex_lock(&rt5677_dsp->dma_lock); - rt5677_dsp->substream = 0; + rt5677_dsp->substream = NULL; mutex_unlock(&rt5677_dsp->dma_lock); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -362,7 +362,7 @@ static void rt5677_spi_copy_work(struct work_struct *work) mutex_unlock(&rt5677_dsp->dma_lock); } -struct page *rt5677_spi_pcm_page( +static struct page *rt5677_spi_pcm_page( struct snd_soc_component *component, struct snd_pcm_substream *substream, unsigned long offset) -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH v2] ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro
by Kuninori Morimoto 25 Oct '19

25 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> for_each_dpcm_xx() macro is using "dpcm" as parameter (1), but, it is also struct member (2). #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream]...) ^^^^(1) ^^^^(2) Thus, it will be compile error if user not used "dpcm" as parameter for_each_dpcm_fe(be, stream, dp) ^^ This patch fixup it. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - based on mark/for-5.5 branch include/sound/soc-dpcm.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index e55aeb0..b654ebf 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,15 +103,15 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; -#define for_each_dpcm_fe(be, stream, dpcm) \ - list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) - -#define for_each_dpcm_be(fe, stream, dpcm) \ - list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ - list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_rollback(fe, stream, dpcm) \ - list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_fe(be, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) + +#define for_each_dpcm_be(fe, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, _dpcm, __dpcm) \ + list_for_each_entry_safe(_dpcm, __dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_rollback(fe, stream, _dpcm) \ + list_for_each_entry_continue_reverse(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, -- 2.7.4

2 1

[alsa-devel] [PATCH v2 0/2] ASoC: SOF: make hw_info platform specific
by Pierre-Louis Bossart 25 Oct '19

25 Oct '19

The SOF core blindly sets the hw_info, when it should really be platform-specific. This move enable us to temporarily fix PulseAudio issues resulting from inaccuracies in the hw_ptr position provided by the firmware on Baytrail-Broadwell. This is the first time we experiment with the GitHub low-level API to automagically keep track of 'approvers' by inserting a Reviewed-by tag when merging a pull-request, thanks to Ranjani for her initial work and bearing with my habit of breaking things :-) v2: make the code apply on Mark's tree. Sorry about that bad patch. Pierre-Louis Bossart (2): ASoC: SOF: define INFO_ flags in dsp_ops ASoC: SOF: Intel: only support INFO_BATCH for legacy platforms sound/soc/sof/imx/imx8.c | 7 +++++++ sound/soc/sof/intel/apl.c | 7 +++++++ sound/soc/sof/intel/bdw.c | 9 ++++++++- sound/soc/sof/intel/byt.c | 21 +++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 7 +++++++ sound/soc/sof/pcm.c | 8 +++----- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 56 insertions(+), 6 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early for rt5514
by Brent Lu 25 Oct '19

25 Oct '19

The first DMIC capture always fail (zero sequence data from PCM port) after using DSP hotwording function (i.e. Google assistant). The DMIC is attached to rt5514 which also require eaily mclk/sclk like rt5663. Therefore we add a dapm route to provide ssp mclk/sclk early to rt5514. The rt5514 is attached to SSP0 and the rt5663 is attached to SSP1 so we create another supply widget to provide SSP0 mclk/sclk to rt5514. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 136 ++++++++++++++------- 1 file changed, 92 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index dc09a85..1755efa 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -53,8 +53,10 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; - struct clk *mclk; - struct clk *sclk; + struct clk *ssp0_mclk; + struct clk *ssp0_sclk; + struct clk *ssp1_mclk; + struct clk *ssp1_sclk; }; enum { @@ -77,13 +79,31 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *mclk, *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + mclk = priv->ssp0_mclk; + sclk = priv->ssp0_sclk; + sclk_rate = 6144000; + break; + case 1: + mclk = priv->ssp1_mclk; + sclk = priv->ssp1_sclk; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -92,37 +112,39 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(mclk); if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + dev_err(card->dev, "Can't enable ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't set rate for ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); + ret = clk_prepare_enable(sclk); if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't enable ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + clk_disable_unprepare(mclk); + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +153,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +173,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +192,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +216,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -704,6 +741,29 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + return ret; + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -724,33 +784,21 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_mclk", &ctx->ssp0_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->ssp0_sclk); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->ssp1_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->ssp1_sclk); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 2

Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by S.j. Wang 25 Oct '19

25 Oct '19

Hi > > On Wed, Oct 23, 2019 at 06:25:20AM +0000, S.j. Wang wrote: > > > On Thu, Oct 17, 2019 at 02:21:08PM +0800, Shengjiu Wang wrote: > > > > For P2P output, the output divider should align with the output > > > > sample > > > > > > I think we should avoid "P2P" (or "M2M") keyword in the mainline > > > code as we know M2M will never get merged while somebody working > > > with the mainline and caring about new feature might be confused. > > > > Ok. But we still curious that is there a way to upstream m2m? > > Hmm..I would love to see that happening. Here is an old discussion that > you may want to take a look: > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fmail > man.alsa-project.org%2Fpipermail%2Falsa-devel%2F2014- > May%2F076797.html&data=02%7C01%7Cshengjiu.wang%40nxp.com%7 > Ce902d2bac4254d2faa0f08d757ecac0e%7C686ea1d3bc2b4c6fa92cd99c5c301 > 635%7C0%7C0%7C637074546320396681&sdata=bg%2BLwRQnUPhW8f > mE972O%2F53MyVftJkK140PSnmC%2FDKQ%3D&reserved=0 > > > > It makes sense to me, yet I feel that the delay at the beginning of > > > the audio playback might be longer as a compromise. I am okay with > > > this decision though... > > > > > > > The maximum divider of asrc clock is 1024, but there is no > > > > judgement for this limitaion in driver, which may cause the > > > > divider setting not correct. > > > > > > > > For non-ideal ratio mode, the clock rate should divide the sample > > > > rate with no remainder, and the quotient should be less than 1024. > > > > > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > > > > @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct > > > > fsl_asrc_pair > > > *pair) > > > > /* We only have output clock for ideal ratio mode */ > > > > clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; > > > > > > > > - div[IN] = clk_get_rate(clk) / inrate; > > > > + clk_rate = clk_get_rate(clk); > > > > > > The fsl_asrc.c file has config.inclk being set to INCLK_NONE and > > > this sets the "ideal" in this function to true. So, although we tend > > > to not use ideal ratio setting for p2p cases, yet the input clock is > > > still not physically connected, so we still use output clock for div[IN] > calculation? > > > > For p2p case, it can be ideal or non-ideal. For non-ideal, we still > > use Output clock for div calculation. > > > > > > > > I am thinking something simplier: if we decided not to use ideal > > > ratio for "P2P", instead of adding "bool p2p" with the confusing > > > "ideal" in this function, could we just set config.inclk to the same > > > clock as the output one for "P2P"? By doing so, "P2P" won't go > > > through ideal ratio mode while still having a clock rate from the output > clock for div[IN] calculation here. > > > > Bool p2p is to force output rate to be sample rate, no impact to ideal > > Ratio mode. > > I just realized that the function has a bottom part for ideal mode > exclusively -- if we treat p2p as !ideal, those configurations will be missing. > So you're right, should have an extra boolean variable. > > > > > > > > + rem[IN] = do_div(clk_rate, inrate); > > > > + div[IN] = (u32)clk_rate; > > > > if (div[IN] == 0) { > > > > > > Could we check div[IN] and rem[IN] here? Like: > > > if (div[IN] == 0 || div[IN] > 1024) { > > > pair_err(); > > > goto out; > > > } > > > > > > if (!ideal && rem[IN]) { > > > pair_err(); > > > goto out; > > > } > > > > > > According to your commit log, I think the max-1024 limitation should > > > be applied to all cases, not confined to "!ideal" cases right? And > > > we should add some comments also, indicating it is limited by hardware. > > > > For ideal mode, my test result is the divider not impact the output result. > > Which means it is ok for ideal mode even divider is not correct... > > OK. > > > > > > > > pair_err("failed to support input sample rate %dHz > > > > by > > > asrck_%x\n", > > > > inrate, clk_index[ideal ? OUT : > > > > IN]); @@ > > > > -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct > > > > fsl_asrc_pair *pair) > > > > > > > > clk = asrc_priv->asrck_clk[clk_index[OUT]]; > > > > > > > > - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ > > > > - if (ideal) > > > > - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; > > > > - else > > > > - div[OUT] = clk_get_rate(clk) / outrate; > > > > + /* > > > > + * When P2P mode, output rate should align with the out > samplerate. > > > > + * if set too high output rate, there will be lots of Overload. > > > > + * When M2M mode, output rate should also need to align with > > > > + the out > > > > > > For this "should", do you actually mean "M2M could also"? Sorry, I'm > > > just trying to understand everyting here, not intentionally being picky at > words. > > > My understanding is that we still keep the ideal ratio setting > > > because "M2M" still uses it. > > > > We use IDEAL_RATIO_RATE as output rate for m2m mode, it likes a Tricky > > operation, in order to improve the performance. I think The correct > > operation is to use the real output rate, but the performance Is bad. > > So it is a compromise. > > I see. > > > > > > > > + * samplerate, but M2M must use less time to achieve good > > > performance. > > > > + */ > > > > + clk_rate = clk_get_rate(clk); > > > > + if (p2p || !ideal) { > > > > + rem[OUT] = do_div(clk_rate, outrate); > > > > + div[OUT] = clk_rate; > > > > + } else { > > > > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > > > > + div[OUT] = clk_rate; > > > > + } > > > > > > > > if (div[OUT] == 0) { > > > > pair_err("failed to support output sample rate %dHz > > > > by asrck_%x\n", @@ -372,6 +384,16 @@ static int > > > > fsl_asrc_config_pair(struct > > > fsl_asrc_pair *pair) > > > > return -EINVAL; > > > > } > > > > > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || > > > > + rem[IN] != 0 || rem[OUT] != 0)) { > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || rem[IN] > > > > + || rem[OUT] != 0)) { > > > > > > So for ideal == true, these limitaions are not applied any more? > > > Remember that the "ideal" is true for "p2p == true" cases here. > > > > No, not applied. for ideal, the div don't impact the output result > > Even it is not accurate. > > I see. > > > > > > > > + pair_err("The divider can't be used for non ideal mode\n"); > > > > + return -EINVAL; > > > > + } > > > > + > > > > + /* Divider range is [1, 1024] */ > > > > + div[IN] = min_t(u32, 1024, div[IN]); > > > > + div[OUT] = min_t(u32, 1024, div[OUT]); > > > > > > Hmm, this looks like we want to allow ideal ratio cases and p2p > > > cases to operate any way, even if the divider wasn't within the > > > range to get the in/out rates from the output clock? > > > > Yes. We still allow the p2p = true, ideal = false. Note that p2p is > > not Equal to ideal. > > Got it. > > Overall, I feel it's better to have a naming to state the purpose of using > ideal configurations without the IDEAL_RATIO_RATE setup. > bool use_ideal_rate; > And we can put into the asrc_config structure if there's no major problem. > Asrc_config may exposed to user, I don't think user need to care about The using of ideal rate or not. > So the condition check for the calculation would be: > + if (ideal && config->use_ideal_rate) > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > + else > + rem[OUT] = do_div(clk_rate, outrate); > + div[OUT] = clk_rate; > > And for that if (!ideal && div[IN]....rem[OUT]), I feel it would be clear to > have them separately, as the existing "div[IN] == 0" > and "div[OUT] == 0" checks, so that we can tell users which side of the > divider is out of range and what the sample rate and clock rate are. > Do you mean need to combine this judgement with "div[IN] == 0" Or "div[OUT] == 0"? Best regards Wang shengjiu

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl_esai: Add spin lock to protect reset and stop
by S.j. Wang 25 Oct '19

25 Oct '19

Hi > > On Wed, Oct 23, 2019 at 03:29:49PM +0800, Shengjiu Wang wrote: > > xrun may happen at the end of stream, the > > trigger->fsl_esai_trigger_stop maybe called in the middle of > > fsl_esai_hw_reset, this may cause esai in wrong state after stop, and > > there may be endless xrun interrupt. > > What about fsl_esai_trigger_start? It touches ESAI_xFCR_xFEN bit that is > being checked in the beginning of fsl_esai_hw_reset. > > Could the scenario below be possible also? > > 1) ESAI TX starts > 2) Xrun happens to TX > 3) Starting fsl_esai_hw_reset (enabled[TX] = true; enabled[RX] = false) > 4) ESAI RX starts > 5) Finishing fsl_esai_hw_reset (enabled[RX] is still false) > > Good catch, this may possible. Will update in v2. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH v2 00/22] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 25 Oct '19

25 Oct '19

Hi Mark These are v2 of soc-core cleanup. These are related to topology, but according to Intel CI, there is no damage to it. Some of them need more review. I added some mark to such patch. I merged posted-but-not-yet-accepted patch on top of it. Kuninori Morimoto (22): ASoC: soc-pcm: fixup dpcm_prune_paths() loop continue ASoC: soc-core: remove for_each_rtdcom_safe() ASoC: soc-core: add for_each_rtd_components() and replace ASoC: soc-core: snd_soc_unbind_card() cleanup ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: tidyup soc_init_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component_unlocked() ASoC: soc-core: remove snd_soc_component_add/del() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 25 ++- sound/soc/soc-component.c | 43 +--- sound/soc/soc-compress.c | 52 ++--- sound/soc/soc-core.c | 561 ++++++++++++++++++++-------------------------- sound/soc/soc-pcm.c | 55 ++--- sound/soc/soc-topology.c | 17 +- 6 files changed, 322 insertions(+), 431 deletions(-) -- 2.7.4

2 28

[alsa-devel] [PATCH] ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro
by Kuninori Morimoto 25 Oct '19

25 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> for_each_dpcm_xx() macro is using "dpcm" as parameter (1), but, it is also struct member (2). #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream]...) ^^^^(1) ^^^^(2) Thus, it will be compile error if user not used "dpcm" as parameter for_each_dpcm_fe(be, stream, dp) ^^ This patch fixup it. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dpcm.h | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ff53037..fbe9e3d 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,13 +103,13 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; -#define for_each_dpcm_fe(be, stream, dpcm) \ - list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) +#define for_each_dpcm_fe(be, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) -#define for_each_dpcm_be(fe, stream, dpcm) \ - list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ - list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be(fe, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, _dpcm, __dpcm) \ + list_for_each_entry_safe(_dpcm, __dpcm, &(fe)->dpcm[stream].be_clients, list_be) /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, -- 2.7.4

2 2

[alsa-devel] [PATCH v4 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 25 Oct '19

25 Oct '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- V3 Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Changes from V3: repost with more recipients and edits to commit messages Changes from V2: re ordered match table to ensure non i2c device matches are at the end Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 1.9.1

3 5

[alsa-devel] KASAN: use-after-free Read in build_audio_procunit
by syzbot 24 Oct '19

24 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=17cdccc4e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=fd965c77e4711eb13b82 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=13fe752f600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17989fd8e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+fd965c77e4711eb13b82(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: use-after-free in uac_extension_unit_iExtension include/uapi/linux/usb/audio.h:483 [inline] BUG: KASAN: use-after-free in build_audio_procunit+0xeab/0x13f0 sound/usb/mixer.c:2434 Read of size 1 at addr ffff8881d4c0140d by task kworker/0:1/12 CPU: 0 PID: 12 Comm: kworker/0:1 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_extension_unit_iExtension include/uapi/linux/usb/audio.h:483 [inline] build_audio_procunit+0xeab/0x13f0 sound/usb/mixer.c:2434 parse_audio_extension_unit sound/usb/mixer.c:2483 [inline] parse_audio_unit+0x1812/0x36f0 sound/usb/mixer.c:2761 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 12: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_alloc_urb+0x65/0xb0 drivers/usb/core/urb.c:73 usb_internal_control_msg drivers/usb/core/message.c:94 [inline] usb_control_msg+0x1c9/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 12: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d4c01400 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 13 bytes inside of 192-byte region [ffff8881d4c01400, ffff8881d4c014c0) The buggy address belongs to the page: page:ffffea0007530040 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000754fec0 0000000200000002 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d4c01300: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d4c01380: 00 00 00 07 fc fc fc fc fc fc fc fc fc fc fc fc > ffff8881d4c01400: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8881d4c01480: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d4c01500: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] KASAN: use-after-free Read in parse_term_proc_unit
by syzbot 24 Oct '19

24 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=13a38628e00000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=97e24236f1e2918ad968 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12184764e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1432ae28e00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+97e24236f1e2918ad968(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: use-after-free in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: use-after-free in parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 Read of size 1 at addr ffff8881d5346d0e by task kworker/1:2/83 CPU: 1 PID: 83 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 __check_input_term+0xc32/0x13f0 sound/usb/mixer.c:984 parse_audio_feature_unit sound/usb/mixer.c:1875 [inline] parse_audio_unit+0x101d/0x36f0 sound/usb/mixer.c:2753 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_alloc_urb+0x65/0xb0 drivers/usb/core/urb.c:73 usb_internal_control_msg drivers/usb/core/message.c:94 [inline] usb_control_msg+0x1c9/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 usb_get_descriptor+0xc1/0x1b0 drivers/usb/core/message.c:647 usb_get_configuration+0x28e/0x3050 drivers/usb/core/config.c:836 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d5346d00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 14 bytes inside of 192-byte region [ffff8881d5346d00, ffff8881d5346dc0) The buggy address belongs to the page: page:ffffea000754d180 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea0007548ac0 0000000900000009 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d5346c00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d5346c80: 00 00 00 00 05 fc fc fc fc fc fc fc fc fc fc fc > ffff8881d5346d00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8881d5346d80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d5346e00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix 2 front mics of codec 0x623
by Aaron Ma 24 Oct '19

24 Oct '19

These 2 ThinkCentres installed a new realtek codec ID 0x623, it has 2 front mics with the same location on pin 0x18 and 0x19. Apply fixup ALC283_FIXUP_HEADSET_MIC to change 1 front mic location to right, then pulseaudio can handle them. One "Front Mic" and one "Mic" will be shown, and audio output works fine. Signed-off-by: Aaron Ma <aaron.ma(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..c34d8b435f58 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7186,6 +7186,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -9187,6 +9189,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: hdmi-codec: fix locking issue
by Jerome Brunet 24 Oct '19

24 Oct '19

This patchset fixes the locking issue reported by Russell. As explained a mutex was used as flag and held while returning to userspace. Patch 2 is entirely optional and switches from bit atomic operation to mutex again. I tend to prefer bit atomic operation in this particular case but either way should be fine. Jerome Brunet (2): Revert "ASoC: hdmi-codec: re-introduce mutex locking" ASoC: hdmi-codec: re-introduce mutex locking again sound/soc/codecs/hdmi-codec.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH] ASoC: rt5650: Add Kahlee platform specfic changes
by Curtis Malainey 24 Oct '19

24 Oct '19

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Add platform specific data for Kahlee project. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Shirish S <shirish.s(a)amd.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5645.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1c06b3b9218c5..a15e4ecd2a241 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3636,6 +3636,12 @@ static const struct rt5645_platform_data lattepanda_board_platform_data = { .inv_jd1_1 = true }; +static const struct rt5645_platform_data kahlee_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + static const struct dmi_system_id dmi_platform_data[] = { { .ident = "Chrome Buddy", @@ -3742,6 +3748,13 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&lattepanda_board_platform_data, }, + { + .ident = "Chrome Kahlee", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Kahlee"), + }, + .driver_data = (void *)&kahlee_platform_data, + }, { } }; -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH v3 0/5] soundwire: intel/cadence: better initialization
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

Changes since v2: addressed feedback from Vinod Koul on patch 2&4 Add kernel taint when using debugfs hw_reset (similar to regmap) Remove useless goto label Changes since v1: addressed feedback from Vinod Koul clarified init changes impact Intel and Cadence sides remove unnecessary intermediate variable disable interrupts when exit_reset fails, updated error handling returned -EINVAL on debugfs invalid parameter Pierre-Louis Bossart (5): soundwire: intel/cadence: fix startup sequence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel/cadence: add flag for interrupt enable soundwire: cadence_master: make clock stop exit configurable on init drivers/soundwire/cadence_master.c | 134 +++++++++++++++++++++-------- drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 39 ++++++--- 3 files changed, 129 insertions(+), 49 deletions(-) -- 2.20.1

2 6

[alsa-devel] [PATCH 0/2] ASoC: SOF: make hw_info platform specific
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

The SOF core blindly sets the hw_info, when it should really be platform-specific. This move enable us to temporarily fix PulseAudio issues resulting from inaccuracies in the hw_ptr position provided by the firmware on Baytrail-Broadwell. This is the first time we experiment with the GitHub low-level API to automagically keep track of 'approvers' by inserting a Reviewed-by tag when merging a pull-request, thanks to Ranjani for her initial work and bearing with my habit of breaking things :-) Pierre-Louis Bossart (2): ASoC: SOF: define INFO_ flags in dsp_ops ASoC: SOF: Intel: only support INFO_BATCH for legacy platforms sound/soc/sof/imx/imx8.c | 7 +++++++ sound/soc/sof/intel/apl.c | 7 +++++++ sound/soc/sof/intel/bdw.c | 9 ++++++++- sound/soc/sof/intel/byt.c | 21 +++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 7 +++++++ sound/soc/sof/pcm.c | 8 +++----- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 56 insertions(+), 6 deletions(-) -- 2.20.1

2 3

[alsa-devel] [PATCH] soundwire: slave: fix scanf format
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

fix cppcheck warning: [drivers/soundwire/slave.c:145]: (warning) %x in format string (no. 1) requires 'unsigned int *' but the argument type is 'signed int *'. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/slave.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 48a63ca130d2..6473fa602f82 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -128,7 +128,8 @@ int sdw_of_find_slaves(struct sdw_bus *bus) struct device_node *node; for_each_child_of_node(bus->dev->of_node, node) { - int link_id, sdw_version, ret, len; + int link_id, ret, len; + unsigned int sdw_version; const char *compat = NULL; struct sdw_slave_id id; const __be32 *addr; -- 2.20.1

2 1

[alsa-devel] AC97 audio-routing [was: Recording on WM9712]
by Matej Kupljen 24 Oct '19

24 Oct '19

Hi all, I am not sure how to specify audio-routing for AC97 codec. If I use the similar setting as the UDOO board (which was the only board that uses AC97 codec I could find) I get warnings during booting: [ 3.261369] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 3.267917] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 3.274642] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 3.283309] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 3.290141] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture This was tested on kernel Linux version 4.14.134. However, the card is detected: [ 4.041646] ALSA device list: [ 4.044628] #0: ac97-audio and I can use it for playback normally, but the capture is "slow" as described in my previous e-mail. Then I tried to build and boot the latest kernel from GIT (Linux version 5.4.0-rc4) using the same DTS, I get additional warnings from the kernel: [ 2.639878] fsl-asoc-card sound: ac97-hifi <-> 202c000.ssi mapping ok [ 2.651321] fsl-asoc-card sound: ASoC: source widget AC97 Playback overwritten [ 2.658595] fsl-asoc-card sound: ASoC: no sink widget found for Playback [ 2.665317] fsl-asoc-card sound: ASoC: Failed to add route AC97 Playback -> direct -> Playback [ 2.673974] fsl-asoc-card sound: ASoC: sink widget AC97 Capture overwritten [ 2.680986] fsl-asoc-card sound: ASoC: no source widget found for Capture [ 2.687822] fsl-asoc-card sound: ASoC: Failed to add route Capture -> direct -> AC97 Capture [ 2.718636] fsl-asoc-card sound: snd_soc_register_card failed (-19) and the card is not detected: [ 3.285543] ALSA device list: [ 3.288518] No soundcards found. As you can see, there are additional warnings about widget overwrite, because this check was added to the latest kernel, as seen from the kernel source code. This is the setting from my DTS file: sound { compatible = "fsl,imx-audio-ac97"; model = "fsl,imx6q-ac97"; audio-cpu = <&ssi2>; audio-routing = "RX", "Mic Jack", "Headphone Jack", "TX"; mux-int-port = <2>; mux-ext-port = <5>; }; As you can see, I do not even specify any routing about AC97 Playback/Capture and I still get those warnings. For the test, I commented out the code in sound/soc/soc-core.c in function snd_soc_instantiate_card, to ignore the return values from the call to snd_soc_dapm_add_routes and then I can use the card for playback, but the recording has the issues again. What seems to be wrong with this AC97 audio routing? Any suggestions? I am willing to test this on the latest kernel, or from some other GIT repo if needed. Thanks and BR, Matej

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add Gustard U16/X26 device family to quirks for native DSD support
by Fly Pig 24 Oct '19

24 Oct '19

Hi There This patch add native DSD support for Gustard U16/X26 USB Interface. Tested on my unit, working properly. Hence submit the patch. Cheers. Thank you very much. Pig --- a/sound/usb/quirks.c 2019-09-22 21:28:09.757897000 +0000 +++ b/sound/usb/quirks.c 2019-09-22 21:33:36.892274800 +0000 @@ -1369,6 +1369,7 @@ case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ + case USB_ID(0x292b, 0xc4b3): /* Gustard U16 */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break;

2 3

[alsa-devel] [asoc:for-5.5 160/172] sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 24 Oct '19

