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

[PATCH] ASoC: cs43130: Constify struct reg_sequence and reg_sequences
by Christophe JAILLET 13 Aug '24

13 Aug '24

'struct reg_sequence' and 'struct reg_sequences' are not modified in this drivers. Constifying these structures moves some data to a read-only section, so increase overall security. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o After: ===== text data bss dec hex filename 55562 6729 64 62355 f393 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. --- sound/soc/codecs/cs43130.c | 40 +++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cb4ca80f36d2..f8e2fb69ada2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1805,7 +1805,7 @@ static struct attribute *hpload_attrs[] = { }; ATTRIBUTE_GROUPS(hpload); -static struct reg_sequence hp_en_cal_seq[] = { +static const struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1820,7 +1820,7 @@ static struct reg_sequence hp_en_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_en_cal_seq2[] = { +static const struct reg_sequence hp_en_cal_seq2[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1828,7 +1828,7 @@ static struct reg_sequence hp_en_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_dis_cal_seq[] = { +static const struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_DXD1, 0x99}, {CS43130_DXD12, 0}, @@ -1836,12 +1836,12 @@ static struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dis_cal_seq2[] = { +static const struct reg_sequence hp_dis_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dc_ch_l_seq[] = { +static const struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1851,12 +1851,12 @@ static struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_l_seq2[] = { +static const struct reg_sequence hp_dc_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_r_seq[] = { +static const struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1866,12 +1866,12 @@ static struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_dc_ch_r_seq2[] = { +static const struct reg_sequence hp_dc_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_ac_ch_l_seq[] = { +static const struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1881,12 +1881,12 @@ static struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_l_seq2[] = { +static const struct reg_sequence hp_ac_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_r_seq[] = { +static const struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1896,24 +1896,24 @@ static struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_ac_ch_r_seq2[] = { +static const struct reg_sequence hp_ac_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_cln_seq[] = { +static const struct reg_sequence hp_cln_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, }; struct reg_sequences { - struct reg_sequence *seq; - int size; - unsigned int msk; + const struct reg_sequence *seq; + int size; + unsigned int msk; }; -static struct reg_sequences hpload_seq1[] = { +static const struct reg_sequences hpload_seq1[] = { { .seq = hp_en_cal_seq, .size = ARRAY_SIZE(hp_en_cal_seq), @@ -1951,7 +1951,7 @@ static struct reg_sequences hpload_seq1[] = { }, }; -static struct reg_sequences hpload_seq2[] = { +static const struct reg_sequences hpload_seq2[] = { { .seq = hp_en_cal_seq2, .size = ARRAY_SIZE(hp_en_cal_seq2), @@ -2041,7 +2041,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, } static int cs43130_hpload_proc(struct cs43130_private *cs43130, - struct reg_sequence *seq, int seq_size, + const struct reg_sequence *seq, int seq_size, unsigned int rslt_msk, int ac_idx) { int ret; @@ -2122,7 +2122,7 @@ static void cs43130_imp_meas(struct work_struct *wk) int i, ret, ac_idx; struct cs43130_private *cs43130; struct snd_soc_component *component; - struct reg_sequences *hpload_seq; + const struct reg_sequences *hpload_seq; cs43130 = container_of(wk, struct cs43130_private, work); component = cs43130->component; -- 2.45.2

2 1

[PATCH V2 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

The existing code modifies IRAM and DRAM size after sha dma start for vangogh platform. The problem with this sequence is that it might cause sha dma failure when firmware code binary size is greater than the default IRAM size. To fix this issue, Move the iram-dram fence register sequence prior to sha dma start. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes since v1: - Rephrase commit description sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

4 4

[PATCH 1/2] ASoC: SOF: amd: move iram-dram fence register programming sequence
by Vijendar Mukunda 13 Aug '24

13 Aug '24

As per design, ACP iram-dram fence register sequence should be initiated before triggering SHA dma. This ensures that IRAM size will programmed correctly before initiaing SHA dma. Fixes: 094d11768f74 ("ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..b8d4f986f89d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -263,6 +263,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_STRT_ADDR, start_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); + + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_CMD, ACP_SHA_RUN); ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_TRANSFER_BYTE_CNT, @@ -280,17 +291,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { - /* Modify IRAM and DRAM size */ - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); - if (ret) - return ret; - ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); - if (ret) - return ret; - } - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); -- 2.34.1

