July 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v1] ASoc: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files
by Shenghao Ding 03 Jul '24

03 Jul '24

Add name_prefix as the prefix name of DSP firmwares and calibrated data files which stored speaker calibrated impedance. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..cc765d45c6b5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -394,8 +394,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; - scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", - tas_priv->dev_name); + if (tas_priv->name_prefix) + scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + tas_priv->name_prefix, tas_priv->dev_name); + else + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", @@ -418,8 +422,15 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * calibrated data inside algo. */ for (i = 0; i < tas_priv->ndev; i++) { - scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", - tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); ret = tas2781_load_calibration(tas_priv, tas_priv->cal_binaryname[i], i); if (ret != 0) -- 2.34.1

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[PATCH v3 0/2] Add master clock handling for nau8824
by Maxim Kochetkov 02 Jul '24

02 Jul '24

nau8824 has external MCLK pin. So add enable/disable external clock management. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- Changes in v3: - move bindings before their users - use devm_clk_get_optional() - let clock API care about NULL ptr Changes in v2: - move the devm_clk_get() to the nau8824_read_device() Maxim Kochetkov (2): ASoC: dt-bindings: nau8824: Add master clock handling ASoC: codecs: nau8824: Add master clock handling .../bindings/sound/nuvoton,nau8824.yaml | 8 +++++++ sound/soc/codecs/nau8824.c | 21 +++++++++++++++++-- sound/soc/codecs/nau8824.h | 1 + 3 files changed, 28 insertions(+), 2 deletions(-) -- 2.45.2

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[PATCH] firmware: cs_dsp: Don't allow writes to read-only controls
by Richard Fitzgerald 02 Jul '24

02 Jul '24

Add a check to cs_dsp_coeff_write_ctrl() to abort if the control is not writeable. The cs_dsp code originated as an ASoC driver (wm_adsp) where all controls were exported as ALSA controls. It relied on ALSA to enforce the read-only permission. Now that the code has been separated from ALSA/ASoC it must perform its own permission check. This isn't currently causing any problems so there shouldn't be any need to backport this. If the client of cs_dsp exposes the control as an ALSA control, it should set permissions on that ALSA control to protect it. The few uses of cs_dsp_coeff_write_ctrl() inside drivers are for writable controls. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 16484ab9b09d..be8747d227db 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -796,6 +796,9 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, lockdep_assert_held(&ctl->dsp->pwr_lock); + if (ctl->flags && !(ctl->flags & WMFW_CTL_FLAG_WRITEABLE)) + return -EPERM; + if (len + off * sizeof(u32) > ctl->len) return -EINVAL; -- 2.39.2

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[PATCH 0/3] ASoC: cs35l56: Remove obsolete and redundant code
by Richard Fitzgerald 02 Jul '24

02 Jul '24

These commits remove various code that is either no longer needed, or is redundant. Richard Fitzgerald (3): ASoC: cs35l56: Revert support for dual-ownership of ASP registers ASoC: cs35l56: Remove support for A1 silicon ASoC: cs35l56: Remove redundant clearing of clk_stop_mode1 include/sound/cs35l56.h | 11 +- sound/soc/codecs/cs35l56-sdw.c | 75 ----------- sound/soc/codecs/cs35l56-shared.c | 122 ++++-------------- sound/soc/codecs/cs35l56.c | 205 ++---------------------------- sound/soc/codecs/cs35l56.h | 2 - 5 files changed, 45 insertions(+), 370 deletions(-) -- 2.39.2

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[PATCH] ASoC: fsl: lpc3xxx-i2s: Include bitfield.h for FIELD_PREP
by Nathan Chancellor 02 Jul '24

02 Jul '24

bitfield.h is not explicitly included but it is required for FIELD_PREP to be expanded by the preprocessor. If it is not implicitly included, there will be a compiler error (as seen with ARCH=hexagon allmodconfig): sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP); | ^ sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8' 42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH' 65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH' 70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */ | ^ Include bitfield.h explicitly, so that FIELD_PREP is always expanded, clearing up the compiler error. Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/fsl/lpc3xxx-i2s.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/lpc3xxx-i2s.h b/sound/soc/fsl/lpc3xxx-i2s.h index eec755448478..b6657853017a 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.h +++ b/sound/soc/fsl/lpc3xxx-i2s.h @@ -9,6 +9,7 @@ #ifndef __SOUND_SOC_LPC3XXX_I2S_H #define __SOUND_SOC_LPC3XXX_I2S_H +#include <linux/bitfield.h> #include <linux/types.h> #include <linux/regmap.h> --- base-commit: 55f0a1fa6ea4e339c797e9a3292ca0caa4ab3885 change-id: 20240701-lpc32xx-asoc-fix-include-for-field_prep-140db845a209 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

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[PATCH 1/2] ASoC: codecs: lpass-rx-macro: Fix using NULL pointer in probe() dev_err
by Krzysztof Kozlowski 02 Jul '24

02 Jul '24

The 'rx->dev' is assigned closer to the end of the probe() function, so the dev_err() must not use it - it is still NULL at this point. Instead there is already a local 'dev' variable. Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 08308a63d1b7..8d4a61f7debc 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3843,7 +3843,7 @@ static int rx_macro_probe(struct platform_device *pdev) rx_2_5_defaults, sizeof(rx_2_5_defaults)); break; default: - dev_err(rx->dev, "Unsupported Codec version (%d)\n", rx->codec_version); + dev_err(dev, "Unsupported Codec version (%d)\n", rx->codec_version); ret = -EINVAL; goto err; } -- 2.43.0