24 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 175fc928198236037174e5c5c066fe3c4691903e commit: a0e0d135427cf699fe2dee77da0924e0b47f3170 [160/172] ASoC: rt5677: Add a PCM device for streaming hotword via SPI reproduce: # apt-get install sparse # sparse version: v0.6.1-dirty git checkout a0e0d135427cf699fe2dee77da0924e0b47f3170 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/rt5677-spi.c:148:33: sparse: sparse: Using plain integer as NULL pointer >> sound/soc/codecs/rt5677-spi.c:365:13: sparse: sparse: symbol 'rt5677_spi_pcm_page' was not declared. Should it be static? vim +148 sound/soc/codecs/rt5677-spi.c 139 140 static int rt5677_spi_hw_free( 141 struct snd_soc_component *component, 142 struct snd_pcm_substream *substream) 143 { 144 struct rt5677_dsp *rt5677_dsp = 145 snd_soc_component_get_drvdata(component); 146 147 mutex_lock(&rt5677_dsp->dma_lock); > 148 rt5677_dsp->substream = 0; 149 mutex_unlock(&rt5677_dsp->dma_lock); 150 151 return snd_pcm_lib_free_vmalloc_buffer(substream); 152 } 153 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 00/18] soundwire: code hardening and suspend-resume support
by Pierre-Louis Bossart 24 Oct '19

24 Oct '19

this patchset applies on top of "[PATCH 00/14] soundwire: intel: implement new ASoC interfaces". It implements a series of improvements for: a) interrupt handling on Intel platforms in MSI mode b) race conditions on codec probe and enumeration c) suspend-resume issues (clock-stop mode not supported for now) d) underflow handling e) updates to the stream state machine which did not support valid ALSA transitions. These patches were tested extensively on 4 different platforms and are viewed as required for any sort of SoundWire-based product. Bard Liao (3): soundwire: intel/cadence: fix timeouts in MSI mode soundwire: stream: only prepare stream when it is configured. soundwire: intel: reinitialize IP+DSP in .prepare() Pierre-Louis Bossart (15): soundwire: fix race between driver probe and update_status callback soundwire: bus: add PM/no-PM versions of read/write functions soundwire: bus: write Slave Device Number without runtime_pm soundwire: intel: add helpers for link power down and shim wake soundwire: intel: Add basic power management support soundwire: intel: add pm_runtime support soundwire: intel: reset pm_runtime status during system resume soundwire: bus: add helper to reset Slave status to UNATTACHED soundwire: intel: call helper to reset Slave states on resume soundwire: bus: check first if Slaves become UNATTACHED soundwire: add enumeration_complete signaling soundwire: intel: disable pm_runtime when removing a master soundwire: bus: disable pm_runtime in sdw_slave_delete soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Documentation/driver-api/soundwire/stream.rst | 63 +++- drivers/soundwire/bus.c | 157 +++++++-- drivers/soundwire/bus.h | 3 + drivers/soundwire/bus_type.c | 5 + drivers/soundwire/cadence_master.c | 17 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 328 ++++++++++++++++-- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 45 ++- drivers/soundwire/slave.c | 3 + drivers/soundwire/stream.c | 64 +++- include/linux/soundwire/sdw.h | 1 + include/linux/soundwire/sdw_intel.h | 2 + 13 files changed, 623 insertions(+), 71 deletions(-) -- 2.20.1

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Re: [alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by S.j. Wang 23 Oct '19

23 Oct '19

Hi > > On Thu, Oct 17, 2019 at 02:21:08PM +0800, Shengjiu Wang wrote: > > For P2P output, the output divider should align with the output sample > > I think we should avoid "P2P" (or "M2M") keyword in the mainline code as > we know M2M will never get merged while somebody working with the > mainline and caring about new feature might be confused. Ok. But we still curious that is there a way to upstream m2m? > > > rate, if use ideal sample rate, there will be a lot of overload, which > > would cause underrun. > > If I understand it correctly, setting to ideal ratio provides a faster converting > speed but increases the load of the processor of ASRC. So we choose a > slower converting speed here since real- time playback mode doesn't really > need a faster conversion? Yes. Slower speed is enough for real-time playback > > It makes sense to me, yet I feel that the delay at the beginning of the audio > playback might be longer as a compromise. I am okay with this decision > though... > > > The maximum divider of asrc clock is 1024, but there is no judgement > > for this limitaion in driver, which may cause the divider setting not > > correct. > > > > For non-ideal ratio mode, the clock rate should divide the sample rate > > with no remainder, and the quotient should be less than 1024. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 40 > > +++++++++++++++++++++++++++++++--------- > > 1 file changed, 31 insertions(+), 9 deletions(-) > > > > diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index > > 0bf91a6f54b9..44d05ec28bd3 100644 > > --- a/sound/soc/fsl/fsl_asrc.c > > +++ b/sound/soc/fsl/fsl_asrc.c > > @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct > fsl_asrc_pair *pair, > > * of struct asrc_config which includes in/output sample rate, width, > channel > > * and clock settings. > > */ > > -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) > > { > > struct asrc_config *config = pair->config; > > struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 > > @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) > > enum asrc_word_width input_word_width; > > enum asrc_word_width output_word_width; > > u32 inrate, outrate, indiv, outdiv; > > - u32 clk_index[2], div[2]; > > + u32 clk_index[2], div[2], rem[2]; > > + u64 clk_rate; > > int in, out, channels; > > int pre_proc, post_proc; > > struct clk *clk; > > @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair > *pair) > > /* We only have output clock for ideal ratio mode */ > > clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; > > > > - div[IN] = clk_get_rate(clk) / inrate; > > + clk_rate = clk_get_rate(clk); > > The fsl_asrc.c file has config.inclk being set to INCLK_NONE and this sets the > "ideal" in this function to true. So, although we tend to not use ideal ratio > setting for p2p cases, yet the input clock is still not physically connected, so > we still use output clock for div[IN] calculation? For p2p case, it can be ideal or non-ideal. For non-ideal, we still use Output clock for div calculation. > > I am thinking something simplier: if we decided not to use ideal ratio for > "P2P", instead of adding "bool p2p" with the confusing "ideal" in this > function, could we just set config.inclk to the same clock as the output one > for "P2P"? By doing so, "P2P" won't go through ideal ratio mode while still > having a clock rate from the output clock for div[IN] calculation here. Bool p2p is to force output rate to be sample rate, no impact to ideal Ratio mode. > > > + rem[IN] = do_div(clk_rate, inrate); > > + div[IN] = (u32)clk_rate; > > if (div[IN] == 0) { > > Could we check div[IN] and rem[IN] here? Like: > if (div[IN] == 0 || div[IN] > 1024) { > pair_err(); > goto out; > } > > if (!ideal && rem[IN]) { > pair_err(); > goto out; > } > > According to your commit log, I think the max-1024 limitation should be > applied to all cases, not confined to "!ideal" cases right? And we should > add some comments also, indicating it is limited by hardware. For ideal mode, my test result is the divider not impact the output result. Which means it is ok for ideal mode even divider is not correct... > > > pair_err("failed to support input sample rate %dHz by > asrck_%x\n", > > inrate, clk_index[ideal ? OUT : IN]); @@ > > -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct > > fsl_asrc_pair *pair) > > > > clk = asrc_priv->asrck_clk[clk_index[OUT]]; > > > > - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ > > - if (ideal) > > - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; > > - else > > - div[OUT] = clk_get_rate(clk) / outrate; > > + /* > > + * When P2P mode, output rate should align with the out samplerate. > > + * if set too high output rate, there will be lots of Overload. > > + * When M2M mode, output rate should also need to align with the > > + out > > For this "should", do you actually mean "M2M could also"? Sorry, I'm just > trying to understand everyting here, not intentionally being picky at words. > My understanding is that we still keep the ideal ratio setting because > "M2M" still uses it. We use IDEAL_RATIO_RATE as output rate for m2m mode, it likes a Tricky operation, in order to improve the performance. I think The correct operation is to use the real output rate, but the performance Is bad. So it is a compromise. > > > + * samplerate, but M2M must use less time to achieve good > performance. > > + */ > > + clk_rate = clk_get_rate(clk); > > + if (p2p || !ideal) { > > + rem[OUT] = do_div(clk_rate, outrate); > > + div[OUT] = clk_rate; > > + } else { > > + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); > > + div[OUT] = clk_rate; > > + } > > > > if (div[OUT] == 0) { > > pair_err("failed to support output sample rate %dHz by > > asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct > fsl_asrc_pair *pair) > > return -EINVAL; > > } > > > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || > > + rem[IN] != 0 || rem[OUT] != 0)) { > > + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || rem[IN] || > > + rem[OUT] != 0)) { > > So for ideal == true, these limitaions are not applied any more? > Remember that the "ideal" is true for "p2p == true" cases here. No, not applied. for ideal, the div don't impact the output result Even it is not accurate. > > > + pair_err("The divider can't be used for non ideal mode\n"); > > + return -EINVAL; > > + } > > + > > + /* Divider range is [1, 1024] */ > > + div[IN] = min_t(u32, 1024, div[IN]); > > + div[OUT] = min_t(u32, 1024, div[OUT]); > > Hmm, this looks like we want to allow ideal ratio cases and p2p cases to > operate any way, even if the divider wasn't within the range to get the > in/out rates from the output clock? Yes. We still allow the p2p = true, ideal = false. Note that p2p is not Equal to ideal. Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH] ASoC: rsnd: dma: set bus width to data width for monaural data
by Eugeniu Rosca 23 Oct '19

23 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> According to R-Car3 HW manual 40.3.3 (Data Format on Audio Local Bus), in case of monaural data writing or reading through Audio-DMAC, it's always in Left Justified format, so both src and dst DMA Bus width should be equal to physical data width. Therefore set src and dst's DMA bus width to: - [monaural case] data width - [non-monaural case] 32bits (as prior applying the patch) Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 28f65eba2bb4..95aa26d62e4f 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -165,14 +165,40 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; + enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; int is_play = rsnd_io_is_play(io); int ret; + /* + * in case of monaural data writing or reading through Audio-DMAC + * data is always in Left Justified format, so both src and dst + * DMA Bus width need to be set equal to physical data width. + */ + if (rsnd_runtime_channel_original(io) == 1) { + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + int bits = snd_pcm_format_physical_width(runtime->format); + + switch (bits) { + case 8: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case 16: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(dev, "invalid format width %d\n", bits); + return -EINVAL; + } + } + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; - cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.src_addr_width = buswidth; + cfg.dst_addr_width = buswidth; dev_dbg(dev, "%s %pad -> %pad\n", rsnd_mod_name(mod), -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: add missing of_node_put()
by Kuninori Morimoto 23 Oct '19

23 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch adds missing of_node_put() for rsnd_parse_tdm_split_mode() rsnd_parse_connect_graph() Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index fe681c5..14e86e4 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1075,7 +1075,10 @@ static void rsnd_parse_tdm_split_mode(struct rsnd_priv *priv, j++; } + of_node_put(node); } + + of_node_put(ssiu_np); } static void rsnd_parse_connect_simple(struct rsnd_priv *priv, @@ -1093,11 +1096,13 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, struct device_node *endpoint) { struct device *dev = rsnd_priv_to_dev(priv); - struct device_node *remote_node = of_graph_get_remote_port_parent(endpoint); + struct device_node *remote_node; if (!rsnd_io_to_mod_ssi(io)) return; + remote_node = of_graph_get_remote_port_parent(endpoint); + /* HDMI0 */ if (strstr(remote_node->full_name, "hdmi@fead0000")) { rsnd_flags_set(io, RSND_STREAM_HDMI0); @@ -1111,6 +1116,8 @@ static void rsnd_parse_connect_graph(struct rsnd_priv *priv, } rsnd_parse_tdm_split_mode(priv, io, endpoint); + + of_node_put(remote_node); } void rsnd_parse_connect_common(struct rsnd_dai *rdai, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: mediatek: Check SND_SOC_CROS_EC_CODEC dependency
by Mao Wenan 23 Oct '19

23 Oct '19

If SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=y, below errors can be seen: sound/soc/codecs/cros_ec_codec.o: In function `send_ec_host_command': cros_ec_codec.c:(.text+0x534): undefined reference to `cros_ec_cmd_xfer_status' cros_ec_codec.c:(.text+0x101c): undefined reference to `cros_ec_get_host_event' This is because it will select SND_SOC_CROS_EC_CODEC after commit 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV"), but SND_SOC_CROS_EC_CODEC depends on CROS_EC. Fixes: 2cc3cd5fdc8b ("ASoC: mediatek: mt8183: support WoV") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 8b29f39..a656d20 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -125,7 +125,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A select SND_SOC_MAX98357A select SND_SOC_BT_SCO select SND_SOC_TS3A227E - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help This adds ASoC driver for Mediatek MT8183 boards with the MT6358 TS3A227E MAX98357A audio codec. -- 2.7.4

4 6

[alsa-devel] [PATCH] ASoC: kirkwood: fix external clock probe defer
by Russell King 23 Oct '19

23 Oct '19

When our call to get the external clock fails, we forget to clean up the enabled internal clock correctly. Enable the clock after we have obtained all our resources. Fixes: 84aac6c79bfd ("ASoC: kirkwood: fix loss of external clock at probe time") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3446a113f482..c323ae314b55 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,10 +559,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->clk); } - err = clk_prepare_enable(priv->clk); - if (err < 0) - return err; - priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (IS_ERR(priv->extclk)) { if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) @@ -578,6 +574,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } } + err = clk_prepare_enable(priv->clk); + if (err < 0) + return err; + /* Some sensible defaults - this reflects the powerup values */ priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24; priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; -- 2.20.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: SOF: Intel: improve loader error handling
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda-loader: improve error handling ASoC: SOF: Intel: hda: add dev_err() traces for snd_sof_dsp_read_poll_timeout() sound/soc/sof/intel/hda-dsp.c | 30 +++++++++++++++++++++++++++--- sound/soc/sof/intel/hda-loader.c | 25 ++++++++++++++++++++++--- sound/soc/sof/intel/hda-stream.c | 24 ++++++++++++++++++++---- 3 files changed, 69 insertions(+), 10 deletions(-) -- 2.20.1

2 4

[alsa-devel] [PATCH 0/2] ASoC: SOF: initial support for JasperLake
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

Add PCI IDs and dummy tables for ACPI matching (to be populated when machine drivers are provided, used for nocodec mode for now). Pan Xiuli (2): ASoC: Intel: common: add ACPI matching tables for JSL ASoC: SOF: Intel: initial support to JasperLake. include/sound/soc-acpi-intel-match.h | 1 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-jsl-match.c | 18 +++++++++++++++ sound/soc/sof/intel/Kconfig | 16 ++++++++++++++ sound/soc/sof/intel/cnl.c | 17 ++++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/sof-pci-dev.c | 22 +++++++++++++++++++ 7 files changed, 76 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-jsl-match.c -- 2.20.1

2 4

[alsa-devel] [PATCH] ASoC: kirkwood: fix IRQ error handling
by Russell King 23 Oct '19

23 Oct '19

Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 6f69f314f2c2..d2d5c25bf550 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.20.1

2 1

[alsa-devel] [RESEND PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by Eugeniu Rosca 23 Oct '19

23 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit's busif dma address is calculated by following calculation formula: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to R-Car3 HW manual 41.1.4 Register Configuration, ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formula to correct ssi9 4/5/6/7 busif data register address. Fixes: 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> [erosca: minor improvements in commit description] Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: stable(a)vger.kernel.org # v4.20+ Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- Originally submitted as https://patchwork.kernel.org/patch/10825513/ ("ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address") --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c39619..28f65eba2bb4 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.23.0

3 3

[alsa-devel] [PATCH] ASoC: kirkwood: fix device remove ordering
by Russell King 23 Oct '19

23 Oct '19

The devm conversion of kirkwood was incorrect; on removal, devm takes effect after the "remove" function has returned. So, the effect of the conversion was to change the order during remove from: - snd_soc_unregister_component() (unpublishes interfaces) - clk_disable_unprepare() - cleanup resources After the conversion, this became: - clk_disable_unprepare() - while the device may still be active - snd_soc_unregister_component() - cleanup resources Hence, it introduces a bug, where the internal clock for the device may be shut down before the device itself has been shut down. It is known that Marvell SoCs, including Dove, locks up if registers for a peripheral that has its clocks disabled are accessed. Fixes: f98fc0f8154e ("ASoC: kirkwood: replace platform to component") Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index c323ae314b55..eb38cdb37f0e 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -591,7 +591,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_128; } - err = devm_snd_soc_register_component(&pdev->dev, &kirkwood_soc_component, + err = snd_soc_register_component(&pdev->dev, &kirkwood_soc_component, soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); @@ -614,6 +614,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) { struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); if (!IS_ERR(priv->extclk)) clk_disable_unprepare(priv->extclk); clk_disable_unprepare(priv->clk); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_esai: Add spin lock to protect reset and stop
by Shengjiu Wang 23 Oct '19

23 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. So Add spin lock to lock these two function. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..6a797648b66d 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -734,7 +741,9 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1011,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

Re: [alsa-devel] Question about "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params"
by S.j. Wang 23 Oct '19

23 Oct '19

Hi Mark > > On Wed, Oct 16, 2019 at 03:05:16AM +0000, S.j. Wang wrote: > > > The soc_pcm_params_symmetry can't take effect, because its > > position is after the change of cpu_dai->rate. > > > Is this an issue? > > I think so, yes. Thanks, but how to fix it? because I don't really understand this commit want to do, it is hard for me to send a fixed patch. Or shall we revert it? Best regards Wang shengjiu

2 1

[alsa-devel] alsa-info: lots of mostly syntactic changes
by GitHub pull_request - opened 23 Oct '19

23 Oct '19

alsa-project/alsa-utils pull request #29 was opened from E5ten: - remove all bashisms and change shebang to #!/bin/sh - replace use of which with command -v for portability - remove awk dependency - general cosmetic consistency: - indentation - always put then or done on the same line as if, for, or while - quote much more, especially when the variable refers to a file - replace many uses of >> $FILE with command grouping and single redirects of the command groups - simplify text processing pipelines, mostly by turning them into a single sed command Request URL : https://github.com/alsa-project/alsa-utils/pull/29 Patch URL : https://github.com/alsa-project/alsa-utils/pull/29.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [RFC] sof-skl_hda: change to use common HDMI codec driver
by GitHub pull_request - opened 23 Oct '19

23 Oct '19

alsa-project/alsa-ucm-conf pull request #4 was opened from kv2019i: Here's a companion patch to the hda-hdmi kernel patchset on alsa-devel: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/157201.h… Modify UCM sequences to use user-space interface of the common HDMI codec driver. Use of hdac-hdmi driver is no longer supported by this UCM file. To use this UCM file, kernel must be compiled with SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/4 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

2 1

[alsa-devel] alsaloop change avail_min question
by ed nwave 23 Oct '19

23 Oct '19

Hi, Over time I see the message below and the value slowly creeps up. Is this a simple information output or more of a warning? If run alsaloop for a long time - many hours of music, will alsaloop be prone to failure because of this value increasing? playback plughw:Siso: change avail_min=7299 Thanks.

2 2

[alsa-devel] [PATCH] ASoC: Intel: Reduce Resource notification trace to once per minute.
by vani.ganji＠intel.com 23 Oct '19

23 Oct '19

From: Vani Ganji <vani.ganji(a)intel.com> Resource notification trace "MCPS Budget violation" observed very frequently which might cause losing critical debug information. Added a counter to print trace on first instance and once per minute. It is required to be notified when module resource is used more than a threshold so enabled once per minute. Suggested-by: Grant Grundler <grundler(a)google.com> Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/soc/intel/skylake/skl-sst-ipc.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 667cdddc289f..03ff0d8cea3e 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -346,6 +346,7 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, struct skl_ipc_header header) { struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc); + static unsigned long j; if (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { switch (IPC_GLB_NOTIFY_TYPE(header.primary)) { @@ -355,12 +356,17 @@ int skl_ipc_process_notification(struct sst_generic_ipc *ipc, break; case IPC_GLB_NOTIFY_RESOURCE_EVENT: - dev_err(ipc->dev, "MCPS Budget Violation: %x\n", + /*Print trace on first instance and for every minute*/ + if (printk_timed_ratelimit(&j, 60000)) { + dev_err(ipc->dev, "MCPS Budget Violation: %x\n", header.primary); + } break; case IPC_GLB_NOTIFY_FW_READY: skl->boot_complete = true; + /* reset timer on every FW load */ + j = 0; wake_up(&skl->boot_wait); break; -- 2.17.1

1 0

[alsa-devel] [PATCH 00/36] ARM: samsung platform cleanup
by Arnd Bergmann 23 Oct '19