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[PATCH] ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers
by Krzysztof Kozlowski 13 Aug '24

13 Aug '24

There was no active maintenance from Banajit Goswami - last email is from 2019 - so make obvious that Qualcomm sound drivers are maintained by only one person. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Not moving to credits because lore does not point to significant contributions. Just ~140 emails: https://lore.kernel.org/all/?q=f%3A%22Banajit+Goswami%22 --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c8e16a649a0e..43e7668aacb0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18419,7 +18419,6 @@ F: drivers/crypto/intel/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> -M: Banajit Goswami <bgoswami(a)quicinc.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linux-arm-msm(a)vger.kernel.org S: Supported -- 2.43.0

2 2

[PATCH] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781
by Shenghao Ding 13 Aug '24

13 Aug '24

Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..8f9574189e1f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -673,7 +673,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", -- 2.34.1

3 2

[PATCH] ASoC: codecs: ES8326: Adjust buttons
by Zhang Yi 13 Aug '24

13 Aug '24

To adapt to chrome, we have adjusted the buttons to match the system. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..5b4586d9e67a 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -765,11 +765,11 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + cur_button = SND_JACK_BTN_2; break; case 0x27: /* button volume down */ - cur_button = SND_JACK_BTN_2; + cur_button = SND_JACK_BTN_3; break; case 0x1e: case 0xe2: @@ -785,7 +785,7 @@ static void es8326_jack_button_handler(struct work_struct *work) if (press_count > 3) { /* report a press every 120ms */ snd_soc_jack_report(es8326->jack, cur_button, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); press_count = 0; } button_to_report = cur_button; @@ -800,9 +800,9 @@ static void es8326_jack_button_handler(struct work_struct *work) /* released or no pressed */ if (button_to_report != 0) { snd_soc_jack_report(es8326->jack, button_to_report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); button_to_report = 0; } es8326_disable_micbias(es8326->component); @@ -845,7 +845,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->jack->status & SND_JACK_HEADPHONE) { dev_dbg(comp->dev, "Report hp remove event\n"); snd_soc_jack_report(es8326->jack, 0, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); /* mute adc when mic path switch */ regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); -- 2.17.1

2 1

[PATCH v2] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 13 Aug '24

13 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Change data type from int to __be32. v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 8a7fe48043d2..5a8193285f9f 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,22 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; + int offset = 0; int i, j, rc; + __be32 data; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.43.0

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[PATCH 1/8] ASoC: SOF: amd: Fix for incorrect acp error satus register offset
by Vijendar Mukunda 13 Aug '24

13 Aug '24

Adding 'dsp_intr_base' to ACP error status register offset in irq handler points to wrong register offset. ACP error status register offset got changed from ACP 6.0 onwards. Add 'acp_error_stat' descriptor field and update the value based on the ACP variant. Fixes: 0e44572a28a4 ("ASoC: SOF: amd: Add helper callbacks for ACP's DMA configuration") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp-dsp-offset.h | 3 ++- sound/soc/sof/amd/acp.c | 5 +++-- sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + sound/soc/sof/amd/pci-rmb.c | 1 + sound/soc/sof/amd/pci-rn.c | 1 + 6 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 59afbe2e0f42..66968efda869 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -76,7 +76,8 @@ #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 -#define ACP_ERROR_STATUS 0x18C4 +#define ACP3X_ERROR_STATUS 0x18C4 +#define ACP6X_ERROR_STATUS 0x1A4C #define ACP3X_AXI2DAGB_SEM_0 0x1880 #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7b122656efd1..d0b7d1c54248 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -92,6 +92,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, unsigned int idx, unsigned int dscr_count) { struct snd_sof_dev *sdev = adata->dev; + const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int val, status; int ret; @@ -102,7 +103,7 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, val & (1 << ch), ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); if (ret < 0) { - status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_ERROR_STATUS); + status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->acp_error_stat); val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_ERR_STS_0 + ch * sizeof(u32)); dev_err(sdev->dev, "ACP_DMA_ERR_STS :0x%x ACP_ERROR_STATUS :0x%x\n", val, status); @@ -404,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW0_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_SW1_I2S_ERROR_REASON, 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, base + ACP_ERROR_STATUS, 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index ec9170b3f068..6ac853ff6093 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -203,6 +203,7 @@ struct sof_amd_acp_desc { u32 probe_reg_offset; u32 reg_start_addr; u32 reg_end_addr; + u32 acp_error_stat; u32 sdw_max_link_count; u64 sdw_acpi_dev_addr; }; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index 54d42f83ce9e..c3da70549995 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -35,6 +35,7 @@ static const struct sof_amd_acp_desc acp63_chip_info = { .ext_intr_cntl = ACP6X_EXTERNAL_INTR_CNTL, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .ext_intr_stat1 = ACP6X_EXT_INTR_STAT1, + .acp_error_stat = ACP6X_ERROR_STATUS, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 4bc30951f8b0..194b7ff37e9e 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc rembrandt_chip_info = { .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, .dsp_intr_base = ACP6X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP6X_ERROR_STATUS, .sram_pte_offset = ACP6X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP6X_AXI2DAGB_SEM_0, .fusion_dsp_offset = ACP6X_DSP_FUSION_RUNSTALL, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index e08875bdfa8b..bff2d979ea6a 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -33,6 +33,7 @@ static const struct sof_amd_acp_desc renoir_chip_info = { .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, .dsp_intr_base = ACP3X_DSP_SW_INTR_BASE, + .acp_error_stat = ACP3X_ERROR_STATUS, .sram_pte_offset = ACP3X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP3X_AXI2DAGB_SEM_0, .acp_clkmux_sel = ACP3X_CLKMUX_SEL, -- 2.34.1