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[PATCH] drivers:soundwire: qcom: cleanup port maask calculations
by Srinivas Kandagatla 02 Jul '24

02 Jul '24

Cleanup the port map calculations, existing masks of having separate masks for in and out ports is not really required. Having a single mask for all the ports in the controller is simple and cuts of some unnecessary code. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 22 ++++++---------------- 1 file changed, 6 insertions(+), 16 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index ce5cf3ecceb5..aed57002fd0e 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -197,8 +197,7 @@ struct qcom_swrm_ctrl { int num_dout_ports; int cols_index; int rows_index; - unsigned long dout_port_mask; - unsigned long din_port_mask; + unsigned long port_mask; u32 intr_mask; u8 rcmd_id; u8 wcmd_id; @@ -1146,11 +1145,7 @@ static void qcom_swrm_stream_free_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { - if (m_rt->direction == SDW_DATA_DIR_RX) - port_mask = &ctrl->dout_port_mask; - else - port_mask = &ctrl->din_port_mask; - + port_mask = &ctrl->port_mask; list_for_each_entry(p_rt, &m_rt->port_list, port_node) clear_bit(p_rt->num, port_mask); } @@ -1195,13 +1190,9 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, if (ctrl->bus.id != m_rt->bus->id) continue; - if (m_rt->direction == SDW_DATA_DIR_RX) { - maxport = ctrl->num_dout_ports; - port_mask = &ctrl->dout_port_mask; - } else { - maxport = ctrl->num_din_ports; - port_mask = &ctrl->din_port_mask; - } + port_mask = &ctrl->port_mask; + maxport = ctrl->num_dout_ports + ctrl->num_din_ports; + list_for_each_entry(s_rt, &m_rt->slave_rt_list, m_rt_node) { slave = s_rt->slave; @@ -1401,8 +1392,7 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) return -EINVAL; /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ - set_bit(0, &ctrl->dout_port_mask); - set_bit(0, &ctrl->din_port_mask); + set_bit(0, &ctrl->port_mask); ret = of_property_read_u8_array(np, "qcom,ports-offset1", off1, nports); --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240618-soundwire-port-map-008f9dfae0f9 Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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[PATCH] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 02 Jul '24

02 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

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[Patch v4 00/10] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 02 Jul '24

02 Jul '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (10): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Add missing dma and i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Use threaded irq to fix warning from del_timer_sync() .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 56 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 21 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 10 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 4 +- drivers/mtd/nand/raw/lpc32xx_mlc.c | 10 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 10 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 376 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 73 ++++ 19 files changed, 954 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

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[PATCH] ASoC: tas5805m: demystify DSP volume control coefficients
by Felix Kaechele 02 Jul '24

02 Jul '24

The original manufacturer sample driver code uses direct DSP register writes to control digital volume on this amplifier without further context as to why it's not using the documented DIG_VOL_CTRL register. A thread in the manufacturer's forums [1] suggests this might have been done to work around volume ramping being used when controlling the volume through the DIG_VOL_CTRL register. When volume is controlled through this register, reading and/or writing any register is blocked until the volume ramping has concluded and the setpoint is reached. Additionally, the sample code uses a lookup table to map decibel values to 9.23 formatted coefficients. For posterity, add references to where this is documented and why a lookup table may be used. Signed-off-by: Felix Kaechele <felix(a)kaechele.ca> [1]: https://e2e.ti.com/support/audio-group/audio/f/audio-forum/1165952/tas5805m… --- sound/soc/codecs/tas5805m.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index 3b53eba38a0b..59536c8b8e38 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -67,6 +67,14 @@ static const uint8_t dsp_cfg_preboot[] = { 0x00, 0x00, 0x7f, 0x00, 0x03, 0x02, }; +/* + * Lookup table for DSP volume coefficients. + * The formula uses floating point math, so a lookup table is used + * instead of computing values on the fly. + * Formula: round(10^(volume in dB/20)*2^23) + * The 9.23 format used here is documented in + * SLAA894 - "General Tuning Guide for TAS58xx Family" + */ static const uint32_t tas5805m_volume[] = { 0x0000001B, /* 0, -110dB */ 0x0000001E, /* 1, -109dB */ 0x00000021, /* 2, -108dB */ 0x00000025, /* 3, -107dB */ @@ -196,9 +204,13 @@ static void tas5805m_refresh(struct tas5805m_priv *tas5805m) regmap_write(rm, REG_BOOK, 0x8c); regmap_write(rm, REG_PAGE, 0x2a); - /* Refresh volume. The actual volume control documented in the - * datasheet doesn't seem to work correctly. This is a pair of - * DSP registers which are *not* documented in the datasheet. + /* Refresh volume. This writes the volume coefficients from + * the lookup table directly into the DSP registers. + * Digital volume control on this chip involves ramping which + * blocks register reads and writes until the desired setpoint + * is reached. + * The DSP memory maps are documented in + * SLOA263A - "TAS5805M, TAS5806M and TAS5806MD Process Flows" */ set_dsp_scale(rm, 0x24, tas5805m->vol[0]); set_dsp_scale(rm, 0x28, tas5805m->vol[1]); base-commit: 9fa5527b19b21848dfb09928ee66af1aac4a5700 -- 2.45.2

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