23 Oct '19

I've spent some time looking at the remaining ARMv4/ARMv5 platforms that are not part of ARCH_MULTIPLATFORM, and tried to get them closer to that. Here is what came out of that for the samsung platforms: * Exynos and s5pv210 are made independent of plat-samsung * device drivers stop using mach/*.h headers for s3c24xx (and other platforms not in this series) * s3c24xx and s3c64xx get merged into mach-s3c, removing the need for plat-samsung (I have other patches for the remaining plat-* directories) * mach/io.h gets cleaned up to only be needed for BAST PC104 mode (looking for ideas to proceed) * mach/irqs.h remains for now, this still needs to be converted to sparse IRQs. Some bits are a little ugly, but overall I think this a big improvement. The contents are available for testing in git://kernel.org:/pub/scm/linux/kernel/git/arnd/playground.git s3c-multiplatform Given the overall size of the series I expect that there will be bugs, so at this points I'm looking for reviews, acks and testing from anyone who still has the hardware, in particular s3c24xx, but also the other samsung platforms I'm touching. Arnd Arnd Bergmann (36): ARM: samsung: make S3C24XX_MISCCR access indirect ARM: s3c: simplify mach/io.h usb: gadget: s3c: use platform resources usb: gadget: s3c-hsudc: remove platform header dependency ARM: samsung: make pm-debug platform independent ARM: samsung: move CONFIG_DEBUG_S3C_UART to Kconfig.debug ARM: exynos: use private samsung_cpu_id copy ARM: exynos: stop selecting PLAT_SAMSUNG ARM: samsung: move pm check code to drivers/soc ARM: s5pv210: use private pm save/restore ARM: s5pv210: split from plat-samsung ARM: s3c64xx: remove mach/hardware.h ARM: s3c: move regs-spi.h into spi driver ARM: s3c: move irqchip driver back into platform ARM: s3c: adc: move header to linux/soc/samsung ARM: s3c: move spi fiq handler into platform ARM: s3c: h1940-audio: turn into platform driver ARM: s3c: gta02-audio: turn into platform driver ARM: s3c: rx1950: turn into platform driver ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS ARM: s3c: move iis pinctrl config into boards ARM: s3c: leds: move setpull() calls into board files ARM: s3c: move s3cmci pinctrl handling into board files ARM: s3c: include mach/irqs.h where needed ARM: s3c: spi: avoid hardcoding fiq number in driver ARM: s3c: bast: avoid irq_desc array usage ARM: s3c: fix mmc gpio lookup tables fbdev: s3c2410fb: remove mach header dependency ARM: s3c: cpufreq: split out registers ARM: s3c: remove cpufreq header dependencies ARM: s3c: cpufreq: use global s3c2412_cpufreq_setrefresh ARM: s3c: iotiming: make functions static ARM: s3c: move low-level clk reg access into platform code ARM: s3c: stop including mach/hardware.h from mach/io.h ARM: s3c: move into a common directory ARM: s3c: make headers local if possible MAINTAINERS | 16 +- arch/arm/Kconfig | 5 +- arch/arm/Kconfig.debug | 18 +- arch/arm/Makefile | 7 +- arch/arm/mach-exynos/Makefile | 4 - arch/arm/mach-exynos/common.h | 6 +- arch/arm/mach-exynos/exynos.c | 19 +- arch/arm/mach-exynos/include/mach/map.h | 18 -- arch/arm/mach-exynos/platsmp.c | 4 +- arch/arm/mach-exynos/pm.c | 8 +- arch/arm/{plat-samsung => mach-s3c}/Kconfig | 62 +---- .../Kconfig => mach-s3c/Kconfig.s3c24xx} | 0 .../Kconfig => mach-s3c/Kconfig.s3c64xx} | 0 arch/arm/{plat-samsung => mach-s3c}/Makefile | 15 +- .../{mach-s3c24xx => mach-s3c}/Makefile.boot | 0 .../Makefile => mach-s3c/Makefile.s3c24xx} | 15 +- .../Makefile => mach-s3c/Makefile.s3c64xx} | 14 +- .../include/plat => mach-s3c}/adc-core.h | 0 arch/arm/{plat-samsung => mach-s3c}/adc.c | 4 +- arch/arm/{mach-s3c24xx => mach-s3c}/anubis.h | 0 .../arm/{mach-s3c64xx => mach-s3c}/ata-core.h | 0 .../{mach-s3c64xx => mach-s3c}/backlight.h | 0 .../arm/{mach-s3c24xx => mach-s3c}/bast-ide.c | 3 +- .../arm/{mach-s3c24xx => mach-s3c}/bast-irq.c | 8 +- arch/arm/{mach-s3c24xx => mach-s3c}/bast.h | 0 .../{mach-s3c24xx => mach-s3c}/common-smdk.c | 17 +- .../{mach-s3c24xx => mach-s3c}/common-smdk.h | 0 arch/arm/{plat-samsung => mach-s3c}/cpu.c | 21 +- .../include/plat => mach-s3c}/cpu.h | 11 - .../cpufreq-utils.c | 38 +++- arch/arm/{mach-s3c64xx => mach-s3c}/cpuidle.c | 4 +- .../arm/{mach-s3c64xx => mach-s3c}/crag6410.h | 2 +- .../{mach-s3c64xx => mach-s3c}/dev-audio.c | 9 +- .../dev-backlight.c | 4 +- .../dev-uart-s3c64xx.c} | 5 +- .../arm/{plat-samsung => mach-s3c}/dev-uart.c | 2 +- arch/arm/{plat-samsung => mach-s3c}/devs.c | 56 +++-- .../include/plat => mach-s3c}/devs.h | 0 .../mach/dma.h => mach-s3c/dma-s3c24xx.h} | 0 .../mach/dma.h => mach-s3c/dma-s3c64xx.h} | 0 arch/arm/mach-s3c/dma.h | 7 + arch/arm/{mach-s3c24xx => mach-s3c}/fb-core.h | 0 .../include/plat => mach-s3c}/fb.h | 0 .../plat => mach-s3c}/gpio-cfg-helpers.h | 0 .../include/plat => mach-s3c}/gpio-cfg.h | 0 .../include/plat => mach-s3c}/gpio-core.h | 2 +- .../gpio-samsung-s3c24xx.h} | 2 + .../gpio-samsung-s3c64xx.h} | 0 .../{plat-samsung => mach-s3c}/gpio-samsung.c | 18 +- arch/arm/mach-s3c/gpio-samsung.h | 7 + arch/arm/{mach-s3c24xx => mach-s3c}/gta02.h | 2 +- .../h1940-bluetooth.c | 7 +- arch/arm/{mach-s3c24xx => mach-s3c}/h1940.h | 0 .../include/mach => mach-s3c}/hardware.h | 7 - .../include/plat => mach-s3c}/iic-core.h | 0 arch/arm/mach-s3c/include/mach/io.h | 50 +++++ .../include/mach/irqs-s3c24xx.h} | 0 .../include/mach/irqs-s3c64xx.h} | 0 arch/arm/mach-s3c/include/mach/irqs.h | 7 + .../plat => mach-s3c/include/mach}/map-base.h | 0 arch/arm/{plat-samsung => mach-s3c}/init.c | 4 +- .../iotiming-s3c2410.c | 12 +- .../iotiming-s3c2412.c | 6 +- .../irq-pm.c => mach-s3c/irq-pm-s3c24xx.c} | 16 +- .../irq-pm.c => mach-s3c/irq-pm-s3c64xx.c} | 8 +- arch/arm/mach-s3c/irq-s3c24xx-fiq-exports.c | 9 + .../arm/mach-s3c/irq-s3c24xx-fiq.S | 10 +- .../arm/mach-s3c}/irq-s3c24xx.c | 23 +- .../arm/{mach-s3c64xx => mach-s3c}/irq-uart.h | 0 .../include/plat => mach-s3c}/keypad.h | 0 .../mach-amlm5900.c | 37 ++- .../{mach-s3c24xx => mach-s3c}/mach-anubis.c | 14 +- .../{mach-s3c64xx => mach-s3c}/mach-anw6410.c | 17 +- .../mach-at2440evb.c | 28 ++- .../{mach-s3c24xx => mach-s3c}/mach-bast.c | 22 +- .../mach-crag6410-module.c | 2 +- .../mach-crag6410.c | 27 ++- .../{mach-s3c24xx => mach-s3c}/mach-gta02.c | 57 ++++- .../{mach-s3c24xx => mach-s3c}/mach-h1940.c | 65 ++++-- .../arm/{mach-s3c64xx => mach-s3c}/mach-hmt.c | 15 +- .../{mach-s3c24xx => mach-s3c}/mach-jive.c | 27 ++- .../mach-mini2440.c | 48 ++-- .../mach-mini6410.c | 22 +- .../arm/{mach-s3c24xx => mach-s3c}/mach-n30.c | 40 +++- .../arm/{mach-s3c64xx => mach-s3c}/mach-ncp.c | 13 +- .../mach-nexcoder.c | 20 +- .../mach-osiris-dvs.c | 4 +- .../{mach-s3c24xx => mach-s3c}/mach-osiris.c | 22 +- .../{mach-s3c24xx => mach-s3c}/mach-otom.c | 17 +- .../{mach-s3c24xx => mach-s3c}/mach-qt2410.c | 37 ++- .../mach-real6410.c | 18 +- .../{mach-s3c24xx => mach-s3c}/mach-rx1950.c | 57 +++-- .../{mach-s3c24xx => mach-s3c}/mach-rx3715.c | 26 ++- .../mach-s3c2416-dt.c | 8 +- .../mach-s3c64xx-dt.c | 6 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq.c | 18 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq.h | 0 .../{mach-s3c64xx => mach-s3c}/mach-smartq5.c | 18 +- .../{mach-s3c64xx => mach-s3c}/mach-smartq7.c | 18 +- .../mach-smdk2410.c | 14 +- .../mach-smdk2413.c | 21 +- .../mach-smdk2416.c | 23 +- .../mach-smdk2440.c | 25 ++- .../mach-smdk2443.c | 14 +- .../mach-smdk6400.c | 13 +- .../mach-smdk6410.c | 23 +- .../mach-tct_hammer.c | 24 +- .../{mach-s3c24xx => mach-s3c}/mach-vr1000.c | 20 +- .../{mach-s3c24xx => mach-s3c}/mach-vstms.c | 20 +- .../include/plat => mach-s3c}/map-s3c.h | 12 +- .../mach/map.h => mach-s3c/map-s3c24xx.h} | 6 +- .../mach/map.h => mach-s3c/map-s3c64xx.h} | 4 +- .../include/plat => mach-s3c}/map-s5p.h | 4 +- arch/arm/mach-s3c/map.h | 7 + .../{mach-s3c24xx => mach-s3c}/nand-core.h | 0 .../{mach-s3c64xx => mach-s3c}/onenand-core.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/osiris.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/otom.h | 0 arch/arm/{mach-s3c64xx => mach-s3c}/pl080.c | 4 +- .../{plat-samsung => mach-s3c}/platformdata.c | 4 +- .../{mach-s3c24xx => mach-s3c}/pll-s3c2410.c | 4 +- .../pll-s3c2440-12000000.c | 4 +- .../pll-s3c2440-16934400.c | 4 +- .../{plat-samsung => mach-s3c}/pm-common.c | 2 +- arch/arm/mach-s3c/pm-common.h | 40 ++++ .../pm-core.h => mach-s3c/pm-core-s3c24xx.h} | 8 +- .../pm-core.h => mach-s3c/pm-core-s3c64xx.h} | 50 +---- arch/arm/mach-s3c/pm-core.h | 7 + arch/arm/{plat-samsung => mach-s3c}/pm-gpio.c | 6 +- .../arm/{mach-s3c24xx => mach-s3c}/pm-h1940.S | 5 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2410.c | 11 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2412.c | 11 +- .../{mach-s3c24xx => mach-s3c}/pm-s3c2416.c | 6 +- .../pm.c => mach-s3c/pm-s3c24xx.c} | 12 +- .../pm.c => mach-s3c/pm-s3c64xx.c} | 66 +++++- arch/arm/{plat-samsung => mach-s3c}/pm.c | 20 +- .../include/plat => mach-s3c}/pm.h | 2 +- .../include/plat => mach-s3c}/pwm-core.h | 0 .../include/plat => mach-s3c}/regs-adc.h | 0 .../regs-clock-s3c24xx.h} | 2 + .../regs-clock-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-clock.h | 7 + .../arm/{mach-s3c24xx => mach-s3c}/regs-dsc.h | 0 .../regs-gpio-memport.h | 0 .../regs-gpio-s3c24xx.h} | 2 + .../regs-gpio-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-gpio.h | 7 + .../regs-irq-s3c24xx.h} | 2 + .../regs-irq-s3c64xx.h} | 0 arch/arm/mach-s3c/regs-irq.h | 7 + .../include/plat => mach-s3c}/regs-irqtype.h | 0 .../arm/{mach-s3c24xx => mach-s3c}/regs-mem.h | 2 + .../{mach-s3c64xx => mach-s3c}/regs-modem.h | 0 .../mach => mach-s3c}/regs-s3c2443-clock.h | 50 +++++ .../{mach-s3c64xx => mach-s3c}/regs-srom.h | 0 .../arm/{mach-s3c64xx => mach-s3c}/regs-sys.h | 0 .../regs-syscon-power.h | 0 .../regs-usb-hsotg-phy.h | 0 .../include/mach => mach-s3c}/rtc-core.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/s3c2410.c | 21 +- .../s3c2412-power.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412.c | 16 +- .../include/mach => mach-s3c}/s3c2412.h | 2 + arch/arm/{mach-s3c24xx => mach-s3c}/s3c2416.c | 28 +-- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2440.c | 17 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2442.c | 17 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c2443.c | 21 +- arch/arm/{mach-s3c24xx => mach-s3c}/s3c244x.c | 16 +- .../common.c => mach-s3c/s3c24xx.c} | 21 +- .../common.h => mach-s3c/s3c24xx.h} | 1 + arch/arm/{mach-s3c64xx => mach-s3c}/s3c6400.c | 14 +- arch/arm/{mach-s3c64xx => mach-s3c}/s3c6410.c | 17 +- .../common.c => mach-s3c/s3c64xx.c} | 24 +- .../common.h => mach-s3c/s3c64xx.h} | 0 .../include/plat => mach-s3c}/samsung-time.h | 0 .../include/plat => mach-s3c}/sdhci.h | 2 +- .../{mach-s3c24xx => mach-s3c}/setup-camif.c | 4 +- .../setup-fb-24bpp.c | 6 +- .../{mach-s3c24xx => mach-s3c}/setup-i2c.c | 7 +- .../{mach-s3c64xx => mach-s3c}/setup-i2c0.c | 4 +- .../{mach-s3c64xx => mach-s3c}/setup-i2c1.c | 4 +- .../{mach-s3c64xx => mach-s3c}/setup-ide.c | 8 +- .../{mach-s3c64xx => mach-s3c}/setup-keypad.c | 6 +- .../setup-sdhci-gpio-s3c24xx.c} | 6 +- .../setup-sdhci-gpio-s3c64xx.c} | 6 +- .../setup-spi-s3c24xx.c} | 6 +- .../setup-spi-s3c64xx.c} | 4 +- .../arm/{mach-s3c24xx => mach-s3c}/setup-ts.c | 5 +- .../setup-usb-phy.c | 6 +- .../{mach-s3c24xx => mach-s3c}/simtec-audio.c | 11 +- .../{mach-s3c24xx => mach-s3c}/simtec-nor.c | 2 +- .../{mach-s3c24xx => mach-s3c}/simtec-pm.c | 8 +- .../{mach-s3c24xx => mach-s3c}/simtec-usb.c | 6 +- arch/arm/{mach-s3c24xx => mach-s3c}/simtec.h | 0 .../sleep-s3c2410.S | 7 +- .../sleep-s3c2412.S | 5 +- .../sleep.S => mach-s3c/sleep-s3c24xx.S} | 7 +- .../sleep.S => mach-s3c/sleep-s3c64xx.S} | 4 +- .../arm/{mach-s3c24xx => mach-s3c}/spi-core.h | 0 .../include/plat => mach-s3c}/usb-phy.h | 0 arch/arm/{mach-s3c24xx => mach-s3c}/vr1000.h | 0 .../{plat-samsung => mach-s3c}/wakeup-mask.c | 4 +- .../include/plat => mach-s3c}/wakeup-mask.h | 0 .../watchdog-reset.c | 0 .../watchdog-reset.h | 0 arch/arm/mach-s3c24xx/include/mach/fb.h | 2 - arch/arm/mach-s3c24xx/include/mach/io.h | 212 ------------------ arch/arm/mach-s3c64xx/include/mach/hardware.h | 17 -- arch/arm/mach-s5pv210/Kconfig | 1 + arch/arm/mach-s5pv210/Makefile | 7 - arch/arm/mach-s5pv210/pm.c | 51 ++++- arch/arm/mach-s5pv210/regs-clock.h | 2 +- arch/arm/mach-s5pv210/s5pv210.c | 2 - drivers/clk/samsung/clk-s3c2410-dclk.c | 10 +- drivers/cpufreq/s3c2410-cpufreq.c | 10 +- drivers/cpufreq/s3c2412-cpufreq.c | 44 ++-- drivers/cpufreq/s3c2440-cpufreq.c | 29 ++- drivers/cpufreq/s3c24xx-cpufreq-debugfs.c | 2 +- drivers/cpufreq/s3c24xx-cpufreq.c | 14 +- drivers/hwmon/s3c-hwmon.c | 2 +- drivers/input/touchscreen/s3c2410_ts.c | 37 ++- drivers/irqchip/Makefile | 1 - drivers/leds/leds-s3c24xx.c | 7 - drivers/mmc/host/Kconfig | 2 +- drivers/mmc/host/s3cmci.c | 72 ++---- drivers/power/supply/s3c_adc_battery.c | 2 +- drivers/pwm/Kconfig | 2 +- drivers/soc/samsung/Kconfig | 48 +++- drivers/soc/samsung/Makefile | 3 + .../soc/samsung/s3c-pm-check.c | 2 +- .../soc/samsung/s3c-pm-debug.c | 26 +-- drivers/spi/Kconfig | 2 +- drivers/spi/Makefile | 1 - .../spi/spi-s3c24xx-regs.h | 3 +- drivers/spi/spi-s3c24xx.c | 28 +-- drivers/tty/serial/Kconfig | 2 +- drivers/usb/gadget/udc/s3c-hsudc.c | 55 +---- drivers/usb/gadget/udc/s3c2410_udc.c | 31 +-- drivers/usb/gadget/udc/s3c2410_udc.h | 1 + .../usb/gadget/udc/s3c2410_udc_regs.h | 2 + .../video/fbdev/s3c2410fb-regs-lcd.h | 28 +-- drivers/video/fbdev/s3c2410fb.c | 16 +- .../linux/platform_data}/fb-s3c2410.h | 33 ++- include/linux/platform_data/mmc-s3cmci.h | 2 + include/linux/platform_data/s3c-hsudc.h | 2 + .../linux/soc/samsung/s3c-adc.h | 0 .../linux/soc/samsung/s3c-cpu-freq.h | 0 .../linux/soc/samsung/s3c-cpufreq-core.h | 10 +- .../linux/soc/samsung/s3c-pm.h | 69 +++--- .../linux/spi/s3c24xx-fiq.h | 5 + include/linux/spi/s3c24xx.h | 2 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/h1940_uda1380.c | 71 ++---- sound/soc/samsung/neo1973_wm8753.c | 85 +++---- sound/soc/samsung/rx1950_uda1380.c | 72 ++---- sound/soc/samsung/s3c-i2s-v2.c | 3 +- sound/soc/samsung/s3c-i2s-v2.h | 3 +- sound/soc/samsung/s3c2412-i2s.c | 9 +- sound/soc/samsung/s3c24xx-i2s.c | 7 - 259 files changed, 1716 insertions(+), 1648 deletions(-) delete mode 100644 arch/arm/mach-exynos/include/mach/map.h rename arch/arm/{plat-samsung => mach-s3c}/Kconfig (73%) rename arch/arm/{mach-s3c24xx/Kconfig => mach-s3c/Kconfig.s3c24xx} (100%) rename arch/arm/{mach-s3c64xx/Kconfig => mach-s3c/Kconfig.s3c64xx} (100%) rename arch/arm/{plat-samsung => mach-s3c}/Makefile (67%) rename arch/arm/{mach-s3c24xx => mach-s3c}/Makefile.boot (100%) rename arch/arm/{mach-s3c24xx/Makefile => mach-s3c/Makefile.s3c24xx} (86%) rename arch/arm/{mach-s3c64xx/Makefile => mach-s3c/Makefile.s3c64xx} (83%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/adc-core.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/adc.c (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/anubis.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/ata-core.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/backlight.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast-ide.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast-irq.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/bast.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/common-smdk.c (93%) rename arch/arm/{mach-s3c24xx => mach-s3c}/common-smdk.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/cpu.c (62%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/cpu.h (91%) rename arch/arm/{mach-s3c24xx => mach-s3c}/cpufreq-utils.c (70%) rename arch/arm/{mach-s3c64xx => mach-s3c}/cpuidle.c (96%) rename arch/arm/{mach-s3c64xx => mach-s3c}/crag6410.h (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/dev-audio.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/dev-backlight.c (98%) rename arch/arm/{mach-s3c64xx/dev-uart.c => mach-s3c/dev-uart-s3c64xx.c} (95%) rename arch/arm/{plat-samsung => mach-s3c}/dev-uart.c (97%) rename arch/arm/{plat-samsung => mach-s3c}/devs.c (96%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/devs.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/dma.h => mach-s3c/dma-s3c24xx.h} (100%) rename arch/arm/{mach-s3c64xx/include/mach/dma.h => mach-s3c/dma-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/dma.h rename arch/arm/{mach-s3c24xx => mach-s3c}/fb-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/fb.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-cfg-helpers.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-cfg.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/gpio-core.h (99%) rename arch/arm/{mach-s3c24xx/include/mach/gpio-samsung.h => mach-s3c/gpio-samsung-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/gpio-samsung.h => mach-s3c/gpio-samsung-s3c64xx.h} (100%) rename arch/arm/{plat-samsung => mach-s3c}/gpio-samsung.c (99%) create mode 100644 arch/arm/mach-s3c/gpio-samsung.h rename arch/arm/{mach-s3c24xx => mach-s3c}/gta02.h (94%) rename arch/arm/{mach-s3c24xx => mach-s3c}/h1940-bluetooth.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/h1940.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/hardware.h (76%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/iic-core.h (100%) create mode 100644 arch/arm/mach-s3c/include/mach/io.h rename arch/arm/{mach-s3c24xx/include/mach/irqs.h => mach-s3c/include/mach/irqs-s3c24xx.h} (100%) rename arch/arm/{mach-s3c64xx/include/mach/irqs.h => mach-s3c/include/mach/irqs-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/include/mach/irqs.h rename arch/arm/{plat-samsung/include/plat => mach-s3c/include/mach}/map-base.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/init.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/iotiming-s3c2410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/iotiming-s3c2412.c (98%) rename arch/arm/{mach-s3c24xx/irq-pm.c => mach-s3c/irq-pm-s3c24xx.c} (93%) rename arch/arm/{mach-s3c64xx/irq-pm.c => mach-s3c/irq-pm-s3c64xx.c} (97%) create mode 100644 arch/arm/mach-s3c/irq-s3c24xx-fiq-exports.c rename drivers/spi/spi-s3c24xx-fiq.S => arch/arm/mach-s3c/irq-s3c24xx-fiq.S (94%) rename {drivers/irqchip => arch/arm/mach-s3c}/irq-s3c24xx.c (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/irq-uart.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/keypad.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-amlm5900.c (84%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-anubis.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-anw6410.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-at2440evb.c (88%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-bast.c (98%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-crag6410-module.c (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-crag6410.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-gta02.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-h1940.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-hmt.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-jive.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-mini2440.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-mini6410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-n30.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-ncp.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-nexcoder.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-osiris-dvs.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-osiris.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-otom.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-qt2410.c (88%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-real6410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-rx1950.c (93%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-rx3715.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-s3c2416-dt.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-s3c64xx-dt.c (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq.c (97%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq5.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smartq7.c (94%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2410.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2413.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2416.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2440.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-smdk2443.c (93%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smdk6400.c (92%) rename arch/arm/{mach-s3c64xx => mach-s3c}/mach-smdk6410.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-tct_hammer.c (82%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-vr1000.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/mach-vstms.c (90%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/map-s3c.h (87%) rename arch/arm/{mach-s3c24xx/include/mach/map.h => mach-s3c/map-s3c24xx.h} (97%) rename arch/arm/{mach-s3c64xx/include/mach/map.h => mach-s3c/map-s3c64xx.h} (98%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/map-s5p.h (85%) create mode 100644 arch/arm/mach-s3c/map.h rename arch/arm/{mach-s3c24xx => mach-s3c}/nand-core.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/onenand-core.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/osiris.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/otom.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/pl080.c (99%) rename arch/arm/{plat-samsung => mach-s3c}/platformdata.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2410.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2440-12000000.c (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pll-s3c2440-16934400.c (98%) rename arch/arm/{plat-samsung => mach-s3c}/pm-common.c (98%) create mode 100644 arch/arm/mach-s3c/pm-common.h rename arch/arm/{mach-s3c24xx/include/mach/pm-core.h => mach-s3c/pm-core-s3c24xx.h} (95%) rename arch/arm/{mach-s3c64xx/include/mach/pm-core.h => mach-s3c/pm-core-s3c64xx.h} (58%) create mode 100644 arch/arm/mach-s3c/pm-core.h rename arch/arm/{plat-samsung => mach-s3c}/pm-gpio.c (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-h1940.S (80%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2410.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2412.c (95%) rename arch/arm/{mach-s3c24xx => mach-s3c}/pm-s3c2416.c (95%) rename arch/arm/{mach-s3c24xx/pm.c => mach-s3c/pm-s3c24xx.c} (94%) rename arch/arm/{mach-s3c64xx/pm.c => mach-s3c/pm-s3c64xx.c} (83%) rename arch/arm/{plat-samsung => mach-s3c}/pm.c (94%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/pm.h (98%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/pwm-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/regs-adc.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/regs-clock.h => mach-s3c/regs-clock-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/regs-clock.h => mach-s3c/regs-clock-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-clock.h rename arch/arm/{mach-s3c24xx => mach-s3c}/regs-dsc.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-gpio-memport.h (100%) rename arch/arm/{mach-s3c24xx/include/mach/regs-gpio.h => mach-s3c/regs-gpio-s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx/include/mach/regs-gpio.h => mach-s3c/regs-gpio-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-gpio.h rename arch/arm/{mach-s3c24xx/include/mach/regs-irq.h => mach-s3c/regs-irq-s3c24xx.h} (98%) rename arch/arm/{mach-s3c64xx/include/mach/regs-irq.h => mach-s3c/regs-irq-s3c64xx.h} (100%) create mode 100644 arch/arm/mach-s3c/regs-irq.h rename arch/arm/{plat-samsung/include/plat => mach-s3c}/regs-irqtype.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/regs-mem.h (98%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-modem.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/regs-s3c2443-clock.h (81%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-srom.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-sys.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-syscon-power.h (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/regs-usb-hsotg-phy.h (100%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/rtc-core.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2410.c (90%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412-power.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2412.c (94%) rename arch/arm/{mach-s3c24xx/include/mach => mach-s3c}/s3c2412.h (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2416.c (88%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2440.c (85%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2442.c (82%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c2443.c (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/s3c244x.c (93%) rename arch/arm/{mach-s3c24xx/common.c => mach-s3c/s3c24xx.c} (98%) rename arch/arm/{mach-s3c24xx/common.h => mach-s3c/s3c24xx.h} (99%) rename arch/arm/{mach-s3c64xx => mach-s3c}/s3c6400.c (89%) rename arch/arm/{mach-s3c64xx => mach-s3c}/s3c6410.c (89%) rename arch/arm/{mach-s3c64xx/common.c => mach-s3c/s3c64xx.c} (97%) rename arch/arm/{mach-s3c64xx/common.h => mach-s3c/s3c64xx.h} (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/samsung-time.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/sdhci.h (99%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-camif.c (95%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-fb-24bpp.c (86%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-i2c.c (79%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-i2c0.c (90%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-i2c1.c (90%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-ide.c (89%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-keypad.c (86%) rename arch/arm/{mach-s3c24xx/setup-sdhci-gpio.c => mach-s3c/setup-sdhci-gpio-s3c24xx.c} (90%) rename arch/arm/{mach-s3c64xx/setup-sdhci-gpio.c => mach-s3c/setup-sdhci-gpio-s3c64xx.c} (95%) rename arch/arm/{mach-s3c24xx/setup-spi.c => mach-s3c/setup-spi-s3c24xx.c} (85%) rename arch/arm/{mach-s3c64xx/setup-spi.c => mach-s3c/setup-spi-s3c64xx.c} (89%) rename arch/arm/{mach-s3c24xx => mach-s3c}/setup-ts.c (88%) rename arch/arm/{mach-s3c64xx => mach-s3c}/setup-usb-phy.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-audio.c (86%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-nor.c (98%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-pm.c (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec-usb.c (96%) rename arch/arm/{mach-s3c24xx => mach-s3c}/simtec.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/sleep-s3c2410.S (92%) rename arch/arm/{mach-s3c24xx => mach-s3c}/sleep-s3c2412.S (93%) rename arch/arm/{mach-s3c24xx/sleep.S => mach-s3c/sleep-s3c24xx.S} (92%) rename arch/arm/{mach-s3c64xx/sleep.S => mach-s3c/sleep-s3c64xx.S} (97%) rename arch/arm/{mach-s3c24xx => mach-s3c}/spi-core.h (100%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/usb-phy.h (100%) rename arch/arm/{mach-s3c24xx => mach-s3c}/vr1000.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/wakeup-mask.c (94%) rename arch/arm/{plat-samsung/include/plat => mach-s3c}/wakeup-mask.h (100%) rename arch/arm/{plat-samsung => mach-s3c}/watchdog-reset.c (100%) rename arch/arm/{mach-s3c64xx => mach-s3c}/watchdog-reset.h (100%) delete mode 100644 arch/arm/mach-s3c24xx/include/mach/fb.h delete mode 100644 arch/arm/mach-s3c24xx/include/mach/io.h delete mode 100644 arch/arm/mach-s3c64xx/include/mach/hardware.h rename arch/arm/plat-samsung/pm-check.c => drivers/soc/samsung/s3c-pm-check.c (99%) rename arch/arm/plat-samsung/pm-debug.c => drivers/soc/samsung/s3c-pm-debug.c (78%) rename arch/arm/plat-samsung/include/plat/regs-spi.h => drivers/spi/spi-s3c24xx-regs.h (95%) rename arch/arm/plat-samsung/include/plat/regs-udc.h => drivers/usb/gadget/udc/s3c2410_udc_regs.h (99%) rename arch/arm/mach-s3c24xx/include/mach/regs-lcd.h => drivers/video/fbdev/s3c2410fb-regs-lcd.h (84%) rename {arch/arm/plat-samsung/include/plat => include/linux/platform_data}/fb-s3c2410.h (57%) rename arch/arm/plat-samsung/include/plat/adc.h => include/linux/soc/samsung/s3c-adc.h (100%) rename arch/arm/plat-samsung/include/plat/cpu-freq.h => include/linux/soc/samsung/s3c-cpu-freq.h (100%) rename arch/arm/plat-samsung/include/plat/cpu-freq-core.h => include/linux/soc/samsung/s3c-cpufreq-core.h (96%) rename arch/arm/plat-samsung/include/plat/pm-common.h => include/linux/soc/samsung/s3c-pm.h (55%) rename drivers/spi/spi-s3c24xx-fiq.h => include/linux/spi/s3c24xx-fiq.h (78%) Cc: linux-kernel(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: patches(a)opensource.cirrus.com Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-clk(a)vger.kernel.org Cc: linux-hwmon(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-pwm(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-fbdev(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org -- 2.20.0