4 16

[PATCH 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 13 Aug '24

13 Aug '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: sma1307: Add bindings for Iron Device SMA1307 amplifier .../bindings/sound/irondevice,sma1307.yaml | 66 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2630 +++++++++++++++++ sound/soc/codecs/sma1307.h | 456 +++ 5 files changed, 3162 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

4 8

[PATCH RESEND 0/8] AMD SOF stack fixes and improvements
by Vijendar Mukunda 12 Aug '24

12 Aug '24

This patch series consists of few fixes and improvments for AMD SOF stack. Link: https://github.com/thesofproject/linux/pull/5103 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (8): ASoC: SOF: amd: Fix for incorrect acp error satus register offset ASoC: SOF: amd: fix for acp error reason registers wrong offset ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: fix for acp init sequence ASoC: SOF: amd: update conditional check for cache register update ASoC: SOF: amd: modify psp send command conditional checks ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure ASoC: amd: acp: Convert comma to semicolon sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +-- sound/soc/sof/amd/acp-dsp-offset.h | 6 ++- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 59 ++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 10 +++-- sound/soc/sof/amd/pci-acp63.c | 3 +- sound/soc/sof/amd/pci-rmb.c | 3 +- sound/soc/sof/amd/pci-rn.c | 3 +- sound/soc/sof/amd/pci-vangogh.c | 1 - 9 files changed, 60 insertions(+), 33 deletions(-) -- 2.34.1

2 9

[PATCH v1] ASoc: tas2781: Add new Kontrol to set tas2563 digital gain
by Shenghao Ding 11 Aug '24

11 Aug '24

Requriment from customer to add new kcontrol to set tas2563 digital gain and set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in tas2781-tlv.h. - Changed the copyright year to 2024 in tas2781.h. - Set "Speaker Force Firmware Load" as the common kcontrol for for both tas27871 and tas2563, that is tasdevice_snd_controls. - Only tas2781_snd_controls only for gain setting. - Add tas2563_dvc_tlv kcontrol for tas2563 digital gain setting. - In codec probe, for tas2563, loading tas2563 gain control; for tas2781, loading tas2781 gain control. --- include/sound/tas2781-tlv.h | 262 ++++++++++++++++++++++++++++++++- include/sound/tas2781.h | 3 +- sound/soc/codecs/tas2781-i2c.c | 135 ++++++++++++++++- 3 files changed, 392 insertions(+), 8 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 1dc59005d241..99c41bfc7827 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -17,5 +17,265 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index cd8ce522b78e..3275575f5bfb 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -50,6 +50,7 @@ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) /* Volume control */ +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..4960ab6577f7 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +#include <asm/unaligned.h> static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, @@ -103,7 +104,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_putvol(tas_priv, ucontrol, mc); } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +119,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +128,114 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + /* find out the member same as or closer to the current volume */ + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +243,13 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -578,6 +680,27 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); @@ -605,8 +728,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

4 3

[PATCH v2] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 11 Aug '24

11 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change to Shenghao Ding and Baojun Xu, drop entries for the audio converter and add entries for both haptics drivers and middle- power audio amplifier drivers. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index f328373463b0..d1f7e096b270 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22569,37 +22569,30 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c F: include/sound/tas2*.h -F: include/sound/tlv320*.h +F: include/sound/tas5086.h F: include/sound/tpa6130a2-plat.h F: sound/pci/hda/tas2781_hda_i2c.c -F: sound/soc/codecs/pcm1681.c -F: sound/soc/codecs/pcm1789*.* -F: sound/soc/codecs/pcm179x*.* -F: sound/soc/codecs/pcm186x*.* -F: sound/soc/codecs/pcm3008.* -F: sound/soc/codecs/pcm3060*.* -F: sound/soc/codecs/pcm3168a*.* -F: sound/soc/codecs/pcm5102a.c -F: sound/soc/codecs/pcm512x*.* F: sound/soc/codecs/tas2*.* -F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tas5*.* F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS -- 2.34.1

1 0

[PATCH 0/3] Add DisplayPort sound support for Fairphone 5 smartphone
by Luca Weiss 10 Aug '24

10 Aug '24

Add the necessary sound card bits and some dts additions to enable sound over DisplayPort-over-USB-C, e.g. to a connected TV or monitor. The UCM files can be found here: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/testing/devic… Two extra notes: 1. I don't quite understand whether the sound driver should have SoC-specific compatible or device-specific compatible. Some earlier patches by another author for a QCM6490 board and a QCS6490 board use device-specific compatible - but from what I can tell this is not how it's generally done for other sound drivers? 2. Unfortunately DisplayPort enablement itself for Fairphone 5 is not upstream yet. This is blocked by DSI display bringup upstream which is blocked by DSC 1:1:1 not being supported upstream yet and just working with a hacky patch. Nevertheless, DisplayPort audio was validated working with no additional sound-related changes so once DisplayPort gets enabled, sound should also just work upstream. Signed-off-by: Luca Weiss <luca.weiss(a)fairphone.com> --- Luca Weiss (3): ASoC: dt-bindings: qcom,sm8250: Add generic QCM6490 sound card ASoC: qcom: sc8280xp: Add support for generic QCM6490 arm64: dts: qcom: qcm6490-fairphone-fp5: Add DisplayPort sound support .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qcm6490-fairphone-fp5.dts | 37 ++++++++++++++++++++++ sound/soc/qcom/sc8280xp.c | 1 + 3 files changed, 39 insertions(+) --- base-commit: 6cdb38a56eaf615abc60fbeec0e4ccbdf93468e3 change-id: 20240809-fp5-dp-sound-b3768f3019bd Best regards, -- Luca Weiss <luca.weiss(a)fairphone.com>

4 9

[PATCH AUTOSEL 5.10 02/11] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 10 Aug '24

10 Aug '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 25bf73a7e7bfa..ad3694d36d969 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -234,6 +234,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

2 2

[PATCH] ALSA: pcm: Add xrun counter for snd_pcm_substream
by Norman Bintang 10 Aug '24

10 Aug '24

This patch adds an xrun counter to snd_pcm_substream as an alternative to using logs from XRUN_DEBUG_BASIC. The counter provides a way to track the number of xrun occurences, accessible through the /proc interface. The counter is enabled when CONFIG_SND_PCM_XRUN_DEBUG is set. Example output: $ cat /proc/asound/card0/pcm9p/sub0/status owner_pid : 1425 trigger_time: 235.248957291 tstamp : 0.000000000 delay : 1912 avail : 480 avail_max : 1920 ----- hw_ptr : 672000 appl_ptr : 673440 xrun_counter: 3 # (new row) Signed-off-by: Norman Bintang <normanbt(a)chromium.org> Reviewed-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Tested-by: Chih-Yang Hsia <paulhsia(a)chromium.org> Reviewed-by: David Riley <davidriley(a)chromium.org> --- include/sound/pcm.h | 3 +++ sound/core/pcm.c | 6 ++++++ sound/core/pcm_lib.c | 3 +++ 3 files changed, 12 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ac8f3aef9205..384032b6c59c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -498,6 +498,9 @@ struct snd_pcm_substream { /* misc flags */ unsigned int hw_opened: 1; unsigned int managed_buffer_alloc:1; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + unsigned int xrun_counter; /* number of times xrun happens */ +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index dc37f3508dc7..290690fc2abc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -462,6 +462,9 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + snd_iprintf(buffer, "xrun_counter: %d\n", substream->xrun_counter); +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -970,6 +973,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter = 0; +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e7905749c4a..6eaa950504cf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,6 +184,9 @@ void __snd_pcm_xrun(struct snd_pcm_substream *substream) pcm_warn(substream->pcm, "XRUN: %s\n", name); dump_stack_on_xrun(substream); } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter++; +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -- 2.46.0.76.ge559c4bf1a-goog