3 3

[alsa-devel] [PATCH 11/36] ARM: s5pv210: split from plat-samsung
by Arnd Bergmann 23 Oct '19

23 Oct '19

These can be build completely independently, so split the two Kconfig symbols. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/Kconfig.debug | 6 +++--- arch/arm/Makefile | 1 - arch/arm/plat-samsung/Kconfig | 2 +- drivers/mmc/host/Kconfig | 2 +- drivers/pwm/Kconfig | 2 +- drivers/spi/Kconfig | 2 +- drivers/tty/serial/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 8 files changed, 9 insertions(+), 10 deletions(-) diff --git a/arch/arm/Kconfig.debug b/arch/arm/Kconfig.debug index 9c4f2d6deb06..4c4e97ae4fcb 100644 --- a/arch/arm/Kconfig.debug +++ b/arch/arm/Kconfig.debug @@ -998,7 +998,7 @@ choice via SCIFA4 on Renesas SH-Mobile AG5 (SH73A0). config DEBUG_S3C_UART0 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1010,7 +1010,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART1 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX @@ -1022,7 +1022,7 @@ choice by the boot-loader before use. config DEBUG_S3C_UART2 - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select DEBUG_EXYNOS_UART if ARCH_EXYNOS select DEBUG_S3C24XX_UART if ARCH_S3C24XX select DEBUG_S3C64XX_UART if ARCH_S3C64XX diff --git a/arch/arm/Makefile b/arch/arm/Makefile index f492d7c338fe..a1bc15cda751 100644 --- a/arch/arm/Makefile +++ b/arch/arm/Makefile @@ -235,7 +235,6 @@ machine-$(CONFIG_PLAT_SPEAR) += spear # by CONFIG_* macro name. plat-$(CONFIG_ARCH_OMAP) += omap plat-$(CONFIG_ARCH_S3C64XX) += samsung -plat-$(CONFIG_ARCH_S5PV210) += samsung plat-$(CONFIG_PLAT_ORION) += orion plat-$(CONFIG_PLAT_PXA) += pxa plat-$(CONFIG_PLAT_S3C24XX) += samsung diff --git a/arch/arm/plat-samsung/Kconfig b/arch/arm/plat-samsung/Kconfig index 740bdb23f38a..1530946cc672 100644 --- a/arch/arm/plat-samsung/Kconfig +++ b/arch/arm/plat-samsung/Kconfig @@ -4,7 +4,7 @@ config PLAT_SAMSUNG bool - depends on PLAT_S3C24XX || ARCH_S3C64XX || ARCH_S5PV210 + depends on PLAT_S3C24XX || ARCH_S3C64XX default y select GENERIC_IRQ_CHIP select NO_IOPORT_MAP diff --git a/drivers/mmc/host/Kconfig b/drivers/mmc/host/Kconfig index 400a581c918c..16a0e5430b44 100644 --- a/drivers/mmc/host/Kconfig +++ b/drivers/mmc/host/Kconfig @@ -275,7 +275,7 @@ config MMC_SDHCI_TEGRA config MMC_SDHCI_S3C tristate "SDHCI support on Samsung S3C SoC" - depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_EXYNOS) + depends on MMC_SDHCI && (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS) help This selects the Secure Digital Host Controller Interface (SDHCI) often referrered to as the HSMMC block in some of the Samsung S3C diff --git a/drivers/pwm/Kconfig b/drivers/pwm/Kconfig index e3a2518503ed..8eb738cac0c7 100644 --- a/drivers/pwm/Kconfig +++ b/drivers/pwm/Kconfig @@ -394,7 +394,7 @@ config PWM_ROCKCHIP config PWM_SAMSUNG tristate "Samsung PWM support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS help Generic PWM framework driver for Samsung. diff --git a/drivers/spi/Kconfig b/drivers/spi/Kconfig index 6f7fdcbb9151..355391ee643d 100644 --- a/drivers/spi/Kconfig +++ b/drivers/spi/Kconfig @@ -625,7 +625,7 @@ config SPI_S3C24XX_FIQ config SPI_S3C64XX tristate "Samsung S3C64XX series type SPI" - depends on (PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST) + depends on (PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST) help SPI driver for Samsung S3C64XX and newer SoCs. diff --git a/drivers/tty/serial/Kconfig b/drivers/tty/serial/Kconfig index 4789b5d62f63..17f01cf3009c 100644 --- a/drivers/tty/serial/Kconfig +++ b/drivers/tty/serial/Kconfig @@ -237,7 +237,7 @@ config SERIAL_CLPS711X_CONSOLE config SERIAL_SAMSUNG tristate "Samsung SoC serial support" - depends on PLAT_SAMSUNG || ARCH_EXYNOS + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS select SERIAL_CORE help Support for the on-chip UARTs on the Samsung S3C24XX series CPUs, diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..7a0035dd9995 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SND_SOC_SAMSUNG tristate "ASoC support for Samsung" - depends on PLAT_SAMSUNG || ARCH_EXYNOS || COMPILE_TEST + depends on PLAT_SAMSUNG || ARCH_S5PV210 || ARCH_EXYNOS || COMPILE_TEST depends on COMMON_CLK select SND_SOC_GENERIC_DMAENGINE_PCM ---help--- -- 2.20.0

5 9

[alsa-devel] [PATCH 21/36] ARM: s3c: move iis pinctrl config into boards
by Arnd Bergmann 23 Oct '19

23 Oct '19

The s3c_gpio_cfgall_range() function is an internal interface of the samsung gpio driver and should not be called directly by drivers, so move the iis pin initialization into the boards. Note that the s3c2412-i2s driver has no boards using it in mainline linux, the driver gets selected for the jive machine but is never instantiated. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-gta02.c | 4 ++++ arch/arm/mach-s3c24xx/mach-h1940.c | 3 +++ arch/arm/mach-s3c24xx/mach-mini2440.c | 5 +++++ arch/arm/mach-s3c24xx/mach-n30.c | 5 +++++ arch/arm/mach-s3c24xx/mach-nexcoder.c | 5 +++++ arch/arm/mach-s3c24xx/mach-otom.c | 6 ++++++ arch/arm/mach-s3c24xx/mach-qt2410.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx1950.c | 3 +++ arch/arm/mach-s3c24xx/mach-rx3715.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2410.c | 5 +++++ arch/arm/mach-s3c24xx/mach-smdk2413.c | 4 ++++ arch/arm/mach-s3c24xx/mach-smdk2440.c | 6 +++++- arch/arm/mach-s3c24xx/mach-vstms.c | 6 +++++- arch/arm/mach-s3c24xx/simtec-audio.c | 6 ++++++ sound/soc/samsung/s3c2412-i2s.c | 7 ------- sound/soc/samsung/s3c24xx-i2s.c | 7 ------- 16 files changed, 63 insertions(+), 16 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 526fd0933289..1ca0460d82f4 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -540,6 +540,10 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 922d53bacbdf..11f705fb5e32 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -709,6 +709,9 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); gpiod_add_lookup_table(&h1940_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/arch/arm/mach-s3c24xx/mach-mini2440.c b/arch/arm/mach-s3c24xx/mach-mini2440.c index 9035f868fb34..bb44b8704d64 100644 --- a/arch/arm/mach-s3c24xx/mach-mini2440.c +++ b/arch/arm/mach-s3c24xx/mach-mini2440.c @@ -685,6 +685,11 @@ static void __init mini2440_init(void) s3c_gpio_setpull(mini2440_buttons[i].gpio, S3C_GPIO_PULL_UP); s3c_gpio_cfgpin(mini2440_buttons[i].gpio, S3C2410_GPIO_INPUT); } + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (features.lcd_index != -1) { int li; diff --git a/arch/arm/mach-s3c24xx/mach-n30.c b/arch/arm/mach-s3c24xx/mach-n30.c index d856f23939af..1836f1cc0992 100644 --- a/arch/arm/mach-s3c24xx/mach-n30.c +++ b/arch/arm/mach-s3c24xx/mach-n30.c @@ -36,6 +36,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/irq.h> @@ -570,6 +571,10 @@ static void __init n30_init(void) S3C2410_MISCCR_USBSUSPND0 | S3C2410_MISCCR_USBSUSPND1, 0x0); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + if (machine_is_n30()) { /* Turn off suspend on both USB ports, and switch the * selectable USB port to USB device mode. */ diff --git a/arch/arm/mach-s3c24xx/mach-nexcoder.c b/arch/arm/mach-s3c24xx/mach-nexcoder.c index c2f34758ccb6..a05c92b6de2e 100644 --- a/arch/arm/mach-s3c24xx/mach-nexcoder.c +++ b/arch/arm/mach-s3c24xx/mach-nexcoder.c @@ -145,6 +145,11 @@ static void __init nexcoder_init_time(void) static void __init nexcoder_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_add_devices(nexcoder_devices, ARRAY_SIZE(nexcoder_devices)); }; diff --git a/arch/arm/mach-s3c24xx/mach-otom.c b/arch/arm/mach-s3c24xx/mach-otom.c index 4e24d89e870b..c834f16e5ba0 100644 --- a/arch/arm/mach-s3c24xx/mach-otom.c +++ b/arch/arm/mach-s3c24xx/mach-otom.c @@ -24,6 +24,8 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -107,6 +109,10 @@ static void __init otom11_init_time(void) static void __init otom11_init(void) { s3c_i2c0_set_platdata(NULL); + + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(otom11_devices, ARRAY_SIZE(otom11_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-qt2410.c b/arch/arm/mach-s3c24xx/mach-qt2410.c index 5d48e5b6e738..3cf8423c76c6 100644 --- a/arch/arm/mach-s3c24xx/mach-qt2410.c +++ b/arch/arm/mach-s3c24xx/mach-qt2410.c @@ -337,6 +337,9 @@ static void __init qt2410_machine_init(void) s3c24xx_udc_set_platdata(&qt2410_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); gpiod_add_lookup_table(&qt2410_spi_gpiod_table); platform_add_devices(qt2410_devices, ARRAY_SIZE(qt2410_devices)); s3c_pm_init(); diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 9f3b0f18cd36..4eeede398775 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -825,6 +825,9 @@ static void __init rx1950_init_machine(void) pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); gpiod_add_lookup_table(&rx1950_audio_gpio_table); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/arch/arm/mach-s3c24xx/mach-rx3715.c b/arch/arm/mach-s3c24xx/mach-rx3715.c index 529c6faf862f..2542c523ad46 100644 --- a/arch/arm/mach-s3c24xx/mach-rx3715.c +++ b/arch/arm/mach-s3c24xx/mach-rx3715.c @@ -39,6 +39,7 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> #include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <plat/cpu.h> #include <plat/devs.h> @@ -199,6 +200,9 @@ static void __init rx3715_init_machine(void) s3c_nand_set_platdata(&rx3715_nand_info); s3c24xx_fb_set_platdata(&rx3715_fb_info); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(rx3715_devices, ARRAY_SIZE(rx3715_devices)); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2410.c b/arch/arm/mach-s3c24xx/mach-smdk2410.c index 18dfef52c8bf..02adaac4d6df 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2410.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2410.c @@ -19,6 +19,8 @@ #include <linux/serial_s3c.h> #include <linux/platform_device.h> #include <linux/io.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <asm/mach/arch.h> #include <asm/mach/map.h> @@ -94,6 +96,9 @@ static void __init smdk2410_init(void) { s3c_i2c0_set_platdata(NULL); platform_add_devices(smdk2410_devices, ARRAY_SIZE(smdk2410_devices)); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-smdk2413.c b/arch/arm/mach-s3c24xx/mach-smdk2413.c index ca80167f268d..2bba4f1fa7ee 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2413.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2413.c @@ -37,6 +37,7 @@ #include <linux/platform_data/i2c-s3c2410.h> #include <mach/gpio-samsung.h> #include <mach/fb.h> +#include <plat/gpio-cfg.h> #include <plat/devs.h> #include <plat/cpu.h> @@ -119,6 +120,9 @@ static void __init smdk2413_machine_init(void) s3c24xx_udc_set_platdata(&smdk2413_udc_cfg); s3c_i2c0_set_platdata(NULL); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2413_devices, ARRAY_SIZE(smdk2413_devices)); smdk_machine_init(); diff --git a/arch/arm/mach-s3c24xx/mach-smdk2440.c b/arch/arm/mach-s3c24xx/mach-smdk2440.c index 7bafcd8ea104..f0fceda3e8e6 100644 --- a/arch/arm/mach-s3c24xx/mach-smdk2440.c +++ b/arch/arm/mach-s3c24xx/mach-smdk2440.c @@ -29,6 +29,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> #include <linux/platform_data/i2c-s3c2410.h> @@ -167,7 +169,9 @@ static void __init smdk2440_machine_init(void) { s3c24xx_fb_set_platdata(&smdk2440_fb_info); s3c_i2c0_set_platdata(NULL); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(smdk2440_devices, ARRAY_SIZE(smdk2440_devices)); smdk_machine_init(); } diff --git a/arch/arm/mach-s3c24xx/mach-vstms.c b/arch/arm/mach-s3c24xx/mach-vstms.c index d76b28b65e65..f333ca05de68 100644 --- a/arch/arm/mach-s3c24xx/mach-vstms.c +++ b/arch/arm/mach-s3c24xx/mach-vstms.c @@ -31,6 +31,8 @@ #include <mach/regs-gpio.h> #include <mach/regs-lcd.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <mach/fb.h> @@ -149,7 +151,9 @@ static void __init vstms_init(void) { s3c_i2c0_set_platdata(NULL); s3c_nand_set_platdata(&vstms_nand_info); - + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); platform_add_devices(vstms_devices, ARRAY_SIZE(vstms_devices)); } diff --git a/arch/arm/mach-s3c24xx/simtec-audio.c b/arch/arm/mach-s3c24xx/simtec-audio.c index 12e17f82dae3..0c12ba4a4704 100644 --- a/arch/arm/mach-s3c24xx/simtec-audio.c +++ b/arch/arm/mach-s3c24xx/simtec-audio.c @@ -14,6 +14,8 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> +#include <mach/gpio-samsung.h> +#include <plat/gpio-cfg.h> #include <linux/platform_data/asoc-s3c24xx_simtec.h> #include <plat/devs.h> @@ -65,6 +67,10 @@ int __init simtec_audio_add(const char *name, bool has_lr_routing, if (has_lr_routing) simtec_audio_platdata.startup = simtec_audio_startup_lrroute; + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); + platform_device_register(&s3c_device_iis); platform_device_register(&simtec_audio_dev); return 0; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 5bf571762e89..fd17765670f1 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -19,9 +19,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> - #include "dma.h" #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" @@ -70,10 +67,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) if (ret) goto err; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - return 0; err: diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 92bdaf0878f8..5ca263f6a39e 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -18,10 +18,7 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/gpio-samsung.h> -#include <plat/gpio-cfg.h> #include "regs-iis.h" - #include "dma.h" #include "s3c24xx-i2s.h" @@ -348,10 +345,6 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) if (ret) return ret; - /* Configure the I2S pins (GPE0...GPE4) in correct mode */ - s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), - S3C_GPIO_PULL_NONE); - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_snd_txctrl(0); -- 2.20.0

2 2

[alsa-devel] [RFC PATCH] ALSA: pcm: fix blocking while loop in snd_pcm_update_hw_ptr0()
by Srinivas Kandagatla 23 Oct '19

23 Oct '19

From: Preetam Singh Ranawat <apranawat(a)codeaurora.org> When period interrupts are disabled, while loop in snd_pcm_update_hw_ptr0() results in the machine locking up if runtime->hw_ptr_buffer_jiffies is 0. Validate runtime->hw_ptr_buffer_jiffies value before while loop to avoid delta check. Signed-off-by: Preetam Singh Ranawat <apranawat(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- This issue was noticed in Android setup from fastcapture path. sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index d80041ea4e01..f407f6895f8a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -341,7 +341,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, * the elapsed time to detect xruns. */ jdelta = curr_jiffies - runtime->hw_ptr_jiffies; - if (jdelta < runtime->hw_ptr_buffer_jiffies / 2) + if ((jdelta < runtime->hw_ptr_buffer_jiffies / 2) || + !runtime->hw_ptr_buffer_jiffies) goto no_delta_check; hdelta = jdelta - delta * HZ / runtime->rate; xrun_threshold = runtime->hw_ptr_buffer_jiffies / 2 + 1; -- 2.21.0

2 1

[alsa-devel] alsa conformance test fails to run for playback (and works for capture)
by Hin-Tak Leung 23 Oct '19

23 Oct '19

Hi Yu-Hsuan, I have encountered another problem with the alsa conformance test. It works for capture, but not for playback. The device is - http://wiki.seeedstudio.com/ReSpeaker_6-Mic_Circular_Array_kit_for_Raspberr… and running on a pi / raspbian buster. The actual message is: Fail - ALSA lib pcm_hw.c:1822:(_snd_pcm_hw_open) Invalid value for card snd_pcm_open hw:CARD=seeed8micvoi: No such device I think the truncation is harmless . The actual device name according to aplay -L is "hw:CARD=seeed8micvoicec,DEV=0" and as I noted earlier, arecord -L shows it as "hw:CARD=seeed8micvoicec,DEV=0" also, and the comformance test running in CAPTURE mode works, just not in PLAYBACK mode. Is there anything I can do to look further? Since it says "invalid value", I tried deleting the parameters around line 190 of src/alsa_conformance_helper.c, but perhaps I should do something else? src/alsa_conformance_helper.c-191- conformance_timer_start(timer, SND_PCM_OPEN); src/alsa_conformance_helper.c:192: rc = snd_pcm_open(handle, src/alsa_conformance_helper.c-193- dev_name, src/alsa_conformance_helper.c-194- stream, src/alsa_conformance_helper.c-195- SND_PCM_NONBLOCK | src/alsa_conformance_helper.c-196- SND_PCM_NO_AUTO_RESAMPLE | src/alsa_conformance_helper.c-197- SND_PCM_NO_AUTO_CHANNELS | src/alsa_conformance_helper.c-198- SND_PCM_NO_AUTO_FORMAT); src/alsa_conformance_helper.c-199- conformance_timer_stop(timer, SND_PCM_OPEN); Regards, Hin-Tak

3 8

[alsa-devel] KASAN: slab-out-of-bounds Read in parse_term_proc_unit
by syzbot 23 Oct '19

23 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=141c4097600000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=f586a582f79122007598 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=169fe2df600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=13e2f3ef600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f586a582f79122007598(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: slab-out-of-bounds in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: slab-out-of-bounds in parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 Read of size 1 at addr ffff8881d48eeca9 by task kworker/1:2/83 CPU: 1 PID: 83 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] parse_term_proc_unit+0x57a/0x5e0 sound/usb/mixer.c:896 __check_input_term+0xc32/0x13f0 sound/usb/mixer.c:984 parse_audio_feature_unit sound/usb/mixer.c:1875 [inline] parse_audio_unit+0x101d/0x36f0 sound/usb/mixer.c:2753 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 83: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x314/0x3050 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 0: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 free_task+0xe3/0x110 kernel/fork.c:472 __put_task_struct+0x1e2/0x4c0 kernel/fork.c:748 put_task_struct include/linux/sched/task.h:119 [inline] delayed_put_task_struct+0x1b4/0x2c0 kernel/exit.c:182 __rcu_reclaim kernel/rcu/rcu.h:222 [inline] rcu_do_batch kernel/rcu/tree.c:2157 [inline] rcu_core+0x630/0x1ca0 kernel/rcu/tree.c:2377 __do_softirq+0x221/0x912 kernel/softirq.c:292 The buggy address belongs to the object at ffff8881d48eec00 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 169 bytes inside of 192-byte region [ffff8881d48eec00, ffff8881d48eecc0) The buggy address belongs to the page: page:ffffea0007523b80 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000753ce40 0000000e0000000e ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d48eeb80: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d48eec00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > ffff8881d48eec80: 00 00 00 00 03 fc fc fc fc fc fc fc fc fc fc fc ^ ffff8881d48eed00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffff8881d48eed80: 00 00 00 00 00 00 00 fc fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] KMSAN: uninit-value in build_audio_procunit
by syzbot 23 Oct '19

23 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 3c8ca708 test_kmsan.c: fix SPDX comment git tree: https://github.com/google/kmsan.git master console output: https://syzkaller.appspot.com/x/log.txt?x=133fb3cf600000 kernel config: https://syzkaller.appspot.com/x/.config?x=c07a3d4f8a59e198 dashboard link: https://syzkaller.appspot.com/bug?extid=85f45d8bd343473faf15 compiler: clang version 9.0.0 (/home/glider/llvm/clang 80fee25776c2fb61e74c1ecb1a523375c2500b69) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=12201cef600000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1281b8ff600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+85f45d8bd343473faf15(a)syzkaller.appspotmail.com usb 1-1: New USB device found, idVendor=1d6b, idProduct=0101, bcdDevice= 0.40 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ===================================================== BUG: KMSAN: uninit-value in build_audio_procunit+0x1be8/0x2490 sound/usb/mixer.c:2438 CPU: 1 PID: 3381 Comm: kworker/1:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x191/0x1f0 lib/dump_stack.c:113 kmsan_report+0x14a/0x2f0 mm/kmsan/kmsan_report.c:110 __msan_warning+0x73/0xf0 mm/kmsan/kmsan_instr.c:245 build_audio_procunit+0x1be8/0x2490 sound/usb/mixer.c:2438 parse_audio_unit+0x115c/0x7490 sound/usb/mixer.c:2483 snd_usb_mixer_controls sound/usb/mixer.c:3095 [inline] snd_usb_create_mixer+0x1d7c/0x4070 sound/usb/mixer.c:3445 usb_audio_probe+0x286b/0x3eb0 sound/usb/card.c:653 usb_probe_interface+0xd19/0x1310 drivers/usb/core/driver.c:361 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_set_configuration+0x309f/0x3710 drivers/usb/core/message.c:2027 generic_probe+0xe7/0x280 drivers/usb/core/generic.c:210 usb_probe_device+0x146/0x200 drivers/usb/core/driver.c:266 really_probe+0xd91/0x1f90 drivers/base/dd.c:552 driver_probe_device+0x1ba/0x510 drivers/base/dd.c:721 __device_attach_driver+0x5b8/0x790 drivers/base/dd.c:828 bus_for_each_drv+0x28e/0x3b0 drivers/base/bus.c:430 __device_attach+0x489/0x750 drivers/base/dd.c:894 device_initial_probe+0x4a/0x60 drivers/base/dd.c:941 bus_probe_device+0x131/0x390 drivers/base/bus.c:490 device_add+0x25b5/0x2df0 drivers/base/core.c:2201 usb_new_device+0x23e5/0x2fb0 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 Uninit was created at: kmsan_save_stack_with_flags mm/kmsan/kmsan.c:151 [inline] kmsan_internal_poison_shadow+0x60/0x110 mm/kmsan/kmsan.c:134 kmsan_slab_alloc+0xaa/0x130 mm/kmsan/kmsan_hooks.c:88 slab_alloc_node mm/slub.c:2792 [inline] slab_alloc mm/slub.c:2801 [inline] __kmalloc+0x28e/0x430 mm/slub.c:3832 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x50d/0x76a0 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0x224/0x2fb0 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5098 [inline] hub_port_connect_change drivers/usb/core/hub.c:5213 [inline] port_event drivers/usb/core/hub.c:5359 [inline] hub_event+0x581d/0x72f0 drivers/usb/core/hub.c:5441 process_one_work+0x1572/0x1ef0 kernel/workqueue.c:2269 worker_thread+0x111b/0x2460 kernel/workqueue.c:2415 kthread+0x4b5/0x4f0 kernel/kthread.c:256 ret_from_fork+0x35/0x40 arch/x86/entry/entry_64.S:355 ===================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH] ALSA: hda: Add Tigerlake/Jasperlake PCI ID
by Pierre-Louis Bossart 23 Oct '19

23 Oct '19

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> Add HD Audio Device PCI ID for the Intel Tigerlake and Jasperlake platform. Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a815bc811799 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2399,6 +2399,12 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Jasperlake */ + { PCI_DEVICE(0x8086, 0x38c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake */ + { PCI_DEVICE(0x8086, 0xa0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- 2.20.1

2 1

[alsa-devel] [PATCH v2 0/5] soundwire: intel/cadence: better initialization
by Pierre-Louis Bossart 22 Oct '19