2 1

[PATCH 0/3] ALSA: update sample rate definition for eARC
by Jerome Brunet 10 Aug '24

10 Aug '24

This patchset updates some sample rate definition for eARC support. Notably it: * adds the definition for 128kHz * update the ASoC spdif codecs to include all supported rates, up to 768kHz * adds missing IEC958 definition for 128k, 352.4k, 384k and 705.6kHz On that last point, as noted in the related commit description, I'd like to stress that I have found the definitions using a testing equipement. I do not have access to the IEC958 specification. It would be best if someone with access to this specification could review/ack patch #2, before applying it. Jerome Brunet (3): ALSA: pcm: add support for 128kHz sample rate ALSA: IEC958 definition for consumer status channel update ASoC: spdif: extend supported rates to 768kHz include/sound/asoundef.h | 6 +++++- include/sound/pcm.h | 13 +++++++------ sound/core/pcm_native.c | 6 +++--- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- 5 files changed, 19 insertions(+), 12 deletions(-) -- 2.43.0

5 15

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - opened 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was opened from ViNi-Arco: What is the difference between `defaults.pcm.srate` and `defaults.pcm.rate`? I know there's `defaults.pcm.!rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

(Question) Difference between defaults.pcm.srate and defaults.pcm.rate
by GitHub issues - edited 09 Aug '24

09 Aug '24

alsa-project/alsa-lib issue #406 was edited from ViNi-Arco: What is the difference between `defaults.pcm.srate`? I know there's `defaults.pcm.!rate` and `defaults.pcm.rate`, but I couldn't find any information about defaults.pcm.srate I'm asking because I was given this configuration: ![IMG](https://github.com/user-attachments/assets/84316655-2d44-4f54-a842-86bd60a780e3) Issue URL : https://github.com/alsa-project/alsa-lib/issues/406 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoc: tas2781: Add Calibration Kcontrols and tas2563 digtial gain for Chromebook
by Shenghao Ding 09 Aug '24