22 Oct '19

V2 of the original series 'soundwire: inits and PM additions for 5.4', with PM additions removed since more tests on hardware are required. Changes since v1: addressed feedback from Vinod Koul clarified init changes impact Intel and Cadence sides remove unnecessary intermediate variable disable interrupts when exit_reset fails, updated error handling returned -EINVAL on debugfs invalid parameter Pierre-Louis Bossart (5): soundwire: intel/cadence: fix startup sequence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel/cadence: add flag for interrupt enable soundwire: cadence_master: make clock stop exit configurable on init drivers/soundwire/cadence_master.c | 131 +++++++++++++++++++++-------- drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 38 ++++++--- 3 files changed, 126 insertions(+), 48 deletions(-) -- 2.20.1

2 13

[alsa-devel] [PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by twischer＠de.adit-jv.com 22 Oct '19

22 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif dma address is calculated by following calculation formulation: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formulation to correct ssi9 4/5/6/7 busif data register address Fixes: commit 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c..28f65eb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.7.4

4 7

[alsa-devel] Recording on WM9712
by Matej Kupljen 22 Oct '19

22 Oct '19

Hi all, I have an iMX6q board with external AC97 WM9712 codec connected to SSI on iMX6. The card is detected correctly and it is using generic AC97 codec driver, not the specific one for WM9712. I was able to hard code the values into fsl-asoc-card.c and fsl_ssi.c files, but my problem still remains. I can playback WAV file normally, also speaker-test works as expected. The problem is when I try to record something from Line In. It does record the file, but when I playback it, it is "too slow". I tried different sample rates, but the problem is only worse (strange beeping in the recording). The command for recording is: # arecord -f dat -vv /tmp/record.wav (I tired also with -M option, but there is no difference). If I copy the recorded file to PC and use audacity to play it back using 1.5x speed, the recording seems O.K. Checking codec register values trough /proc/asound/ac97audio/codec97#0/ac97#0-0+regs it seems that the sample rates are set O.K. to 48kHz. Any idea what might be wrong? Is there anything else I can check? Thanks and BR, Matej

1 0

[alsa-devel] [REGRESSION 5.3] WARNING: pulseaudio/1297 still has locks held!
by Russell King - ARM Linux admin 22 Oct '19

22 Oct '19

When upgrading from 5.2 to 5.3, I now encounter this error after boot: ==================================== WARNING: pulseaudio/1297 still has locks held! 5.3.0+ #1826 Not tainted ------------------------------------ 1 lock held by pulseaudio/1297: #0: ee815308 (&hcp->lock){....}, at: hdmi_codec_startup+0x20/0x130 stack backtrace: CPU: 0 PID: 1297 Comm: pulseaudio Not tainted 5.3.0+ #1826 Hardware name: Marvell Dove (Cubox) [<c0017b4c>] (unwind_backtrace) from [<c0014d10>] (show_stack+0x10/0x14) [<c0014d10>] (show_stack) from [<c00a2d18>] (futex_wait_queue_me+0x13c/0x19c) [<c00a2d18>] (futex_wait_queue_me) from [<c00a3630>] (futex_wait+0x184/0x24c) [<c00a3630>] (futex_wait) from [<c00a5e1c>] (do_futex+0x334/0x598) [<c00a5e1c>] (do_futex) from [<c00a62e8>] (sys_futex_time32+0x118/0x180) [<c00a62e8>] (sys_futex_time32) from [<c0009000>] (ret_fast_syscall+0x0/0x54) Exception stack(0xebd31fa8 to 0xebd31ff0) 1fa0: 00000000 ffffffff 000c8748 00000189 00000001 00000000 1fc0: 00000000 ffffffff 00000000 000000f0 00000000 00000000 00000000 00056200 1fe0: 000000f0 beac03a8 b6d6c835 b6d6f456 This seems to be caused by: eb1ecadb7f67 ("ASoC: hdmi-codec: re-introduce mutex locking") which holds a mutex and returns to userspace. Meanwhile, userspace issues a futex operation, which ends up in futex_wait_queue_me(). futex_wait_queue_me() can wait, and if the system will be hibernating, is a potential hibernation point. One of the requirements there is that no locks shall be held, and the freezer code verifies that by calling debug_check_no_locks_held(). With lockdep enabled, it reveals that hdmi-codec is indeed holding a lock. Reverting this commit solves the problem. -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

3 2

[alsa-devel] [PATCH v6 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 22 Oct '19

22 Oct '19

Hey, here's the 6th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v6 changes: - Since v5, changes in patches 002, 003 and 005. - Fixed a bug with S3 suspend/resume during HDMI playback on Intel ICL platforms. As part of the bug fix, cleaned up the new display power control code added in v5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 114 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 97 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 345 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

3 17

[alsa-devel] sof-skl_hda: add device definition to speaker disable sequence
by GitHub pull_request - opened 22 Oct '19

22 Oct '19

alsa-project/alsa-ucm-conf pull request #3 was opened from juimonen: For example Pulseaudio is not able to run the disable sequence if the device is not defined. So add the definition. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [asoc:for-5.5 122/122] sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 22 Oct '19

22 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: c173dba44c2d2ec2adaa4e607e9729b74236bd57 commit: c173dba44c2d2ec2adaa4e607e9729b74236bd57 [122/122] ASoC: tas2562: Introduce the TAS2562 amplifier reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-42-g38eda53-dirty git checkout c173dba44c2d2ec2adaa4e607e9729b74236bd57 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2562.c:195:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:202:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2562.c:209:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2562.c:246:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2562.c:246:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2562.c:246:58: sparse: got restricted snd_pcm_format_t sound/soc/codecs/tas2562.c:472:39: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +195 sound/soc/codecs/tas2562.c 189 190 static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) 191 { 192 int ret; 193 194 switch (bitwidth) { > 195 case SNDRV_PCM_FORMAT_S16_LE: 196 snd_soc_component_update_bits(tas2562->component, 197 TAS2562_TDM_CFG2, 198 TAS2562_TDM_CFG2_RXWLEN_MASK, 199 TAS2562_TDM_CFG2_RXWLEN_16B); 200 tas2562->v_sense_slot = tas2562->i_sense_slot + 2; 201 break; 202 case SNDRV_PCM_FORMAT_S24_LE: 203 snd_soc_component_update_bits(tas2562->component, 204 TAS2562_TDM_CFG2, 205 TAS2562_TDM_CFG2_RXWLEN_MASK, 206 TAS2562_TDM_CFG2_RXWLEN_24B); 207 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 208 break; 209 case SNDRV_PCM_FORMAT_S32_LE: 210 snd_soc_component_update_bits(tas2562->component, 211 TAS2562_TDM_CFG2, 212 TAS2562_TDM_CFG2_RXWLEN_MASK, 213 TAS2562_TDM_CFG2_RXWLEN_32B); 214 tas2562->v_sense_slot = tas2562->i_sense_slot + 4; 215 break; 216 217 default: 218 dev_info(tas2562->dev, "Not supported params format\n"); 219 } 220 221 ret = snd_soc_component_update_bits(tas2562->component, 222 TAS2562_TDM_CFG5, 223 TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, 224 TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); 225 if (ret < 0) 226 return ret; 227 228 ret = snd_soc_component_update_bits(tas2562->component, 229 TAS2562_TDM_CFG6, 230 TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, 231 TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); 232 if (ret < 0) 233 return ret; 234 235 return 0; 236 } 237 238 static int tas2562_hw_params(struct snd_pcm_substream *substream, 239 struct snd_pcm_hw_params *params, 240 struct snd_soc_dai *dai) 241 { 242 struct snd_soc_component *component = dai->component; 243 struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); 244 int ret; 245 > 246 ret = tas2562_set_bitwidth(tas2562, params_format(params)); 247 if (ret) { 248 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 249 return ret; 250 } 251 252 ret = tas2562_set_samplerate(tas2562, params_rate(params)); 253 if (ret) 254 dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); 255 256 return ret; 257 } 258 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

5 4

[alsa-devel] [PATCH v2] ASoC: samsung: i2s: Document clocks macros
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Document clocks macros with their description from 'Documentation/devicetree/bindings/sound/samsung-i2s.txt' Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added proper commit message and description - Moved comments to the right side Best regards, Maciej Falkowski --- include/dt-bindings/sound/samsung-i2s.h | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..250de0d6c734 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,14 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +#define CLK_I2S_CDCLK 0 /* the CDCLK (CODECLKO) gate clock */ + +#define CLK_I2S_RCLK_SRC 1 /* the RCLKSRC mux clock (corresponding to + * RCLKSRC bit in IISMOD register) + */ + +#define CLK_I2S_RCLK_PSR 2 /* the RCLK prescaler divider clock + * (corresponding to the IISPSR register) + */ #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

4 3

[alsa-devel] [PATCH 1/2] ASoC: msm8916-wcd-analog: Fix RX1 selection in RDAC2 MUX
by Stephan Gerhold 21 Oct '19

21 Oct '19

According to the PM8916 Hardware Register Description, CDC_D_CDC_CONN_HPHR_DAC_CTL has only a single bit (RX_SEL) to switch between RX1 (0) and RX2 (1). It is not possible to disable it entirely to achieve the "ZERO" state. However, at the moment the "RDAC2 MUX" mixer defines three possible values ("ZERO", "RX2" and "RX1"). Setting the mixer to "ZERO" actually configures it to RX1. Setting the mixer to "RX1" has (seemingly) no effect. Remove "ZERO" and replace it with "RX1" to fix this. Fixes: 585e881e5b9e ("ASoC: codecs: Add msm8916-wcd analog codec") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 667e9f73aba3..e3d311fb510e 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -306,7 +306,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -321,7 +321,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) -- 2.23.0

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[alsa-devel] [PATCH] ASoC: compress: fix unsigned integer overflow check
by Srinivas Kandagatla 21 Oct '19

21 Oct '19

From: Xiaojun Sang <xsang(a)codeaurora.org> Parameter fragments and fragment_size are type of u32. U32_MAX is the correct check. Signed-off-by: Xiaojun Sang <xsang(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 41905afada63..f34ce564d92c 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -528,7 +528,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments > U32_MAX / params->buffer.fragment_size || params->buffer.fragments == 0) return -EINVAL; -- 2.21.0

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[alsa-devel] [PATCH v5 0/4] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 21 Oct '19

21 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). ASoC: cros_ec_codec: support WoV - makes cros_ec_codec support WoV. ASoC: mediatek: mt6358: support WoV - sets necessary registers on mt6358 to support WoV. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV - make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. Changes from v4: - remove the first 6 commits which have merged - fix a compile error by adding Kconfig dependency https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156992.h… Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (4): ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 1 + .../linux/platform_data/cros_ec_commands.h | 69 ++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/cros_ec_codec.c | 700 +++++++++++++++++- sound/soc/codecs/mt6358.c | 105 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 946 insertions(+), 5 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

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[alsa-devel] [PATCH 0/6] ALSA: firewire: handle several IR/IT context in callback of the IT context
by Takashi Sakamoto 21 Oct '19

21 Oct '19

Hi, This patchset is a part of my continuous work to improve ALSA IEC 61883-1/6 packet streaming engine for clock-recovery, addressed in my previous message: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/153388.ht… For the clock-recovery, information in the sequence of tx packet from device should be used to generate the sequence of rx packet to the device. In current implementation of the engine, the packets are processed in different tasklet contexts for each IR/IT context. This is inconvenient to bypass information between these IR/IT contexts. In this patchset, the engine is improved to process all of IR/IT contexts in the same domain by a tasklet context for one of IT context. For convenience, the IT context is called as 'IRQ target'. As a result, 1394 OHCI controller is managed to generate hardware IRQ just for the IRQ target. All of rx/tx packets are processed in tasklet for the hardware IRQ. Takashi Sakamoto (6): ALSA: firewire-lib: add irq_target member into amdtp_domain struct ALSA: firewire-lib: replace pointer callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: replace ack callback to flush isoc contexts in AMDTP domain ALSA: firewire-lib: cancel flushing isoc context in the laste step to process context callback ALSA: firewire-lib: handle several AMDTP streams in callback handler of IRQ target ALSA: firewire-lib: postpone to start IR context sound/firewire/amdtp-stream.c | 329 +++++++++++++++----- sound/firewire/amdtp-stream.h | 17 +- sound/firewire/bebob/bebob_pcm.c | 18 +- sound/firewire/bebob/bebob_stream.c | 10 +- sound/firewire/dice/dice-pcm.c | 8 +- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 8 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-pcm.c | 8 +- sound/firewire/fireface/ff-stream.c | 10 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-pcm.c | 8 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 8 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-pcm.c | 8 +- sound/firewire/tascam/tascam-stream.c | 2 +- 18 files changed, 325 insertions(+), 129 deletions(-) -- 2.20.1

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[alsa-devel] [sound:for-next 37/38] include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used
by kbuild test robot 21 Oct '19

21 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: bacd861452d2be86a4df341b12e32db7dac8021e commit: 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 [37/38] PCI: Add a helper to check Power Resource Requirements _PR3 existence config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 52525b7a3cf82adec5c6cf0ecbd23ff228badc94 # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=sparc64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): In file included from arch/sparc/kernel/iommu.c:20:0: >> include/linux/pci.h:2317:13: error: 'pci_pr3_present' defined but not used [-Werror=unused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: all warnings being treated as errors vim +/pci_pr3_present +2317 include/linux/pci.h 2310 2311 void 2312 pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); 2313 bool pci_pr3_present(struct pci_dev *pdev); 2314 #else 2315 static inline struct irq_domain * 2316 pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } > 2317 static bool pci_pr3_present(struct pci_dev *pdev) { return false; } 2318 #endif 2319 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [GIT PULL] ASoC fixes for 5.4
by Mark Brown 21 Oct '19

21 Oct '19

The following changes since commit 147162f575152db80000fb3de26358264768ee9f: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (2019-09-20 12:06:48 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.4-rc4 for you to fetch changes up to 95a32c98055f664f9b3f34c41e153d4dcedd0eff: ASoC: SOF: control: return true when kcontrol values change (2019-10-18 18:09:22 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.4 A collection of fixes that have arrived since the merge window. There are a small number of core fixes here but they are smaller ones around error handling. ---------------------------------------------------------------- Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Chuhong Yuan (1): ASoC: Intel: sof-rt5682: add a check for devm_clk_get Dan Carpenter (2): ASoC: soc-component: fix a couple missing error assignments ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create() Daniel Baluta (1): ASoC: simple_card_utils.h: Fix potential multiple redefinition error Dragos Tarcatu (1): ASoC: SOF: control: return true when kcontrol values change Hans de Goede (1): ASoC: core: Fix pcm code debugfs error Jaska Uimonen (3): ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack ASoC: intel: bytcr_rt5651: add null check to support_button_press Junya Monden (1): ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Li Xu (1): ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region Liam Girdwood (1): ASoC: SOF: Intel: initialise and verify FW crash dump data. Olivier Moysan (1): ASoC: stm32: sai: fix sysclk management on shutdown Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (1): ASoC: SOF: loader: fix kernel oops on firmware boot failure Ranjani Sridharan (1): ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture Robin Murphy (1): ASoc: rockchip: i2s: Fix RPM imbalance Sathyanarayana Nujella (1): ASoC: max98373: check for device node before parsing Srinivas Kandagatla (1): ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2 Stuart Henderson (1): ASoC: wm_adsp: Don't generate kcontrols without READ flags Sylwester Nawrocki (2): ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Do not register inapplicable controls for WM1811 include/sound/simple_card_utils.h | 8 ++--- sound/soc/codecs/max98373.c | 20 +++++++----- sound/soc/codecs/msm8916-wcd-digital.c | 22 +++++++++++++ sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/codecs/wm8994.c | 43 ++++++++++++++---------- sound/soc/codecs/wm_adsp.c | 10 ++++-- sound/soc/intel/boards/sof_rt5682.c | 60 +++++++++++++++++++++++++++++++--- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/samsung/arndale_rt5631.c | 34 ++++++++++++++++--- sound/soc/sh/rcar/core.c | 1 + sound/soc/soc-pcm.c | 17 +++++++--- sound/soc/soc-topology.c | 2 +- sound/soc/sof/control.c | 26 ++++++++++----- sound/soc/sof/intel/Kconfig | 10 ++++++ sound/soc/sof/intel/bdw.c | 7 ++++ sound/soc/sof/intel/byt.c | 6 ++++ sound/soc/sof/intel/hda-ctrl.c | 12 +++---- sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++---- sound/soc/sof/intel/hda.c | 7 ++++ sound/soc/sof/intel/hda.h | 5 ++- sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++------- sound/soc/sof/topology.c | 4 ++- sound/soc/stm/stm32_sai_sub.c | 21 ++++++------ 26 files changed, 320 insertions(+), 97 deletions(-)

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[alsa-devel] ALC711 support
by Kailang 21 Oct '19

21 Oct '19

Hi Takashi, Attach patch was support ALC711 HDA mode. BR, Kailang

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[alsa-devel] Question about "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params"
by S.j. Wang 21 Oct '19

21 Oct '19

Hi Jiada, Mark After this commit "957ce0c6b8a1 ASoC: soc-pcm: check symmetry after hw_params" The soc_pcm_params_symmetry can't take effect, because its position is after the change of cpu_dai->rate. The code of soc_pcm_params_symmetry is: static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... if (symmetry && cpu_dai->rate && cpu_dai->rate != rate) { dev_err(rtd->dev, "ASoC: unmatched rate symmetry: %d - %d\n", cpu_dai->rate, rate); return -EINVAL; } ... } This code soc_pcm_hw_params of is: static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { ... /* store the parameters for each DAIs */ cpu_dai->rate = params_rate(params); cpu_dai->channels = params_channels(params); cpu_dai->sample_bits = snd_pcm_format_physical_width(params_format(params)); snd_soc_dapm_update_dai(substream, params, cpu_dai); ... ret = soc_pcm_params_symmetry(substream, params); if (ret) goto component_err; ... } Is this an issue? Best regards Wang shengjiu

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[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos Odroid XU3/XU4 audio complex to dt-schema
by Marek Szyprowski 21 Oct '19

21 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec to newer dt-schema format. 'clocks' property is unneeded in the bindings and is left undefined in 'properties'. 'samsung,audio-widgets' and 'samsung,audio-routing' are optional from driver perspective and they are set as unrequired. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,odroid.txt | 54 ----------- .../bindings/sound/samsung,odroid.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - samsung,audio-routing - a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - For Odroid XU4: - no entries - -Required sub-nodes: - - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - -Example: - -sound { - compatible = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..c6b244352d05 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the CODEC nodes, + first entry must be corresponding to the MAX98090 CODEC and + the second entry must be the phandle of the HDMI IP block node. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + +required: + - compatible + - model + - cpu + - codec + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; + -- 2.17.1

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[alsa-devel] [PATCH 0/6] soundwire: intel/cadence: simplify PDI handling
by Pierre-Louis Bossart 21 Oct '19

21 Oct '19

These patches were originally submitted as '[RFC PATCH 00/11] soundwire: intel: simplify DAI/PDI handling'. There were no comments received. This series only provides the PDI changes, which makes it simpler to review. The DAI changes will be provided with the complete series for ASoC/SOF integration, which is a larger change. Bard Liao (3): soundwire: intel: fix intel_register_dai PDI offsets and numbers soundwire: intel: remove playback/capture stream_name soundwire: cadence_master: improve PDI allocation Pierre-Louis Bossart (3): soundwire: remove DAI_ID_RANGE definitions soundwire: cadence/intel: simplify PDI/port mapping soundwire: intel: don't filter out PDI0/1 drivers/soundwire/cadence_master.c | 158 +++++++---------------------- drivers/soundwire/cadence_master.h | 34 ++----- drivers/soundwire/intel.c | 155 ++++++---------------------- include/linux/soundwire/sdw.h | 3 - 4 files changed, 73 insertions(+), 277 deletions(-) -- 2.20.1

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[alsa-devel] Fwd: [PATCH 29/46] Input: touchscreen: use wrapper for pxa2xx ac97 registers
by Arnd Bergmann 18 Oct '19

18 Oct '19

On Fri, Oct 18, 2019 at 8:48 PM Dmitry Torokhov <dmitry.torokhov(a)gmail.com> wrote: > > On Fri, Oct 18, 2019 at 05:41:44PM +0200, Arnd Bergmann wrote: > > To avoid a dependency on the pxa platform header files with > > hardcoded registers, change the driver to call a wrapper > > in the pxa2xx-ac97-lib that encapsulates all the other > > ac97 stuff. > > Not supper happy about adding module dependencies. Can we include > mach/regs-ac97.h from include/sound/pxa2xx-lib.h and use static inlines? > Someone needs to include mach/regs-ac97.h in the end... > > Or there is something later in the series that needs it? One of the goals of the series is to completely remove all mach/*.h headers and place them somewhere else, ideally inaccessible to device drivers. In case of mach/regs-ac97.h, the later patch "ASoC: pxa: ac97: use normal MMIO accessors" passes the physical register base address as a platform device resource that gets ioremapped in the ac97 driver, rather than hardcoding the virtual address in a global header. I agree that the exported function is still ugly, but I hope it's enough of an improvement over the previous state that we can do it anyway. If you have any better ideas though, I can give that a try, too. One possibility would be a higher-level interface exported on top of 'struct snd_ac97', but I could not figure out how to do this. Arnd

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[alsa-devel] [PATCH v4 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v3: - fix a compile error by adding ifdef https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156750.h… - fix a coccinelle warning https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156754.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

2 16

[alsa-devel] [PATCH] ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting
by Eugeniu Rosca 18 Oct '19

18 Oct '19

From: Junya Monden <jmonden(a)jp.adit-jv.com> Unlike other format-related DAI parameters, rdai->bit_clk_inv flag is not properly re-initialized when setting format for new stream processing. The inversion, if requested, is then applied not to default, but to a previous value, which leads to SCKP bit in SSICR register being set incorrectly. Fix this by re-setting the flag to its initial value, determined by format. Fixes: 1a7889ca8aba3 ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior") Cc: Andrew Gabbasov <andrew_gabbasov(a)mentor.com> Cc: Jiada Wang <jiada_wang(a)mentor.com> Cc: Timo Wischer <twischer(a)de.adit-jv.com> Cc: stable(a)vger.kernel.org # v3.17+ Signed-off-by: Junya Monden <jmonden(a)jp.adit-jv.com> Signed-off-by: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bda5b958d0dc..e9596c2096cd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: Intel: sof-rt5682: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

sof_audio_probe misses a check for devm_clk_get and may cause problems. Add a check for it to fix the bug. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a437567b8cee..6d15c7ff66bf 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -576,6 +576,15 @@ static int sof_audio_probe(struct platform_device *pdev) /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + ret = clk_prepare_enable(ctx->mclk); if (ret < 0) { dev_err(&pdev->dev, -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: stm32: sai: fix sysclk management on shutdown
by Olivier Moysan 18 Oct '19

18 Oct '19

The commit below, adds a call to sysclk callback on shutdown. This introduces a regression in stm32 SAI driver, as some clock services are called twice, leading to unbalanced calls. Move processing related to mclk from shutdown to sysclk callback. When requested frequency is 0, assume shutdown and release mclk. Fixes: 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7501f88aaa6..a4060813bc74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - (unsigned int)~SAI_XCR1_NODIV); + freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; + /* Assume shutdown if requested frequency is 0Hz */ + if (!freq) { + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } + return 0; + } + /* If master clock is used, set parent clock now */ ret = stm32_sai_set_parent_clock(sai, freq); if (ret) @@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - SAI_XCR1_NODIV); - - /* Release mclk rate only if rate was actually set */ - if (sai->mclk_rate) { - clk_rate_exclusive_put(sai->sai_mclk); - sai->mclk_rate = 0; - } - clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] ASoc: Add Texas Instruments TAS2562 amplifier binding
by Dan Murphy 18 Oct '19

18 Oct '19

Add the DT binding for the TAS2562 amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt new file mode 100644 index 000000000000..658e1fb18a99 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS2562 Smart PA + +The TAS2562 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - compatible: - Should contain "ti,tas2562". + - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. + - ti,imon-slot-no:- TDM TX current sense time slot. + +Optional properties: +- interrupt-parent: phandle to the interrupt controller which provides + the interrupt. +- interrupts: (GPIO) interrupt to which the chip is connected. +- shut-down: GPIO used to control the state of the device. + +Examples: +tas2562@4c { + #address-cells = <1>; + #size-cells = <0>; + compatible = "ti,tas2562"; + reg = <0x4c>; + + interrupt-parent = <&gpio1>; + interrupts = <14>; + + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; +}; + -- 2.22.0.214.g8dca754b1e

3 4

[alsa-devel] [PATCH v3 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Tzung-Bi Shih (10): WIP: platform/chrome: cros_ec: remove unused EC feature WIP: ASoC: cros_ec_codec: refactor I2S RX WIP: ASoC: cros_ec_codec: extract DMIC EC command from I2S RX WIP: platform/chrome: cros_ec: add common commands for EC codec WIP: ASoC: cros_ec_codec: read max DMIC gain from EC codec WIP: ASoC: dt-bindings: cros_ec_codec: add SHM bindings WIP: ASoC: cros_ec_codec: support WoV WIP: ASoC: mediatek: mt6358: support WoV WIP: ASoC: dt-bindings: mt8183: add ec-codec WIP: ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

5 28

[alsa-devel] [PATCH] ASoC: SOF: control: return true when kcontrol values change
by Pierre-Louis Bossart 18 Oct '19

18 Oct '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> All the kcontrol put() functions are currently returning 0 when successful. This does not go well with alsamixer as it does not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks when values change for (some of) the sof kcontrols. This patch fixes that by returning true for volume, switch and enum type kcontrols when values do change in put(). Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 2c4abd406c4f..7baf7f1507c3 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -91,13 +91,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -107,8 +110,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -136,11 +138,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } if (scontrol->led_ctl.use_led) @@ -154,7 +160,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -182,11 +188,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -197,7 +207,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, -- 2.20.1

2 1

[alsa-devel] [PATCH v3] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 18 Oct '19

18 Oct '19

The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <mripard(a)kernel.org> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 267 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------ 2 files changed, 267 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..b8f89c7258eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.23.0

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

aic32x4_set_dai_sysclk misses a check for devm_clk_get and may miss the failure. Add a check to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 68165de1c8de..b4e9a6c73f90 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -573,6 +573,9 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct clk *pll; pll = devm_clk_get(component->dev, "pll"); + if (IS_ERR(pll)) + return PTR_ERR(pll); + mclk = clk_get_parent(pll); return clk_set_rate(mclk, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH][next] ASoC: rt1011: fix spelling mistake "temperture" -> "temperature"
by Colin King 18 Oct '19

18 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_dbg message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index ad049cfddcb0..dcd397a83cb4 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2373,7 +2373,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) device_property_read_u32(dev, "realtek,r0_calib", &rt1011->r0_calib); - dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + dev_dbg(dev, "%s: r0_calib: 0x%x, temperature_calib: 0x%x", __func__, rt1011->r0_calib, rt1011->temperature_calib); return 0; -- 2.20.1

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[alsa-devel] [PATCH v6] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 18 Oct '19

18 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: integrated fix for minor spelling issues] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v6: - integrated fix for minor spelling issues - collected tags --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 138 ++++++++++++++++++ 2 files changed, 138 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..53e3bad4178c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,138 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH 00/12] ALSA: firewire: configure interval of hw IRQ via ALSA PCM interface
by Takashi Sakamoto 18 Oct '19

18 Oct '19

Hi, ALSA IEC 61883-1/6 packet streaming engine controls OHCI 1394 controller to generate hardware IRQ for fixed number (=16) of isoc packets. Additionally, the size of packet buffer is fixed for triple of interval of hardware IRQ (=48). This commit allows ALSA PCM applications to configure these parameters by ALSA PCM interface. As a result, the size of PCM period is available in the range between 125 usec and 64 msec (or 32 msec). The hardware IRQ is scheduled for isoc packet which includes event across PCM period boundary. Takashi Sakamoto (12): ALSA: firewire-lib: use variable size of queue for isoc packets instead of fixed size ALSA: bebob: share PCM buffer size for both direction ALSA: fireworks: share PCM buffer size for both direction ALSA: oxfw: share PCM buffer size for both direction ALSA: dice: share PCM buffer size for both direction ALSA: firewire-digi00x: share PCM buffer size for both direction ALSA: firewire-tascam: share PCM buffer size for both direction ALSA: firewire-motu: share PCM buffer size for both direction ALSA: fireface: share PCM buffer size for both direction ALSA: firewire-lib: schedule hardware IRQ according to the size of PCM period ALSA: firewire-lib: tune the maximum available size of PCM period ALSA: firewire-lib: tune the minimum available size of PCM period sound/firewire/amdtp-stream.c | 154 ++++++++++++++------ sound/firewire/amdtp-stream.h | 9 +- sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 5 +- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 20 ++- sound/firewire/dice/dice-stream.c | 5 +- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 12 +- sound/firewire/digi00x/digi00x-stream.c | 5 +- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 13 +- sound/firewire/fireface/ff-stream.c | 5 +- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 5 +- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 12 +- sound/firewire/motu/motu-stream.c | 5 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 17 ++- sound/firewire/oxfw/oxfw-stream.c | 5 +- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 12 +- sound/firewire/tascam/tascam-stream.c | 5 +- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 266 insertions(+), 85 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 18 Oct '19

18 Oct '19

For P2P output, the output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 40 +++++++++++++++++++++++++++++++--------- 1 file changed, 31 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..44d05ec28bd3 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; if (div[IN] == 0) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); @@ -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; - else - div[OUT] = clk_get_rate(clk) / outrate; + /* + * When P2P mode, output rate should align with the out samplerate. + * if set too high output rate, there will be lots of Overload. + * When M2M mode, output rate should also need to align with the out + * samplerate, but M2M must use less time to achieve good performance. + */ + clk_rate = clk_get_rate(clk); + if (p2p || !ideal) { + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; + } else { + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); + div[OUT] = clk_rate; + } if (div[OUT] == 0) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || + rem[IN] != 0 || rem[OUT] != 0)) { + pair_err("The divider can't be used for non ideal mode\n"); + return -EINVAL; + } + + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +582,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, true); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v3] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 17 Oct '19

17 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties, added list of values for widgets, minor other fixes] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 -------- .../bindings/sound/samsung,tm2-audio.yaml | 99 +++++++++++++++++++ 2 files changed, 99 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..c9178d928ad4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are: + the WM5110's pins: + "HPOUT1L", + "HPOUT1R", + "HPOUT2L", + "HPOUT2R", + "HPOUT3L", + "HPOUT3R", + MAX98504's pins: + "SPKOUT" + and the jacks on the board: + "HP", + "SPK", + "RCV", + "Main Mic", + "Sub Mic", + "Third Mic", + "Headset Mic". + +required: + - compatible + - model + - i2s-controller + - audio-codec + - audio-amplifier + - mic-bias-gpios + - samsung,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-codec = <&wm5110>, <&hdmi>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

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Re: [alsa-devel] linux-next: Tree for Oct 17 (soundwire)
by Randy Dunlap 17 Oct '19

17 Oct '19

On 10/16/19 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191016: > on x86_64: # CONFIG_BLOCK is not set CC [M] drivers/soundwire/intel_init.o In file included from ../drivers/soundwire/intel_init.c:12:0: ../include/linux/iomap.h: In function ‘iomap_sector’: ../include/linux/iomap.h:76:48: error: ‘SECTOR_SHIFT’ undeclared (first use in this function); did you mean ‘SECTIONS_SHIFT’? return (iomap->addr + pos - iomap->offset) >> SECTOR_SHIFT; ^~~~~~~~~~~~ SECTIONS_SHIFT -- ~Randy

2 1

[alsa-devel] [PATCH] ALSA: hda - Force runtime PM on Nvidia HDMI codecs
by Lukas Wunner 17 Oct '19

17 Oct '19

Przemysław Kopa reports that since commit b516ea586d71 ("PCI: Enable NVIDIA HDA controllers"), the discrete GPU Nvidia GeForce GT 540M on his 2011 Samsung laptop refuses to runtime suspend, resulting in a power regression and excessive heat. Rivera Valdez witnesses the same issue with a GeForce GT 525M (GF108M) of the same era, as does another Arch Linux user named "R0AR" with a more recent GeForce GTX 1050 Ti (GP107M). The commit exposes the discrete GPU's HDA controller and all four codecs on the controller do not set the CLKSTOP and EPSS bits in the Supported Power States Response. They also do not set the PS-ClkStopOk bit in the Get Power State Response. hda_codec_runtime_suspend() therefore does not call snd_hdac_codec_link_down(), which prevents each codec and the PCI device from runtime suspending. The same issue is present on some AMD discrete GPUs and we addressed it by forcing runtime PM despite the bits not being set, see commit 57cb54e53bdd ("ALSA: hda - Force to link down at runtime suspend on ATI/AMD HDMI"). Do the same for Nvidia HDMI codecs. Fixes: b516ea586d71 ("PCI: Enable NVIDIA HDA controllers") Link: https://bbs.archlinux.org/viewtopic.php?pid=1865512 Link: https://bugs.freedesktop.org/show_bug.cgi?id=75985#c81 Reported-by: Przemysław Kopa <prymoo(a)gmail.com> Reported-by: Rivera Valdez <riveravaldez(a)ysinembargo.com> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> Cc: Daniel Drake <dan(a)reactivated.net> Cc: stable(a)vger.kernel.org # v5.3+ --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bca5de7..795cbda 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH 1/7] ASoC: amd: No need PCI-MSI interrupts
by Ravulapati Vishnu vardhan rao 17 Oct '19

17 Oct '19

ACP-PCI controller driver does not depends msi interrupts. So removed msi related pci functions which have no use and does not impact on existing functionality. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..8f6bf00 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -46,14 +46,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } - /* check for msi interrupt support */ - ret = pci_enable_msi(pci); - if (ret) - /* msi is not enabled */ - irqflags = IRQF_SHARED; - else - /* msi is enabled */ - irqflags = 0; + irqflags = 0; addr = pci_resource_start(pci, 0); adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); @@ -112,7 +105,6 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); @@ -129,7 +121,6 @@ static void snd_acp3x_remove(struct pci_dev *pci) platform_device_unregister(adata->pdev); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.7.4

6 23

[alsa-devel] [PATCH -next] ASoC: atmel: Fix build error
by YueHaibing 17 Oct '19

17 Oct '19

when do randbuilding, I got this error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x12f6): undefined reference to `atmel_pcm_pdc_platform_register' This is because SND_ATMEL_SOC_SSC_DMA=y, SND_ATMEL_SOC_SSC=y, but SND_ATMEL_SOC_SSC_PDC=m. Fix it bt reintroducing the default Kconfig. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/atmel/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c22..79e45f2 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -12,10 +12,14 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC tristate depends on HAS_DMA + default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM + default m if SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_DMA=y || (SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_SSC tristate -- 2.7.4

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[alsa-devel] [sound:for-next 2/20] drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present'
by kbuild test robot 17 Oct '19

17 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: fa5a68b2abcefe2204075c80662e02cf59567a9f commit: 2e1a529c10e18e62da55eac42c278f11216dddaa [2/20] PCI: Add pci_pr3_present() helper to check Power Resource for D3hot config: parisc-c3000_defconfig (attached as .config) compiler: hppa-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 2e1a529c10e18e62da55eac42c278f11216dddaa # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=parisc If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): >> drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present' bool pci_pr3_present(struct pci_dev *pdev) ^~~~~~~~~~~~~~~ In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: include/linux/pci.h:2317:13: note: previous definition of 'pci_pr3_present' was here static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ drivers/pci/pci.c: In function 'pci_pr3_present': >> drivers/pci/pci.c:5870:13: error: dereferencing pointer to incomplete type 'struct acpi_device' return adev->power.flags.power_resources && ^~ >> drivers/pci/pci.c:5871:3: error: implicit declaration of function 'acpi_has_method'; did you mean 'acpi_has_watchdog'? [-Werror=implicit-function-declaration] acpi_has_method(adev->handle, "_PR3"); ^~~~~~~~~~~~~~~ acpi_has_watchdog In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: At top level: include/linux/pci.h:2317:13: warning: 'pci_pr3_present' defined but not used [-Wunused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pci_pr3_present +5859 drivers/pci/pci.c 5858 > 5859 bool pci_pr3_present(struct pci_dev *pdev) 5860 { 5861 struct acpi_device *adev; 5862 5863 if (acpi_disabled) 5864 return false; 5865 5866 adev = ACPI_COMPANION(&pdev->dev); 5867 if (!adev) 5868 return false; 5869 > 5870 return adev->power.flags.power_resources && > 5871 acpi_has_method(adev->handle, "_PR3"); 5872 } 5873 EXPORT_SYMBOL_GPL(pci_pr3_present); 5874 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Enable headset mic on Asus MJ401TA
by Daniel Drake 17 Oct '19

17 Oct '19

On Asus MJ401TA (with Realtek ALC256), the headset mic is connected to pin 0x19, with default configuration value 0x411111f0 (indicating no physical connection). Enable this by quirking the pin. Mic jack detection was also tested and found to be working. This enables use of the headset mic on this product. Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..e9fc5d8ff497 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5869,6 +5869,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, @@ -6912,6 +6913,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7108,6 +7118,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.20.1

2 1

[alsa-devel] [PATCH v5 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 17 Oct '19

17 Oct '19

Hey, here's the 5th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v5 changes: - Fixed a bug with codec power management at probe. To imitate snd_hda_intel driver, SOF needs to enable display power when doing codec probe for HDMI codecs. Affects patches 2 and 5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 110 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 93 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 337 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

2 11

[alsa-devel] [PATCH 00/17] firewire: share the size of period for both playback and capture PCM substreams
by Takashi Sakamoto 17 Oct '19

17 Oct '19

Hi, This patchset is a preparation for enhancement of AMDTP domain that I addressed to my previous patchset: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/152430.html When queued packets for several IT/IR contexts in the same domain are handled in hardware IRQ context for an IT context, the number of events in the packets are mostly the same for each of contexts. This means that the size of period for PCM substreams associated to the contexts is also the same. At present, PCM substreams for the contexts can have own size of period. This commit adds a member into AMDTP domain to share the size of period for PCM substreams on AMDTP streams in the same domain. This patchset includes optimization to detect whether isochronous resources for AMDTP streams are reserved or not in pcm.open callback. Takashi Sakamoto (17): firewire-lib: add a member into AMDTP domain for events per period bebob: register the size of PCM period to AMDTP domain fireworks: register the size of PCM period to AMDTP domain oxfw: register the size of PCM period to AMDTP domain dice: register the size of PCM period to AMDTP domain firewire-digi00x: register the size of PCM period to AMDTP domain firewire-tascam: register the size of PCM period to AMDTP domain firewire-motu: register the size of PCM period to AMDTP domain fireface: register the size of PCM period to AMDTP domain bebob: use the same size of period for PCM substream in AMDTP streams fireworks: use the same size of period for PCM substream in AMDTP streams oxfw: use the same size of period for PCM substream in AMDTP streams dice: use the same size of period for PCM substream in AMDTP streams firewire-digi00x: use the same size of period for PCM substream in AMDTP streams firewire-tascam: use the same size of period for PCM substream in AMDTP streams firewire-motu: use the same size of period for PCM substream in AMDTP streams fireface: use the same size of period for PCM substreams in AMDTP streams sound/firewire/amdtp-stream.c | 4 ++ sound/firewire/amdtp-stream.h | 10 +++++ sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 43 ++++++++++++------ sound/firewire/bebob/bebob_stream.c | 11 ++++- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 50 +++++++++++++++------ sound/firewire/dice/dice-stream.c | 8 +++- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 39 ++++++++++++---- sound/firewire/digi00x/digi00x-stream.c | 11 ++++- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 32 ++++++++++--- sound/firewire/fireface/ff-stream.c | 11 ++++- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 43 +++++++++++++----- sound/firewire/fireworks/fireworks_stream.c | 11 ++++- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 33 ++++++++++---- sound/firewire/motu/motu-stream.c | 11 ++++- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 45 +++++++++++++------ sound/firewire/oxfw/oxfw-stream.c | 12 ++++- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 38 ++++++++++++---- sound/firewire/tascam/tascam-stream.c | 11 ++++- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 354 insertions(+), 107 deletions(-) -- 2.20.1

2 18

[alsa-devel] [PATCH v4 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 17 Oct '19

17 Oct '19

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: - Let caller to find its upstream port device. drivers/pci/pci.c | 16 ++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 18 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..d03f624d8928 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,22 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

3 5

[alsa-devel] [PATCH v5.3] ASoC: sgtl5000: add ADC mute control
by Richard Leitner 17 Oct '19

17 Oct '19

From: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Upstream commit 631bc8f0134a ("ASoC: sgtl5000: Fix of unmute outputs on probe"), which is e9f621efaebd in v5.3 replaced snd_soc_component_write with snd_soc_component_update_bits and therefore no longer cleared the MUTE_ADC flag. This caused the ADC to stay muted and recording doesn't work any longer. This patch fixes this problem by adding a Switch control for MUTE_ADC. commit 694b14554d75 ("ASoC: sgtl5000: add ADC mute control") upstream This control mute/unmute the ADC input of SGTL5000 using its CHIP_ANA_CTRL register. Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Reviewed-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Reviewed-by: Igor Opaniuk <igor.opaniuk(a)toradex.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20190719100524.23300-5-oleksandr.suvorov@toradex.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> Fixes: e9f621efaebd ("ASoC: sgtl5000: Fix of unmute outputs on probe") --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3f28e7862b5b..b65232521ea8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -720,6 +720,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SGTL5000_CHIP_ANA_ADC_CTRL, 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + SOC_SINGLE("Capture Switch", SGTL5000_CHIP_ANA_CTRL, 0, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", SGTL5000_CHIP_ANA_HP_CTRL, -- 2.21.0

2 1

[alsa-devel] [PATCH v6 0/7] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 16 Oct '19

16 Oct '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, To be able to add support for the Allwinner H6 I've changed some of the original reg fields into function calls as this made it easier to setup for multi-channel audio especially across different SoCs. I've also stripped out all the other patches unrelated to this which I will deliver after support for the H6 has gone in. These other patches are required for HDMI audio which is driving this patchset and they can be found here https://github.com/codekipper/linux-sunxi/commits/upstream-i2s BR, CK --- v6 changes compared to v5 are: - rebased onto the recent tdm delivery - stripped out patches not required for the H6 delivery v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Jernej Skrabec (2): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S Marcus Cooper (5): ASoC: sun4i-i2s: Move channel select offset ASoC: sun4i-i2s: Add functions for RX and TX channel offsets ASoC: sun4i-i2s: Add functions for RX and TX channel enables ASoC: sun4i-i2s: Add functions for RX and TX channel selects ASoC: sun4i-i2s: Add functions for RX and TX channel mapping .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + sound/soc/sunxi/sun4i-i2s.c | 337 ++++++++++++++++-- 2 files changed, 305 insertions(+), 34 deletions(-) -- 2.23.0

3 13

[alsa-devel] [PATCH v2] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 142 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++ 2 files changed, 160 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..b927e61b32df 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1829,6 +1829,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; } switch (slot_width) { @@ -1848,6 +1849,135 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + switch (first_bit) { + case 1: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + break; + case 3: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + break; + case 5: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + break; + case 7: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + break; + case 0: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, 0); + break; + case 2: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, 0); + break; + case 4: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, 0); + break; + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, 0); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location error\n"); + goto _set_tdm_err_; + } + } else if (tx_slotnum == 2) { + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1860,10 +1990,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: rt1011: add r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt1011.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt index 35a23e60d679..02d53b9aa247 100644 --- a/Documentation/devicetree/bindings/sound/rt1011.txt +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -20,6 +20,14 @@ Required properties: | 1 | 1 | 0x3b | ------------------------------------- +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + Pins on the device (for linking into audio routes) for RT1011: * SPO @@ -29,4 +37,6 @@ Example: rt1011: codec@38 { compatible = "realtek,rt1011"; reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; }; -- 2.23.0

2 3

[alsa-devel] [PATCH] ASoC: wm8958: use <asm/unaligned.h> to simplify code
by Ben Dooks (Codethink) 16 Oct '19

16 Oct '19

Simplify the memcpy/be32_to_cpu() code by simply using get_unaligned_be32() throughout and makes the code nicer to look at. This fixes the following warnings from sparse: sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: patches(a)opensource.cirrus.com Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wm8958-dsp2.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 18535b326680..ca42445b649d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -25,6 +25,8 @@ #include <linux/mfd/wm8994/pdata.h> #include <linux/mfd/wm8994/gpio.h> +#include <asm/unaligned.h> + #include "wm8994.h" #define WM_FW_BLOCK_INFO 0xff @@ -58,18 +60,15 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (memcmp(fw->data, "WMFW", 4) != 0) { - memcpy(&data32, fw->data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(fw->data); dev_err(component->dev, "%s: firmware has bad file magic %08x\n", name, data32); goto err; } - memcpy(&data32, fw->data + 4, sizeof(data32)); - len = be32_to_cpu(data32); + len = get_unaligned_be32(fw->data + 4); + data32 = get_unaligned_be32(fw->data + 8); - memcpy(&data32, fw->data + 8, sizeof(data32)); - data32 = be32_to_cpu(data32); if ((data32 >> 24) & 0xff) { dev_err(component->dev, "%s: unsupported firmware version %d\n", name, (data32 >> 24) & 0xff); @@ -87,9 +86,8 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (check) { - memcpy(&data64, fw->data + 24, sizeof(u64)); - dev_info(component->dev, "%s timestamp %llx\n", - name, be64_to_cpu(data64)); + data64 = get_unaligned_be64(fw->data + 24); + dev_info(component->dev, "%s timestamp %llx\n", name, data64); } else { snd_soc_component_write(component, 0x102, 0x2); snd_soc_component_write(component, 0x900, 0x2); @@ -104,8 +102,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data + 4, sizeof(data32)); - block_len = be32_to_cpu(data32); + block_len = get_unaligned_be32(data + 4); if (block_len + 8 > len) { dev_err(component->dev, "%zd byte block longer than file\n", block_len); @@ -116,8 +113,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(data); switch ((data32 >> 24) & 0xff) { case WM_FW_BLOCK_INFO: -- 2.23.0

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[alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. In Chromium OS case, the coreboot will read the info from VPD and create the device property for each rt1011. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 ++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index b927e61b32df..ad049cfddcb0 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2328,8 +2328,12 @@ static void rt1011_calibration_work(struct work_struct *work) struct rt1011_priv *rt1011 = container_of(work, struct rt1011_priv, cali_work); struct snd_soc_component *component = rt1011->component; + unsigned int r0_integer, r0_factor, format; - rt1011_calibrate(rt1011, 1); + if (rt1011->r0_calib) + rt1011_calibrate(rt1011, 0); + else + rt1011_calibrate(rt1011, 1); /* * This flag should reset after booting. @@ -2340,6 +2344,39 @@ static void rt1011_calibration_work(struct work_struct *work) /* initial */ rt1011_reg_init(component); + + /* Apply temperature and calibration data from device property */ + if (rt1011->temperature_calib) { + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (rt1011->temperature_calib << 2)); + } + + if (rt1011->r0_calib) { + rt1011->r0_reg = rt1011->r0_calib; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + rt1011_r0_load(rt1011); + } +} + +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) +{ + device_property_read_u32(dev, "realtek,temperature_calib", + &rt1011->temperature_calib); + device_property_read_u32(dev, "realtek,r0_calib", + &rt1011->r0_calib); + + dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + __func__, rt1011->r0_calib, rt1011->temperature_calib); + + return 0; } static int rt1011_i2c_probe(struct i2c_client *i2c, @@ -2356,6 +2393,8 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1011); + rt1011_parse_dp(rt1011, &i2c->dev); + rt1011->regmap = devm_regmap_init_i2c(i2c, &rt1011_regmap); if (IS_ERR(rt1011->regmap)) { ret = PTR_ERR(rt1011->regmap); diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 3f7dea1191ad..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -690,6 +690,7 @@ struct rt1011_priv { int bq_drc_set; unsigned int r0_reg, cali_done; + unsigned int r0_calib, temperature_calib; int recv_spk_mode; }; -- 2.23.0

3 3

[alsa-devel] [PATCH] ASoC: sof: include types.h at header.h
by Kuninori Morimoto 16 Oct '19

16 Oct '19

Content-Transfer-Encoding: 8bit Without <types.h> we will get these error linux/include/sound/sof/header.h:125:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:136:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:137:2: error: unknown type name ‘uint32_t’uint32_t cmd; ... linux/include/sound/sof/dai-imx.h:18:2: error: unknown type name ‘uint16_t’uint16_t reserved1; linux/include/sound/sof/dai-imx.h:30:2: error: unknown type name ‘uint16_t’uint16_t tdm_slot_width; linux/include/sound/sof/dai-imx.h:31:2: error: unknown type name ‘uint16_t’uint16_t reserved2; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/sof/header.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c0..332143f 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -9,6 +9,7 @@ #ifndef __INCLUDE_SOUND_SOF_HEADER_H__ #define __INCLUDE_SOUND_SOF_HEADER_H__ +#include <linux/types.h> #include <uapi/sound/sof/abi.h> /** \addtogroup sof_uapi uAPI -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Mark Brown 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 11:28:47AM +0000, Shuming [范書銘] wrote: > > These are new DT bindings and need to be added to the binding document. > Sure, could I send another patch for it? Yes, the DT bindings should go in a separate patch anyway.

1 0

Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Cheng-yi Chiang 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 7:29 PM Shuming [范書銘] <shumingf(a)realtek.com> wrote: > > > On Wed, Oct 16, 2019 at 04:58:45PM +0800, shumingf(a)realtek.com wrote: > > > > > +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device > > > +*dev) { > > > + device_property_read_u32(dev, "realtek,temperature_calib", > > > + &rt1011->temperature_calib); > > > + device_property_read_u32(dev, "realtek,r0_calib", > > > + &rt1011->r0_calib); > > > > These are new DT bindings and need to be added to the binding document. > > Sure, could I send another patch for it? > > > ------Please consider the environment before printing this e-mail. Hi Shuming, Yes, you will need a patch before this one to add the binding documentation. Thanks!