09 Aug '24

Calibrated data will be set to default after loading DSP config params, which will cause speaker protection work abnormally. Reload calibrated data after loading DSP config params. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in the related files. - Add CAL_DAT_SZ for calibrated data size. - Add TAS2563_DVC_LVL for digtial gain kcontrol. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_restore for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - Add TASDEVICE_RCA_FW_OK to support only register setting bin in the device. - Add tas2563_dvc_table for relationship between tas2563 digtial gains and register values. - Correct the filename in the header comments of tas2781-comlib.c, tas2781-lib.c --> tas2781-comlib.c. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Add mutex into each kcontrol. - rename tas2781_force_fwload_get, tas2781_force_fwload_put--> tasdev_force_fwload_get, tasdev_force_fwload_put - rename tas2781_codec --> tasdev_codec - define tas2781_snd_control and tas2563_snd_control for volume - define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781-dsp.h | 1 + include/sound/tas2781.h | 92 ++- sound/soc/codecs/tas2781-comlib.c | 35 +- sound/soc/codecs/tas2781-fmwlib.c | 51 +- sound/soc/codecs/tas2781-i2c.c | 941 +++++++++++++++++++++++++++++- 5 files changed, 1061 insertions(+), 59 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 7fba7ea26a4b..74ab6f113029 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -121,6 +121,7 @@ enum tasdevice_dsp_fw_state { TASDEVICE_DSP_FW_NONE = 0, TASDEVICE_DSP_FW_PENDING, TASDEVICE_DSP_FW_FAIL, + TASDEVICE_RCA_FW_OK, TASDEVICE_DSP_FW_ALL_OK, }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 99ca3e401fd1..2fb3d03d0040 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -23,6 +23,8 @@ #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 #define TAS2563_GLOBAL_ADDR 0x48 +#define CAL_DAT_SZ 20 + #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) @@ -31,6 +33,11 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + /*PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 @@ -43,21 +50,73 @@ (page * 128)) + reg) /*Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x00, 0X00, 0x01) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) /*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x00, 0x00, 0x7E) + +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) /* Volume control */ -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) -#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x00, 0x00, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x00, 0x00, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) -#define TASDEVICE_CMD_SING_W 0x1 -#define TASDEVICE_CMD_BURST 0x2 -#define TASDEVICE_CMD_DELAY 0x3 -#define TASDEVICE_CMD_FIELD_W 0x4 +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) +#define TAS2563_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x0f, 0x48) +#define TAS2563_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x0f, 0x40) +#define TAS2563_PRM_POW_REG TASDEVICE_REG(0x00, 0x0d, 0x3c) +#define TAS2563_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x10, 0x14) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_R0_REG TASDEVICE_REG(0x00, 0x17, 0x74) +#define TAS2781_PRM_R0_LOW_REG TASDEVICE_REG(0x00, 0x18, 0x0c) +#define TAS2781_PRM_INVR0_REG TASDEVICE_REG(0x00, 0x18, 0x14) +#define TAS2781_PRM_POW_REG TASDEVICE_REG(0x00, 0x13, 0x70) +#define TAS2781_PRM_TLIMIT_REG TASDEVICE_REG(0x00, 0x18, 0x7c) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) enum audio_device { TAS2563, @@ -69,7 +128,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_restore; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -88,6 +155,8 @@ struct tasdevice_irqinfo { struct calidata { unsigned char *data; unsigned long total_sz; + unsigned int *reg_array; + unsigned int reg_array_sz; }; struct tasdevice_priv { @@ -122,6 +191,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -148,6 +218,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 3aa81514dad7..9d7b56663b6b 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tas2781-lib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers +// tas2781-comlib.c -- TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -89,6 +89,33 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdevice_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", + __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdevice_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { @@ -254,8 +281,8 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) } else { for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, - TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); + TASDEVICE_REG_SWRESET, + TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, "dev %d swreset fail, %d\n", diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 265a8ca25cbb..19a5133083dc 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2153,20 +2153,39 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (CAL_DAT_SZ + 1); + int j, rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + if (!priv->is_user_space_calidata && + cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (cal) + } + if (!priv->is_user_space_calidata) return; - - load_calib_data(priv, &cal->dev_data); + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + for (j = 0; j < cali_data->reg_array_sz; j++) { + if (data[k] != j) + rc = tasdevice_dev_bulk_write(priv, i, + cali_data->reg_array[j], + &(data[k + 4 * j]), 4); + if (rc < 0) + dev_err(priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2261,9 +2280,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; @@ -2324,13 +2344,15 @@ void tasdevice_tuning_switch(void *context, int state) struct tasdevice_fw *tas_fmw = tas_priv->fmw; int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + /* Only RCA file loaded still can work without speaker protection */ + if (!