1 0

[alsa-devel] [PATCH 0/2] soundwire: Kconfig/build improvements
by Pierre-Louis Bossart 16 Oct '19

16 Oct '19

The Intel kbuild test reported issues with COMPILE_TEST or cross-compilation when SOF is enabled, fix. This has no functional impact. Pierre-Louis Bossart (2): soundwire: intel: add missing headers for cross-compilation soundwire: intel: remove X86 dependency drivers/soundwire/Kconfig | 2 +- drivers/soundwire/intel.c | 1 + drivers/soundwire/intel_init.c | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 16 Oct '19

16 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. change the name of mtk_wdt_compatible to mtk_wdt_data 3. remove toprgu_reset struct and use mtk_wdt_dev instead 4. get the value of sw_rst_num from .h file 5. adddd mt2712-resets.h for mt2712 6. improve commit message v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH] ASoC: pcm3168a: Use fixup instead of constraint for channels and formats
by Peter Ujfalusi 16 Oct '19

16 Oct '19

The snd_pcm_hw_constraint_minmax() works fine when a single codec is connected to a single CPU DAI, but in multicodec or DPCM setup the constraints placed by the driver will apply to the whole PCM stream (FE included) and thus prevents more than 8 playback channels for example. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have a board which uses two pcm3168a codec connected to one McASP port with parallel serializers. One 16 channel (2x 8 channel) playback and one 12 (2x 6 channel) capture should be possible as I use only one CPU DAI + platform. However no matter how I tried to set things up ALSA blocked more than 8 channel playback in simple multicodec or DPCM setup. It turned out that the constraint set by the codec on the BE is actually applies to the FE as well as they both use the same runtime. The codec still have different constraints depending on the interface format, so I replaced the constraint with a fixup function to make sure that the codec will not advertise non supported formats. Is this a know feature? Another thing which I have came across is the lack of support for parallel data pins in ASoC core. For example the soc-pcm only checks the tdm slots to fix up the channels supported by the codec, but when we have multiple pins in use and they are operate in parallel, then the number of slots must be multiplied by the data pins in use. I have wip patches for this, adding int snd_soc_dai_set_data_pins(struct snd_soc_dai *dai, unsigned int tx_data_pins, unsigned int rx_data_pins); and then using the information in soc-pcm, but need to check further to see if there are other places this might be needed. Regards, Peter sound/soc/codecs/pcm3168a.c | 98 +++++++++++++++---------------------- 1 file changed, 40 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 88b75695fbf7..708dac27feff 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -62,6 +62,7 @@ struct pcm3168a_priv { unsigned long sysclk; struct pcm3168a_io_params io_params[2]; + struct snd_soc_dai_driver dai_drv[2]; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -314,6 +315,37 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } +static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; + unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; + + if (!pcm3168a->io_params[dai->id].fmt) + return; + + if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { + /* S16_LE is only supported in RIGHT_J mode */ + formats |= SNDRV_PCM_FMTBIT_S16_LE; + + /* + * If multi DIN/DOUT is not selected, RIGHT_J can only support + * two channels (no TDM support) + */ + if (pcm3168a->io_params[dai->id].tdm_slots != 2) + channel_max = 2; + } + + if (dai->id == PCM3168A_DAI_DAC) { + dai->driver->playback.channels_max = channel_max; + dai->driver->playback.formats = formats; + } else { + dai->driver->capture.channels_max = channel_max; + dai->driver->capture.formats = formats; + } +} + static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; @@ -376,6 +408,8 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift); + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -409,6 +443,8 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, else io_params->tdm_mask = rx_mask; + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -530,63 +566,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pcm3168a_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - unsigned int sample_min; - unsigned int channel_max; - unsigned int channel_maxs[] = { - 8, /* DAC */ - 6 /* ADC */ - }; - - /* - * Available Data Bits - * - * RIGHT_J : 24 / 16 - * LEFT_J : 24 - * I2S : 24 - * - * TDM available - * - * I2S - * LEFT_J - */ - switch (pcm3168a->io_params[dai->id].fmt) { - case PCM3168A_FMT_RIGHT_J: - sample_min = 16; - channel_max = 2; - break; - case PCM3168A_FMT_LEFT_J: - case PCM3168A_FMT_I2S: - case PCM3168A_FMT_DSP_A: - case PCM3168A_FMT_DSP_B: - sample_min = 24; - channel_max = channel_maxs[dai->id]; - break; - default: - sample_min = 24; - channel_max = 2; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - sample_min, 32); - - /* Allow all channels in multi DIN/DOUT mode */ - if (pcm3168a->io_params[dai->id].tdm_slots == 2) - channel_max = channel_maxs[dai->id]; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, channel_max); - - return 0; -} static const struct snd_soc_dai_ops pcm3168a_dai_ops = { - .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, @@ -776,8 +756,10 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) pm_runtime_enable(dev); pm_runtime_idle(dev); - ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, pcm3168a_dais, - ARRAY_SIZE(pcm3168a_dais)); + memcpy(pcm3168a->dai_drv, pcm3168a_dais, sizeof(pcm3168a->dai_drv)); + ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, + pcm3168a->dai_drv, + ARRAY_SIZE(pcm3168a->dai_drv)); if (ret) { dev_err(dev, "failed to register component: %d\n", ret); goto err_regulator; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface (Re: Alsa-devel Digest, Vol 152, Issue 150)
by Hin-Tak Leung 15 Oct '19

15 Oct '19

> Date: Tue, 15 Oct 2019 00:10:28 +1030 > From: "Geoffrey D. Bennett" <g(a)b4.vu> > To: wwp <subscript(a)free.fr> > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio > interface > Message-ID: <20191014134028.GA11018(a)b4.vu> > Content-Type: text/plain; charset=us-ascii > Hi wwp, > On Sun, Oct 13, 2019 at 10:20:30AM +0200, wwp wrote: > > Hello, > > > > I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, > > great device, but in CentOS 7, with either the available 3.x or 4.x > > kernels, > ... > > Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't > > work. I presume there's more work like the patch submitted here (I saw > > the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 > > mixer interface" mail thread? Have you tried treating it as a 18i8 2nd Gen device first, at least? FWIW, I had adapted Geoffrey's work as a DKMS module: https://github.com/HinTak/sound-usb-dkms But you'll still need to be running as recent as a 5.2.x release kernel to use it. It just makes it easier to play with the code without recompiling the whole kernel. HTH. 3.x and 4.x is definitely way too old!

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[alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface
by wwp 15 Oct '19

15 Oct '19

Hello, I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, great device, but in CentOS 7, with either the available 3.x or 4.x kernels, the device just appears in the available Alsa devices, but no mixer is available, the device direct hardware monitor output is also not working (it is, by default, and works well in Windows, but I presume that w/ no mixer support in Alsa, it's expected that nothing works). I'm able to bound I/O ports in QJackCtl but nothing goes out of the device. Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't work. I presume there's more work like the patch submitted here (I saw the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 mixer interface" mail thread? Do you it is worth trying applying that 2nd Gen patch to one of the v4 kernel available for CentOS 7, adding the 3rd Gen quirks and see how it behaves? Or is there already progress on supporting the 3rd Gen one too? Regards, -- wwp https://useplaintext.email/

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[alsa-devel] [PATCH] soundwire: slave: Fix unused function warning on !ACPI
by Michal Suchanek 15 Oct '19

15 Oct '19

Fixes the following warning on !ACPI systems: drivers/soundwire/slave.c:16:12: warning: ‘sdw_slave_add’ defined but not used [-Wunused-function] static int sdw_slave_add(struct sdw_bus *bus, ^~~~~~~~~~~~~ Signed-off-by: Michal Suchanek <msuchanek(a)suse.de> --- drivers/soundwire/slave.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index f39a5815e25d..34c7e65831d1 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -6,6 +6,7 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +#if IS_ENABLED(CONFIG_ACPI) static void sdw_slave_release(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); @@ -60,7 +61,6 @@ static int sdw_slave_add(struct sdw_bus *bus, return ret; } -#if IS_ENABLED(CONFIG_ACPI) /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance -- 2.22.0

6 15

[alsa-devel] [PATCH] soundwire: qcom: fix devm_platform_ioremap_resource.cocci warnings
by Julia Lawall 15 Oct '19

15 Oct '19

From: kbuild test robot <lkp(a)intel.com> Use devm_platform_ioremap_resource helper which wraps platform_get_resource() and devm_ioremap_resource() together. Generated by: scripts/coccinelle/api/devm_platform_ioremap_resource.cocci Fixes: 4a789cff1848 ("soundwire: qcom: add support for SoundWire controller") CC: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- url: https://github.com/0day-ci/linux/commits/Srinivas-Kandagatla/soundwire-Add-… :::::: branch date: 9 hours ago :::::: commit date: 9 hours ago qcom.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -784,7 +784,6 @@ static int qcom_swrm_probe(struct platfo struct sdw_master_prop *prop; struct sdw_bus_params *params; struct qcom_swrm_ctrl *ctrl; - struct resource *res; int ret; u32 val; @@ -801,8 +800,7 @@ static int qcom_swrm_probe(struct platfo } else { ctrl->reg_read = qcom_swrm_mmio_reg_read; ctrl->reg_write = qcom_swrm_mmio_reg_write; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - ctrl->base = devm_ioremap_resource(dev, res); + ctrl->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(ctrl->base)) return PTR_ERR(ctrl->base); }

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 15 Oct '19

15 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 94 ++++++++++++++++++++++ 2 files changed, 95 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..5e0e7db 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -320,6 +320,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + RT5514 + MAX98927. diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..dc09a85 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,61 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +139,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +156,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +708,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +724,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: simple-card: fixup simple_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fc9c753..10b82bf 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -149,7 +149,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -179,7 +179,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: audio-graph: fixup graph_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 6007e63..9ad35d9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -232,7 +232,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -263,7 +263,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Remove one extra blank line also Regards, Peter sound/soc/codecs/pcm3168a.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..313500ab36df 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,9 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; - if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ formats |= SNDRV_PCM_FMTBIT_S16_LE; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..df8395b361d4 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,8 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH v3] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 15 Oct '19

15 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v2 to v3: - Remove spinlock around setting clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index a15fbf71b9d7..af060162b0af 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -2054,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); } /* not for DVI mode */ -- 2.23.0.581.g78d2f28ef7-goog

2 1

[alsa-devel] [RFC PATCH 0/2] Introduce for-dpcm DT property
by Daniel Baluta 15 Oct '19

15 Oct '19

We need to be able to create DPCM links even if we have a single CPU DAI or just a dummy CPU DAI. Daniel Baluta (2): ASoC: simple-card: Introduce force-dpcm DT property ASoC: simple-card: Add documentation for force-dpcm property .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card_utils.h | 4 +++ sound/soc/generic/simple-card-utils.c | 17 +++++++++++++ sound/soc/generic/simple-card.c | 25 +++++++++++++++++-- 4 files changed, 45 insertions(+), 2 deletions(-) -- 2.17.1

4 7

Re: [alsa-devel] [PATCH 2/2] ASoC: simple_card_utils.h: Fix potential multiple redefinition error
by Kuninori Morimoto 15 Oct '19

15 Oct '19

Hi > asoc_simple_debug_info and asoc_simple_debug_dai must be static > otherwise we might a compilation error if the compiler decides > not to inline the given function. > > Fixes: 0580dde59438686d ("ASoC: simple-card-utils: add asoc_simple_debug_info()") > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > --- Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 14 Oct '19

14 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++++++++++++++++++ 2 files changed, 101 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..d5f167e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -315,6 +315,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..7b6e70b 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,67 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(priv->mclk)) + return 0; + + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (!__clk_is_enabled(priv->mclk)) + return 0; + + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +145,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +162,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +714,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +730,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 7

[alsa-devel] [PATCH -next] ASoC: SOF: Fix randbuild error
by YueHaibing 14 Oct '19

14 Oct '19

When LEDS_TRIGGER_AUDIO is m and SND_SOC_SOF is y, sound/soc/sof/control.o: In function `snd_sof_switch_put': control.c:(.text+0x587): undefined reference to `ledtrig_audio_set' control.c:(.text+0x593): undefined reference to `ledtrig_audio_set' Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 41551e8..2c4abd4 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -36,10 +36,12 @@ static void update_mute_led(struct snd_sof_control *scontrol, scontrol->led_ctl.led_value = temp; +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) if (!scontrol->led_ctl.direction) ledtrig_audio_set(LED_AUDIO_MUTE, temp ? LED_OFF : LED_ON); else ledtrig_audio_set(LED_AUDIO_MICMUTE, temp ? LED_OFF : LED_ON); +#endif } static inline u32 mixer_to_ipc(unsigned int value, u32 *volume_map, int size) -- 2.7.4

3 2

Re: [alsa-devel] linux-next: Tree for Oct 14 (sound/soc/sof/)
by Randy Dunlap 14 Oct '19

14 Oct '19

On 10/13/19 11:47 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191011: > on i386: ld: sound/soc/sof/control.o: in function `snd_sof_switch_put': control.c:(.text+0x49a): undefined reference to `ledtrig_audio_set' ld: control.c:(.text+0x4d1): undefined reference to `ledtrig_audio_set' when CONFIG_LEDS_TRIGGER_AUDIO=m CONFIG_SND_SOC_SOF=y This code in <linux/leds.h> does not handle the config combo above: #if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) enum led_brightness ledtrig_audio_get(enum led_audio type); void ledtrig_audio_set(enum led_audio type, enum led_brightness state); #else static inline enum led_brightness ledtrig_audio_get(enum led_audio type) { return LED_OFF; } static inline void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { } #endif -- ~Randy

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - edited 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was edited from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: `#inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - opened 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was opened from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: ` #inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH v8 0/3] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 14 Oct '19

14 Oct '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v7: - Added support to dai based on Discussion on mailing list. - Moved readable array to proper switch statement. - Remove unnecessary locks which can be added once tsens is supported. - Suffix gain to Volume as suggested by Mark. Srinivas Kandagatla (3): dt-bindings: ASoC: Add WSA881x bindings dt-bindings: ASoC: WSA881x: Add missing #sound-dai-cells ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1153 +++++++++++++++++ 4 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 4

[alsa-devel] [PATCH] ASoC: stm32: spdifrx: retry synchronization in sync state
by Olivier Moysan 14 Oct '19

14 Oct '19

When STM32 SPDIFRX is in sync state, allow multiple synchro attempts, instead of exiting on first unsuccessful trial. This is useful when spdif signal is not immediately available on input. This also allows Pulseaudio to check iec capture device availability when no signal is present. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_spdifrx.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index cd4b235fce57..3fd28ee01675 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -351,6 +351,8 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) SPDIFRX_CR_CUMSK | SPDIFRX_CR_PTMSK | SPDIFRX_CR_RXSTEO; cr_mask = cr; + cr |= SPDIFRX_CR_NBTRSET(SPDIFRX_NBTR_63); + cr_mask |= SPDIFRX_CR_NBTR_MASK; cr |= SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); cr_mask |= SPDIFRX_CR_SPDIFEN_MASK; ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, @@ -666,7 +668,7 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) struct snd_pcm_substream *substream = spdifrx->substream; struct platform_device *pdev = spdifrx->pdev; unsigned int cr, mask, sr, imr; - unsigned int flags; + unsigned int flags, sync_state; int err = 0, err_xrun = 0; regmap_read(spdifrx->regmap, STM32_SPDIFRX_SR, &sr); @@ -726,11 +728,23 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) } if (err) { - /* SPDIFRX in STATE_STOP. Disable SPDIFRX to clear errors */ + regmap_read(spdifrx->regmap, STM32_SPDIFRX_CR, &cr); + sync_state = FIELD_GET(SPDIFRX_CR_SPDIFEN_MASK, cr) && + SPDIFRX_SPDIFEN_SYNC; + + /* SPDIFRX is in STATE_STOP. Disable SPDIFRX to clear errors */ cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_DISABLE); regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, SPDIFRX_CR_SPDIFEN_MASK, cr); + /* If SPDIFRX was in STATE_SYNC, retry synchro */ + if (sync_state) { + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_SPDIFEN_MASK, cr); + return IRQ_HANDLED; + } + if (substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); -- 2.17.1

2 1

[alsa-devel] [PATCH -next] ASoC: adau7118: Fix Kconfig warning without CONFIG_I2C
by YueHaibing 14 Oct '19

14 Oct '19

When building a kernel without CONFIG_I2C, Kconfig warns: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [y]: - SND_SOC_ADAU7118_I2C [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] Add missing I2C dependency to SND_SOC_ADAU7118_I2C to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ca514c0f12b0 ("ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d196752..6a28741 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -416,6 +416,7 @@ config SND_SOC_ADAU7118_HW config SND_SOC_ADAU7118_I2C tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + depends on I2C select SND_SOC_ADAU7118 select REGMAP_I2C help -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: fix old-style function declaration
by YueHaibing 14 Oct '19

14 Oct '19

gcc warn about this: sound/soc/fsl/fsl_mqs.c:146:1: warning: static is not at beginning of declaration [-Wold-style-declaration] Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index f7fc44e..0c813a4 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -143,7 +143,7 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, MQS_EN_MASK, 0); } -const static struct snd_soc_component_driver soc_codec_fsl_mqs = { +static const struct snd_soc_component_driver soc_codec_fsl_mqs = { .idle_bias_on = 1, .non_legacy_dai_naming = 1, }; -- 2.7.4

4 5

[alsa-devel] [PATCH 0/4] ASoC: SOF: fix static analysis warnings
by Pierre-Louis Bossart 14 Oct '19

14 Oct '19

Recent SOF commits generate warnings reported e.g. by cppcheck. Unfortunately those tools are too verbose to be used by our CI, so there's always a delay in detecting such issues. Pierre-Louis Bossart (4): ASoC: SOF: Intel: byt: fix operator precedence warnings ASoC: SOF: Intel: bdw: fix operator precedence warnings ASoC: SOF: topology: remove always-true redundant test ASoC: SOF: topology: check errors when parsing LED tokens sound/soc/sof/intel/bdw.c | 16 ++++++++-------- sound/soc/sof/intel/byt.c | 16 ++++++++-------- sound/soc/sof/topology.c | 12 ++++++++---- 3 files changed, 24 insertions(+), 20 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH v2 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 14 Oct '19

14 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

4 18

[alsa-devel] [PATCH] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 14 Oct '19

14 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/firmware/google/vpd.c | 16 ++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 ++++++++++++++++++ 2 files changed, 34 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..71e9d2da63be 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,22 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kstrndup(info->value, value_len, GFP_KERNEL); + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL(vpd_attribute_read_value); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..6f1160f28af8 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

6 10

[alsa-devel] [PATCH] ASoC: rt1011: export r0 and temperature config
by Shuming [范書銘] 14 Oct '19

14 Oct '19

In chromebook case, the machine driver will get the r0 calibration data and temperature from VPD. Therefore, the codec exports r0 and temperature config API for it. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1011.h | 5 +++++ 2 files changed, 46 insertions(+) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..a992bb2224b2 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1325,6 +1325,47 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0) +{ + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + struct device *dev; + unsigned int r0_integer, r0_factor, format; + + if (!component->card->instantiated) + return 0; + + if (!r0) + return -EINVAL; + + dev = regmap_get_device(rt1011->regmap); + rt1011->r0_reg = r0; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_dbg(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + if (rt1011->r0_reg) + rt1011_r0_load(rt1011); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_r0_config); + +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp) +{ + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (temp << 2)); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_temperature_config); + static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ SOC_ENUM("DIN Source", rt1011_din_source_enum), diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..b99df334be14 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -670,4 +670,9 @@ struct rt1011_priv { int recv_spk_mode; }; +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0); +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp); + + #endif /* end of _RT1011_H_ */ -- 2.23.0

4 9

[alsa-devel] [PATCH RESEND] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by Shuming [范書銘] 14 Oct '19

14 Oct '19

The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 97 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++++++ 2 files changed, 115 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..a36144de11cf 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1850,6 +1850,91 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + if (rx_slotnum) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) + << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } else { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) + << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + if (first_bit == 1) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + else if (first_bit == 3) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + else if (first_bit == 5) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + else if (first_bit == 7) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + } else if (tx_slotnum == 2) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + } else { + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } + } + snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); @@ -1860,10 +1945,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

2 2

Re: [alsa-devel] [PATCH -next] ASoC: atmel: select SND_ATMEL_SOC_DMA for SND_ATMEL_SOC_SSC
by Alexandre Belloni 12 Oct '19

12 Oct '19

On 12/10/2019 10:42:30+0800, Mao Wenan wrote: > If SND_ATMEL_SOC_SSC_PDC=y and SND_ATMEL_SOC_SSC_DMA=m, > below errors can be found: > sound/soc/atmel/atmel_ssc_dai.o: In function > `atmel_ssc_set_audio': > atmel_ssc_dai.c:(.text+0x6fe): undefined reference to > `atmel_pcm_dma_platform_register' > make: *** [vmlinux] Error 1 > > After commit 18291410557f ("ASoC: atmel: enable > SOC_SSC_PDC and SOC_SSC_DMA in Kconfig"), SND_ATMEL_SOC_DMA > and SND_ATMEL_SOC_SSC are selected by SND_ATMEL_SOC_SSC_DMA, > SND_ATMEL_SOC_SSC is also selected by SND_ATMEL_SOC_SSC_PDC, > the results are SND_ATMEL_SOC_DMA=m but SND_ATMEL_SOC_SSC=y, > so the errors happen. > > This patch make SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA. > > Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") > Signed-off-by: Mao Wenan <maowenan(a)huawei.com> > --- > sound/soc/atmel/Kconfig | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig > index f118c22..2938f6b 100644 > --- a/sound/soc/atmel/Kconfig > +++ b/sound/soc/atmel/Kconfig > @@ -19,6 +19,7 @@ config SND_ATMEL_SOC_DMA > > config SND_ATMEL_SOC_SSC > tristate > + select SND_ATMEL_SOC_DMA > This is not the solution because this doesn't allow to compile out DMA and use only PDC. I think Arnd already submitted a proper patch. > config SND_ATMEL_SOC_SSC_PDC > tristate "SoC PCM DAI support for AT91 SSC controller using PDC" > -- > 2.7.4 > -- Alexandre Belloni, Bootlin Embedded Linux and Kernel engineering https://bootlin.com

1 0

[alsa-devel] [PATCH v3 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 11 Oct '19

11 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Changes in v2: * Use C++ coment style on the entire comment in the .c files; * Do not use ternery operator; * Do not use `devm_regulator_get_optional` as the device always requires power; * Let DAPM suspend/resume the device. Changes in v3: * no changes. MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 586 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 770 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index 55199ef7fa74..bf2c71f90eb6 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1002,6 +1002,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ae213c6392ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -396,6 +398,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73b2d5982dcb..2118407a1dd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -305,6 +308,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..45a5d2dcc0f2 --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw +// driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..a8211362fe82 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..bf5a5d75f81a --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,586 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + st->iovdd = devm_regulator_get(st->dev, "IOVDD"); + if (IS_ERR(st->iovdd)) { + dev_err(st->dev, "Could not get iovdd: %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, + st); +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] [PATCH 00/21] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 11 Oct '19

11 Oct '19

Hi Mark Cc Pierre-Louis These are step4 of soc-core cleanup. These related to soc-topology. If my understanding and my code are correct, there is no effect to topology side, but I can't test. I'm not sure who can test these, but I guess Pierre-Louis knows ? Pierre-Louis, can you help it ? Kuninori Morimoto (21): ASoC: soc-core: remove for_each_rtdcom_safe() ASoC: soc-core: add for_each_rtd_components() and replace ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: rename soc_init_dai_link() to soc_dai_link_sanity_check() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: snd_soc_unbind_card() cleanup ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: use mutex_lock() at snd_soc_add_component() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 25 ++- sound/soc/soc-component.c | 43 +--- sound/soc/soc-compress.c | 52 ++--- sound/soc/soc-core.c | 542 ++++++++++++++++++++-------------------------- sound/soc/soc-pcm.c | 49 ++--- sound/soc/soc-topology.c | 32 ++- 6 files changed, 320 insertions(+), 423 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v2 0/5] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 11 Oct '19

11 Oct '19

Thanks for reviewing the RFC patchset. Here is new patchset addressing all the comments from RFC. This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is on DragonBoard DB845c based of SDM845 SoC. WCD934x and WSA881x patches will be posted soon. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since RFC: - updated bindings as suggested to take care of more bus parameters. - fixed error code of snd_soc_dai_get_sdw_stream() dummy function. - Cleaned up driver to handle read/writes in same way without special casing. - removed unused defines Srinivas Kandagatla (4): soundwire: stream: make stream name a const pointer ASoC: core: add support to snd_soc_dai_get_sdw_stream() dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller Vinod Koul (1): soundwire: Add compute_params callback .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 919 ++++++++++++++++++ drivers/soundwire/stream.c | 12 +- include/linux/soundwire/sdw.h | 6 +- include/sound/soc-dai.h | 10 + 7 files changed, 1124 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

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[alsa-devel] [PATCH 20/36] ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS
by Arnd Bergmann 11 Oct '19

11 Oct '19

The constant requires indirectly including a machine header file, but it's not actually used any more since commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so remove it completely. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 2 +- 3 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 7e196b599be1..6a254fdaa6fb 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -616,8 +616,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) + struct s3c_i2sv2_info *i2s) { struct device *dev = dai->dev; unsigned int iismod; diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index fe42b77999fd..8c6fc0d3d77e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -83,8 +83,7 @@ extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, * @base: The base address for the registers. */ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); + struct s3c_i2sv2_info *i2s); /** * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..5bf571762e89 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -49,7 +49,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, &s3c2412_i2s_pcm_stereo_in); - ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s); if (ret) return ret; -- 2.20.0

2 1

[alsa-devel] Counting links in simple-card
by Daniel Baluta 11 Oct '19

11 Oct '19

Hi Morimoto-san, In simple-card.c simple_get_dais_count function there is this comment: * ex1) * CPU0 --- Codec0 * CPU1 --- Codec1 * CPU2 -----/ * CPU3 --- Codec2 * * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec * => 7 DAIs = 4xCPU + 3xCodec * => 1 ccnf = 1xdummy-Codec So, I don't understand exactly how did you get the formula for links. AFAICT there should be 3 x CPU-Codec + 1 x CPU-dummy. thanks, Daniel.

2 7

[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 11 Oct '19

11 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added type and description and removed number of items from 'model' property. Best regards, Maciej Falkowski --- .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2-audio.yaml | 84 +++++++++++++++++++ 2 files changed, 84 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..62b5b9a1dec0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are + the WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + +required: + - compatible + - model + - audio-codec + - i2s-controller + - audio-amplifier + - samsung,audio-routing + - mic-bias-gpios + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

3 2

[alsa-devel] [PATCH 19/36] ARM: s3c: rx1950: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-rx1950.c | 16 +++++++ sound/soc/samsung/rx1950_uda1380.c | 72 +++++++++-------------------- 2 files changed, 39 insertions(+), 49 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 29f9b345a531..9f3b0f18cd36 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -724,6 +724,20 @@ static struct i2c_board_info rx1950_i2c_devices[] = { }, }; +static struct gpiod_lookup_table rx1950_audio_gpio_table = { + .dev_id = "rx1950-audio", + .table = { + GPIO_LOOKUP("GPIOG", 12, "hp-gpio", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOA", 1, "speaker-power", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device rx1950_audio = { + .name = "rx1950-audio", + .id = -1, +}; + static struct platform_device *rx1950_devices[] __initdata = { &s3c2410_device_dclk, &s3c_device_lcd, @@ -742,6 +756,7 @@ static struct platform_device *rx1950_devices[] __initdata = { &power_supply, &rx1950_battery, &rx1950_leds, + &rx1950_audio, }; static void __init rx1950_map_io(void) @@ -809,6 +824,7 @@ static void __init rx1950_init_machine(void) gpio_direction_output(S3C2410_GPJ(6), 0); pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); + gpiod_add_lookup_table(&rx1950_audio_gpio_table); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 4b247e91ae5b..cf96eaaf6b4e 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -12,16 +12,13 @@ // Vasily Khoruzhick <anarsoul(a)gmail.com> #include <linux/types.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> #include "regs-iis.h" -#include <asm/mach-types.h> - #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); @@ -58,7 +55,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { [0] = { - .gpio = S3C2410_GPG(12), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,8 +119,6 @@ static struct snd_soc_card rx1950_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *s3c24xx_snd_device; - static int rx1950_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -134,13 +128,15 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } +struct gpio_desc *gpiod_speaker_power; + static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C2410_GPA(1), 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C2410_GPA(1), 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -214,57 +210,35 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int __init rx1950_init(void) +static int rx1950_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_rx1950()) - return -ENODEV; + struct device *dev = &pdev->dev; /* configure some gpios */ - ret = gpio_request(S3C2410_GPA(1), "speaker-power"); - if (ret) - goto err_gpio; - - ret = gpio_direction_output(S3C2410_GPA(1), 0); - if (ret) - goto err_gpio_conf; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_plat_alloc; - } - - platform_set_drvdata(s3c24xx_snd_device, &rx1950_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) { - platform_device_put(s3c24xx_snd_device); - goto err_plat_add; + gpiod_speaker_power = devm_gpiod_get(dev, "speaker-power", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "cannot get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - return 0; - -err_plat_add: -err_plat_alloc: -err_gpio_conf: - gpio_free(S3C2410_GPA(1)); + hp_jack_gpios[0].gpiod_dev = dev; + rx1950_asoc.dev = dev; -err_gpio: - return ret; + return devm_snd_soc_register_card(dev, &rx1950_asoc); } -static void __exit rx1950_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C2410_GPA(1)); -} +struct platform_driver rx1950_audio = { + .driver = { + .name = "rx1950-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = rx1950_probe, +}; -module_init(rx1950_init); -module_exit(rx1950_exit); +module_platform_driver(rx1950_audio); /* Module information */ MODULE_AUTHOR("Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC RX1950"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:rx1950-audio"); -- 2.20.0

1 0

[alsa-devel] [PATCH 18/36] ARM: s3c: gta02-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-gta02.c | 17 ++++++ sound/soc/samsung/neo1973_wm8753.c | 85 +++++++++--------------------- 2 files changed, 43 insertions(+), 59 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 594901f3b8e5..526fd0933289 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -15,6 +15,7 @@ #include <linux/delay.h> #include <linux/timer.h> #include <linux/init.h> +#include <linux/gpio/machine.h> #include <linux/gpio.h> #include <linux/gpio_keys.h> #include <linux/workqueue.h> @@ -474,6 +475,20 @@ static struct platform_device gta02_buttons_device = { }, }; +static struct gpiod_lookup_table gta02_audio_gpio_table = { + .dev_id = "neo1973-audio", + .table = { + GPIO_LOOKUP("GPIOJ", 2, "amp-shut", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOJ", 1, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device gta02_audio = { + .name = "neo1973-audio", + .id = -1, +}; + static void __init gta02_map_io(void) { s3c24xx_init_io(gta02_iodesc, ARRAY_SIZE(gta02_iodesc)); @@ -498,6 +513,7 @@ static struct platform_device *gta02_devices[] __initdata = { &gta02_buttons_device, &s3c_device_adc, &s3c_device_ts, + &gta02_audio, }; static void gta02_poweroff(void) @@ -524,6 +540,7 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 38f536bafa09..b41e3dc297e7 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -11,14 +11,11 @@ #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> #include "regs-iis.h" - #include "../codecs/wm8753.h" #include "s3c24xx-i2s.h" @@ -166,6 +163,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; +static struct gpio_desc *gpiod_hp_in, *gpiod_amp_shut; static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -173,7 +171,7 @@ static int lm4853_set_spk(struct snd_kcontrol *kcontrol, { gta02_speaker_enabled = ucontrol->value.integer.value[0]; - gpio_set_value(S3C2410_GPJ(2), !gta02_speaker_enabled); + gpiod_set_value(gpiod_hp_in, !gta02_speaker_enabled); return 0; } @@ -188,7 +186,7 @@ static int lm4853_get_spk(struct snd_kcontrol *kcontrol, static int lm4853_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C2410_GPJ(1), SND_SOC_DAPM_EVENT_OFF(event)); + gpiod_set_value(gpiod_amp_shut, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -308,13 +306,8 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -static const struct gpio neo1973_gta02_gpios[] = { - { S3C2410_GPJ(2), GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, - { S3C2410_GPJ(1), GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, -}; - static struct snd_soc_card neo1973 = { - .name = "neo1973", + .name = "neo1973gta02", .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), @@ -332,62 +325,36 @@ static struct snd_soc_card neo1973 = { .fully_routed = true, }; -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) +static int neo1973_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_neo1973_gta02()) - return -ENODEV; + struct device *dev = &pdev->dev; - if (machine_is_neo1973_gta02()) { - neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 1; - - ret = gpio_request_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - if (ret) - return ret; + gpiod_hp_in = devm_gpiod_get(dev, "hp", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_hp_in)) { + dev_err(dev, "missing gpio %s\n", "hp"); + return PTR_ERR(gpiod_hp_in); } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) { - ret = -ENOMEM; - goto err_gpio_free; + gpiod_amp_shut = devm_gpiod_get(dev, "amp-shut", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_amp_shut)) { + dev_err(dev, "missing gpio %s\n", "amp-shut"); + return PTR_ERR(gpiod_amp_shut); } - platform_set_drvdata(neo1973_snd_device, &neo1973); - ret = platform_device_add(neo1973_snd_device); - - if (ret) - goto err_put_device; - - return 0; - -err_put_device: - platform_device_put(neo1973_snd_device); -err_gpio_free: - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } - return ret; + neo1973.dev = dev; + return devm_snd_soc_register_card(dev, &neo1973); } -module_init(neo1973_init); - -static void __exit neo1973_exit(void) -{ - platform_device_unregister(neo1973_snd_device); - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } -} -module_exit(neo1973_exit); +struct platform_driver neo1973_audio = { + .driver = { + .name = "neo1973-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = neo1973_probe, +}; +module_platform_driver(neo1973_audio); /* Module information */ MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:neo1973-audio"); -- 2.20.0

1 0

[alsa-devel] [PATCH 17/36] ARM: s3c: h1940-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-h1940.c | 18 ++++++++ sound/soc/samsung/h1940_uda1380.c | 71 ++++++++++-------------------- 2 files changed, 41 insertions(+), 48 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 74d6b68e91c7..922d53bacbdf 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -475,6 +475,22 @@ static struct gpiod_lookup_table h1940_mmc_gpio_table = { }, }; +static struct gpiod_lookup_table h1940_audio_gpio_table = { + .dev_id = "h1940-audio", + .table = { + GPIO_LOOKUP("H1940_LATCH", + H1940_LATCH_AUDIO_POWER - H1940_LATCH_GPIO(0), + "speaker-power", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOG", 4, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device h1940_audio = { + .name = "h1940-audio", + .id = -1, +}; + static struct pwm_lookup h1940_pwm_lookup[] = { PWM_LOOKUP("samsung-pwm", 0, "pwm-backlight", NULL, 36296, PWM_POLARITY_NORMAL), @@ -652,6 +668,7 @@ static struct platform_device *h1940_devices[] __initdata = { &s3c_device_ts, &power_supply, &h1940_battery, + &h1940_audio, }; static void __init h1940_map_io(void) @@ -691,6 +708,7 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); + gpiod_add_lookup_table(&h1940_audio_gpio_table); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index a95c34e53a2b..5fc001b4effd 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -15,9 +15,6 @@ #include <sound/jack.h> #include "regs-iis.h" -#include <asm/mach-types.h> - -#include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" static const unsigned int rates[] = { @@ -31,6 +28,8 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { .list = rates, }; +static struct gpio_desc *gpiod_speaker_power; + static struct snd_soc_jack hp_jack; static struct snd_soc_jack_pin hp_jack_pins[] = { @@ -47,7 +46,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { { - .gpio = S3C2410_GPG(4), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,9 +121,9 @@ static int h1940_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C_GPIO_END + 9, 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C_GPIO_END + 9, 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -151,8 +149,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; -static struct platform_device *s3c24xx_snd_device; - static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, @@ -194,55 +190,34 @@ static struct snd_soc_card h1940_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static int __init h1940_init(void) +static int h1940_probe(struct platform_device *pdev) { - int ret; + struct device *dev = &pdev->dev; - if (!machine_is_h1940()) - return -ENODEV; + h1940_asoc.dev = dev; + hp_jack_gpios[0].gpiod_dev = dev; + gpiod_speaker_power = devm_gpiod_get(&pdev->dev, "speaker-power", + GPIOD_OUT_LOW); - /* configure some gpios */ - ret = gpio_request(S3C_GPIO_END + 9, "speaker-power"); - if (ret) - goto err_out; - - ret = gpio_direction_output(S3C_GPIO_END + 9, 0); - if (ret) - goto err_gpio; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_gpio; + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "Could not get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) - goto err_plat; - - return 0; - -err_plat: - platform_device_put(s3c24xx_snd_device); -err_gpio: - gpio_free(S3C_GPIO_END + 9); - -err_out: - return ret; -} - -static void __exit h1940_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C_GPIO_END + 9); + return devm_snd_soc_register_card(dev, &h1940_asoc); } -module_init(h1940_init); -module_exit(h1940_exit); +static struct platform_driver h1940_audio_driver = { + .driver = { + .name = "h1940-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = h1940_probe, +}; +module_platform_driver(h1940_audio_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC H1940"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:h1940-audio"); -- 2.20.0

1 0

[alsa-devel] Record at 384 KHz
by GitHub issues - opened 10 Oct '19

10 Oct '19

alsa-project/alsa-lib issue #12 was opened from paddygoat: Tried everything I can think of to record at 384. How do I do it? Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/12 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH 0/9] ASoC: SOF updates for Intel/i.MX
by Pierre-Louis Bossart 10 Oct '19

10 Oct '19

Small set of updates for Intel and NXP platforms. On the Intel side, the code is now aligned with the HDaudio legacy with the sync_write. OEMs reported concerns with mute LEDs not supported with SOF/UCM, we can now tie a topology-defined control with a mute LED. Note that the large set of differences in the topology is largely due to code move, the new functionality is minimal in size. On the i.MX side, add topology support for ESAI parameters. This reflects on-going work on the firmware side. And last add debug module parameters for ACPI and PCI modules. For now the only flag used is for PM runtime, e.g. to measure the suspend-resume latencies of child devices without dealing with DSP reinitialization or keep the parent active while testing the SoundWire clock-stop-mode. Daniel Baluta (2): ASoC: SOF: imx: Describe ESAI parameters to be sent to DSP ASoC: SOF: imx: Read ESAI parameters and send them to DSP Jaska Uimonen (3): ASoC: SOF: enable dual control for pga AsoC: SOF: refactor control load code ASoC: SOF: acpi led support for switch controls Kai Vehmanen (2): ASoC: SOF: enable sync_write in hdac_bus Revert "ASoC: SOF: Force polling mode on CFL and CNL" Pierre-Louis Bossart (2): ASoC: SOF: acpi: add debug module param ASoC: SOF: pci: add debug module param include/sound/sof/dai-imx.h | 34 +++ include/sound/sof/dai.h | 2 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 7 +- sound/soc/sof/control.c | 32 +++ sound/soc/sof/intel/hda.c | 8 +- sound/soc/sof/pcm.c | 8 + sound/soc/sof/sof-acpi-dev.c | 12 +- sound/soc/sof/sof-pci-dev.c | 12 +- sound/soc/sof/sof-priv.h | 9 + sound/soc/sof/topology.c | 401 +++++++++++++++++++------------- 11 files changed, 354 insertions(+), 173 deletions(-) create mode 100644 include/sound/sof/dai-imx.h -- 2.20.1

2 20

[alsa-devel] Applied "ASoC: tas2562: Introduce the TAS2562 amplifier" to the asoc tree
by Mark Brown 10 Oct '19

10 Oct '19

The patch ASoC: tas2562: Introduce the TAS2562 amplifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c173dba44c2d2ec2adaa4e607e9729b74236bd57 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Tue, 8 Oct 2019 13:15:17 -0500 Subject: [PATCH] ASoC: tas2562: Introduce the TAS2562 amplifier Introduce the Texas Instruments TAS2562 amplifier driver with I/V sense for loud speaker applications. http://www.ti.com/lit/gpn/tas2562 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20191008181517.5332-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2562.c | 590 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 85 ++++++ 4 files changed, 682 insertions(+) create mode 100644 sound/soc/codecs/tas2562.c create mode 100644 sound/soc/codecs/tas2562.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ae213c6392ec..f4747ebc251e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2562 if I2C select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -1133,6 +1134,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2562 + tristate "Texas Instruments TAS2562 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2118407a1dd5..ddfd07071925 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-max98504-objs := max98504.o snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o +snd-soc-tas2562-objs := tas2562.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -481,6 +482,7 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o +obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c new file mode 100644 index 000000000000..729acd874c48 --- /dev/null +++ b/sound/soc/codecs/tas2562.c @@ -0,0 +1,590 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for the Texas Instruments TAS2562 CODEC +// Copyright (C) 2019 Texas Instruments Inc. + + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> +#include <linux/delay.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tas2562.h" + +#define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FORMAT_S32_LE) + +struct tas2562_data { + struct snd_soc_component *component; + struct gpio_desc *sdz_gpio; + struct regmap *regmap; + struct device *dev; + struct i2c_client *client; + int v_sense_slot; + int i_sense_slot; +}; + +static int tas2562_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2562_data *tas2562 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_ACTIVE); + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + break; + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_SHUTDOWN); + break; + + default: + dev_err(tas2562->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +static int tas2562_set_samplerate(struct tas2562_data *tas2562, int samplerate) +{ + int samp_rate; + int ramp_rate; + + switch (samplerate) { + case 7350: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 8000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 14700: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 16000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 22050: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 24000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 29400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 32000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 44100: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 48000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 88200: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 96000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 176400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + case 192000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + default: + dev_info(tas2562->dev, "%s, unsupported sample rate, %d\n", + __func__, samplerate); + return -EINVAL; + } + + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_RAMPRATE_MASK, ramp_rate); + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_SAMPRATE_MASK, samp_rate); + + return 0; +} + +static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret = 0; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_16B); + break; + case 24: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_24B); + break; + case 32: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_32B); + break; + + case 0: + /* Do not change slot width */ + break; + default: + dev_err(tas2562->dev, "slot width not supported"); + ret = -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) +{ + int ret; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_16B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_24B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_32B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + + default: + dev_info(tas2562->dev, "Not supported params format\n"); + } + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2562_set_bitwidth(tas2562, params_format(params)); + if (ret) { + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + return ret; + } + + ret = tas2562_set_samplerate(tas2562, params_rate(params)); + if (ret) + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + + return ret; +} + +static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2562_TDM_CFG1_RX_FALLING; + break; + default: + dev_err(tas2562->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_EDGE_MASK, + asi_cfg_1); + if (ret < 0) { + dev_err(tas2562->dev, "Failed to set RX edge\n"); + return ret; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + case (SND_SOC_DAIFMT_DSP_A): + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = BIT(1); + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", + fmt); + ret = -EINVAL; + break; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_OFFSET_MASK, + tdm_rx_start_slot); + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + return snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + mute ? TAS2562_MUTE : 0); +} + +static int tas2562_codec_probe(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + tas2562->component = component; + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + ret = snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + if (ret < 0) + return ret; + + return 0; +} + +#ifdef CONFIG_PM +static int tas2562_suspend(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(tas2562->regmap, true); + regcache_mark_dirty(tas2562->regmap); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 0); + + return 0; +} + +static int tas2562_resume(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + regcache_cache_only(tas2562->regmap, false); + + return regcache_sync(tas2562->regmap); +} +#else +#define tas2562_suspend NULL +#define tas2562_resume NULL +#endif + +static const char * const tas2562_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL(tas2562_ASI1_src_enum, TAS2562_TDM_CFG2, 4, + tas2562_ASI1_src); + +static const struct snd_kcontrol_new tas2562_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2562_ASI1_src_enum); + +static int tas2562_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + break; + default: + break; + } + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_ISENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_VSENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new tas2562_snd_controls[] = { + SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, + tas2562_dac_tlv), +}; + +static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2562_PWR_CTRL, 2, 1, &vsense_switch), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON"), + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route tas2562_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static const struct snd_soc_component_driver soc_component_dev_tas2562 = { + .probe = tas2562_codec_probe, + .suspend = tas2562_suspend, + .resume = tas2562_resume, + .set_bias_level = tas2562_set_bias_level, + .controls = tas2562_snd_controls, + .num_controls = ARRAY_SIZE(tas2562_snd_controls), + .dapm_widgets = tas2562_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2562_dapm_widgets), + .dapm_routes = tas2562_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2562_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { + .hw_params = tas2562_hw_params, + .set_fmt = tas2562_set_dai_fmt, + .set_tdm_slot = tas2562_set_dai_tdm_slot, + .digital_mute = tas2562_mute, +}; + +static struct snd_soc_dai_driver tas2562_dai[] = { + { + .name = "tas2562-amplifier", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, + .ops = &tas2562_speaker_dai_ops, + }, +}; + +static const struct regmap_range_cfg tas2562_ranges[] = { + { + .range_min = 0, + .range_max = 5 * 128, + .selector_reg = TAS2562_PAGE_CTRL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct reg_default tas2562_reg_defaults[] = { + { TAS2562_PAGE_CTRL, 0x00 }, + { TAS2562_SW_RESET, 0x00 }, + { TAS2562_PWR_CTRL, 0x0e }, + { TAS2562_PB_CFG1, 0x20 }, + { TAS2562_TDM_CFG0, 0x09 }, + { TAS2562_TDM_CFG1, 0x02 }, +}; + +static const struct regmap_config tas2562_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = 5 * 128, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = tas2562_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2562_reg_defaults), + .ranges = tas2562_ranges, + .num_ranges = ARRAY_SIZE(tas2562_ranges), +}; + +static int tas2562_parse_dt(struct tas2562_data *tas2562) +{ + struct device *dev = tas2562->dev; + int ret = 0; + + tas2562->sdz_gpio = devm_gpiod_get_optional(dev, "shut-down-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2562->sdz_gpio)) { + if (PTR_ERR(tas2562->sdz_gpio) == -EPROBE_DEFER) { + tas2562->sdz_gpio = NULL; + return -EPROBE_DEFER; + } + } + + ret = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2562->i_sense_slot); + if (ret) + dev_err(dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", ret); + + return ret; +} + +static int tas2562_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas2562_data *data; + int ret; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->client = client; + data->dev = &client->dev; + + tas2562_parse_dt(data); + + data->regmap = devm_regmap_init_i2c(client, &tas2562_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + dev_set_drvdata(&client->dev, data); + + return devm_snd_soc_register_component(dev, &soc_component_dev_tas2562, + tas2562_dai, + ARRAY_SIZE(tas2562_dai)); + +} + +static const struct i2c_device_id tas2562_id[] = { + { "tas2562", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2562_id); + +static const struct of_device_id tas2562_of_match[] = { + { .compatible = "ti,tas2562", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas2562_of_match); + +static struct i2c_driver tas2562_i2c_driver = { + .driver = { + .name = "tas2562", + .of_match_table = of_match_ptr(tas2562_of_match), + }, + .probe = tas2562_probe, + .id_table = tas2562_id, +}; + +module_i2c_driver(tas2562_i2c_driver); + +MODULE_AUTHOR("Dan Murphy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2562 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h new file mode 100644 index 000000000000..60f2bb1d198b --- /dev/null +++ b/sound/soc/codecs/tas2562.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tas2562.h - ALSA SoC Texas Instruments TAS2562 Mono Audio Amplifier + * + * Copyright (C) 2019 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + */ + +#ifndef __TAS2562_H__ +#define __TAS2562_H__ + +#define TAS2562_PAGE_CTRL 0x00 + +#define TAS2562_REG(page, reg) ((page * 128) + reg) + +#define TAS2562_SW_RESET TAS2562_REG(0, 0x01) +#define TAS2562_PWR_CTRL TAS2562_REG(0, 0x02) +#define TAS2562_PB_CFG1 TAS2562_REG(0, 0x03) +#define TAS2562_MISC_CFG1 TAS2562_REG(0, 0x04) +#define TAS2562_MISC_CFG2 TAS2562_REG(0, 0x05) + +#define TAS2562_TDM_CFG0 TAS2562_REG(0, 0x06) +#define TAS2562_TDM_CFG1 TAS2562_REG(0, 0x07) +#define TAS2562_TDM_CFG2 TAS2562_REG(0, 0x08) +#define TAS2562_TDM_CFG3 TAS2562_REG(0, 0x09) +#define TAS2562_TDM_CFG4 TAS2562_REG(0, 0x0a) +#define TAS2562_TDM_CFG5 TAS2562_REG(0, 0x0b) +#define TAS2562_TDM_CFG6 TAS2562_REG(0, 0x0c) +#define TAS2562_TDM_CFG7 TAS2562_REG(0, 0x0d) +#define TAS2562_TDM_CFG8 TAS2562_REG(0, 0x0e) +#define TAS2562_TDM_CFG9 TAS2562_REG(0, 0x0f) +#define TAS2562_TDM_CFG10 TAS2562_REG(0, 0x10) +#define TAS2562_TDM_DET TAS2562_REG(0, 0x11) +#define TAS2562_REV_ID TAS2562_REG(0, 0x7d) + +/* Page 2 */ +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) + +#define TAS2562_RESET BIT(0) + +#define TAS2562_MODE_MASK 0x3 +#define TAS2562_ACTIVE 0x0 +#define TAS2562_MUTE 0x1 +#define TAS2562_SHUTDOWN 0x2 + +#define TAS2562_TDM_CFG1_RX_EDGE_MASK BIT(0) +#define TAS2562_TDM_CFG1_RX_FALLING 1 +#define TAS2562_TDM_CFG1_RX_OFFSET_MASK GENMASK(4, 0) + +#define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) +#define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) +#define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 + +#define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) + +#define TAS2562_TDM_CFG2_RXLEN_MASK GENMASK(0, 1) +#define TAS2562_TDM_CFG2_RXLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXLEN_24B BIT(0) +#define TAS2562_TDM_CFG2_RXLEN_32B BIT(1) + +#define TAS2562_TDM_CFG2_RXWLEN_MASK GENMASK(3, 2) +#define TAS2562_TDM_CFG2_RXWLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXWLEN_20B BIT(2) +#define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) +#define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) + +#define TAS2562_VSENSE_POWER_EN BIT(2) +#define TAS2562_ISENSE_POWER_EN BIT(3) + +#define TAS2562_TDM_CFG5_VSNS_EN BIT(6) +#define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) + +#define TAS2562_TDM_CFG6_ISNS_EN BIT(6) +#define TAS2562_TDM_CFG6_ISNS_SLOT_MASK GENMASK(5, 0) + +#endif /* __TAS2562_H__ */ -- 2.20.1

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[alsa-devel] Question about DPCM FE vs BE
by Kuninori Morimoto 10 Oct '19

10 Oct '19

Hi ALSA ML In my understanding, DPCM needs FE and BE. And, one FE can have multiple BE, and one BE can have multiple FE. My question this time is that one snd_soc_pcm_runtime can be both FE and BE in the same time (= Sometimes FE, sometimes BE) ?? In my understanding, it never happen. But, is this correct ? I'm asking because do we need .be_clients/.fe_clients ? If one pcm_runtime can't be FE / BE in the same time, just .clients is enough I think. static int dpcm_be_connect(...) { ... - list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); - list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + list_add(&dpcm->list_be, &fe->dpcm[stream].clients); + list_add(&dpcm->list_fe, &be->dpcm[stream].clients); ... } Thank you for your help !! Best regards --- Kuninori Morimoto

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