(tas_priv->fw_state == TASDEVICE_RCA_FW_OK || + tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK)) { + dev_err(tas_priv->dev, "No firmware loaded\n"); return; } if (state == 0) { - if (tas_priv->cur_prog < tas_fmw->nr_programs) { + if (tas_fmw && tas_priv->cur_prog < tas_fmw->nr_programs) { /*dsp mode or tuning mode*/ profile_cfg_id = tas_priv->rcabin.profile_cfg_id; tasdevice_select_tuningprm_cfg(tas_priv, @@ -2340,9 +2362,10 @@ void tasdevice_tuning_switch(void *context, int state) tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_POWER_UP); - } else + } else { tasdevice_select_cfg_blk(tas_priv, profile_cfg_id, TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } } EXPORT_SYMBOL_NS_GPL(tasdevice_tuning_switch, SND_SOC_TAS2781_FMWLIB); diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..239f3538c558 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -13,6 +13,7 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // +#include <asm/unaligned.h> #include <linux/crc8.h> #include <linux/firmware.h> #include <linux/gpio/consumer.h> @@ -31,6 +32,122 @@ #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_PRM_ENFF_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -47,6 +164,22 @@ static const struct of_device_id tasdevice_of_match[] = { MODULE_DEVICE_TABLE(of, tasdevice_of_match); #endif +static const int tas2563_cali_data_reg[] = { + TAS2563_PRM_R0_REG, + TAS2563_PRM_R0_LOW_REG, + TAS2563_PRM_INVR0_REG, + TAS2563_PRM_POW_REG, + TAS2563_PRM_TLIMIT_REG, +}; + +static const int tas2781_cali_data_reg[] = { + TAS2781_PRM_R0_REG, + TAS2781_PRM_R0_LOW_REG, + TAS2781_PRM_INVR0_REG, + TAS2781_PRM_POW_REG, + TAS2781_PRM_TLIMIT_REG, +}; + /** * tas2781_digital_getvol - get the volum control * @kcontrol: control pointer @@ -65,8 +198,13 @@ static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, @@ -76,8 +214,13 @@ static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_digital_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_digital_putvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, @@ -87,8 +230,13 @@ static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_getvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_getvol(tas_priv, ucontrol, mc); + return rc; } static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, @@ -99,11 +247,16 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(codec); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int rc; + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_amp_putvol(tas_priv, ucontrol, mc); + mutex_unlock(&tas_priv->codec_lock); - return tasdevice_amp_putvol(tas_priv, ucontrol, mc); + return rc; } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -118,7 +271,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -127,18 +280,496 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, snd_soc_component_get_drvdata(component); bool change, val = (bool)ucontrol->value.integer.value[0]; + mutex_lock(&tas_priv->codec_lock); if (tas_priv->force_fwload_status == val) change = false; else { change = true; tas_priv->force_fwload_status = val; } + mutex_unlock(&tas_priv->codec_lock); dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, tas_priv->force_fwload_status ? "ON" : "OFF"); return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = tas_priv->cali_data.data; + + mutex_lock(&tas_priv->codec_lock); + if (!tas_priv->is_user_space_calidata || + tas_priv->cali_data.total_sz != bytes_ext->max) { + goto out; + } + memcpy(dst, data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst, int data_len) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + if (data_len != 4) + return rc; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + rc = 0; + break; + } + } + + return rc; +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + unsigned char *dat = ucontrol->value.bytes.data; + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2781 && + bytes_ext->max != 8 * tas_priv->ndev) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + int k = i * 9; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + + if (dat[k] != i) { + dev_err(tas_priv->dev, + "%s: no cal-setting for dev %d\n", __func__, + i); + continue; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + &dat[i * 9 + 1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + &dat[i * 9 + 5], 4); + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *)tas2563_cali_start_reg[j].val, + 4); + } + } +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_restore; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->chip_id == TAS2563) + tas2563_calib_stop_put(tas_priv); + else + tas2781_calib_stop_put(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *data = tas_priv->cali_data.data; + int rc = 1; + + mutex_lock(&tas_priv->codec_lock); + if (tas_priv->cali_data.total_sz != bytes_ext->max) { + rc = -1; + goto out; + } + tas_priv->is_user_space_calidata = true; + memcpy(data, ucontrol->value.bytes.data, bytes_ext->max); +out: + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + mutex_lock(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + dst[1] = val; + rc = 0; + break; + } + } + mutex_unlock(&tas_priv->codec_lock); + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_PRM_R0_REG; + else + reg = TAS2563_PRM_R0_REG; + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + mutex_lock(&tas_priv->codec_lock); + rc = calib_data_get(tas_priv, reg, dst, bytes_ext->max - 1); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + int vol = ucontrol->value.integer.value[0]; + int max = mc->max, i, ret = 1; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + ret = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) + dev_err(tas_dev->dev, + "%s, set digital vol error in device %d\n", + __func__, i); + } + +out: + mutex_unlock(&tas_dev->codec_lock); + return ret; +} + +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 5, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 5, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 5, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 5, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 5, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -146,8 +777,22 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 2, tas2781_latch_reg_get, NULL), +}; + +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), +}; + +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -167,6 +812,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = (int)tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -223,6 +893,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -316,15 +997,54 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_active_id_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_id_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int nr_dev = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + nr_dev = clamp(nr_dev, 0, max); + + mutex_lock(&tas_priv->codec_lock); + rc = tasdevice_chn_switch(tas_priv, nr_dev); + mutex_unlock(&tas_priv->codec_lock); + + return rc; +} + +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { + char *active_dev_name, *cali_name, *chip_id, *conf_name, *prog_name; + struct calidata *cali_data = &tas_priv->cali_data; struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + struct soc_bytes_ext *ext_cali_data; + int nr_controls = 5; int mix_index = 0; int ret; + if (tas_priv->chip_id == TAS2781) + nr_controls++; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ @@ -336,11 +1056,20 @@ static int tasdevice_dsp_create_ctrls( } /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, + active_dev_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + cali_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + chip_id = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + prog_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - if (!prog_name || !conf_name) { + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!active_dev_name || !cali_name || !conf_name || !chip_id || + !ext_cali_data || !prog_name) { ret = -ENOMEM; goto out; } @@ -363,6 +1092,68 @@ static int tasdevice_dsp_create_ctrls( dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + scnprintf(active_dev_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Activate Tasdevice Id"); + dsp_ctrls[mix_index].name = active_dev_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_id; + dsp_ctrls[mix_index].get = tasdevice_active_id_get; + dsp_ctrls[mix_index].put = tasdevice_active_id_put; + mix_index++; + + scnprintf(chip_id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Tasdevice Chip Id"); + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + + scnprintf(cali_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Speaker Calibrated Data"); + ext_cali_data->max = tas_priv->ndev * CAL_DAT_SZ; + tas_priv->cali_data.total_sz = ext_cali_data->max; + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + dsp_ctrls[mix_index].name = cali_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].get = tasdev_cali_data_get; + dsp_ctrls[mix_index].put = tasdev_cali_data_put; + dsp_ctrls[mix_index].private_value = (unsigned long)ext_cali_data; + mix_index++; + + cali_data->data = devm_kzalloc(tas_priv->dev, tas_priv->ndev * + (cali_data->reg_array_sz * 4 + 1), GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + cali_start_name = devm_kzalloc(tas_priv->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!cali_start_name || !ext_cali_start) { + ret = -ENOMEM; + goto out; + } + + scnprintf(cali_start_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "Calibration Start"); + ext_cali_data->max = tas_priv->ndev * 9; + dsp_ctrls[mix_index].name = cali_start_name; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = snd_soc_bytes_info_ext; + dsp_ctrls[mix_index].put = tas2781_calib_start_put; + dsp_ctrls[mix_index].get = tasdev_nop_get; + dsp_ctrls[mix_index].private_value = + (unsigned long)ext_cali_start; + mix_index++; + } + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -370,6 +1161,54 @@ static int tasdevice_dsp_create_ctrls( return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + int rc; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali control err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + cali_data->reg_array = (unsigned int *)tas2781_cali_data_reg; + cali_data->reg_array_sz = ARRAY_SIZE(tas2781_cali_data_reg); + } else { + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + cali_data->reg_array = (unsigned int *)tas2563_cali_data_reg; + cali_data->reg_array_sz = sizeof(tas2563_cali_data_reg); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_restore = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_restore) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -380,23 +1219,36 @@ static void tasdevice_fw_ready(const struct firmware *fmw, mutex_lock(&tas_priv->codec_lock); ret = tasdevice_rca_parser(tas_priv, fmw); - if (ret) + if (ret) { + tasdevice_config_info_remove(tas_priv); goto out; + } tasdevice_create_control(tas_priv); tasdevice_dsp_remove(tas_priv); tasdevice_calbin_remove(tas_priv); - tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + tas_priv->fw_state = TASDEVICE_RCA_FW_OK; 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", tas_priv->coef_binaryname); - tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; goto out; } - tasdevice_dsp_create_ctrls(tas_priv); + + ret = tasdevice_dsp_create_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dsp controls error\n"); + goto out; + } + + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; @@ -417,9 +1269,8 @@ static void tasdevice_fw_ready(const struct firmware *fmw, tasdevice_prmg_load(tas_priv, 0); tas_priv->cur_prog = 0; out: - if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { - /*If DSP FW fail, kcontrol won't be created */ - tasdevice_config_info_remove(tas_priv); + if (tas_priv->fw_state == TASDEVICE_RCA_FW_OK) { + /*If DSP FW fail, DSP kcontrol won't be created */ tasdevice_dsp_remove(tas_priv); } mutex_unlock(&tas_priv->codec_lock); @@ -466,14 +1317,14 @@ static int tasdevice_startup(struct snd_pcm_substream *substream, { struct snd_soc_component *codec = dai->component; struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); - int ret = 0; - if (tas_priv->fw_state != TASDEVICE_DSP_FW_ALL_OK) { - dev_err(tas_priv->dev, "DSP bin file not loaded\n"); - ret = -EINVAL; + if (!(tas_priv->fw_state == TASDEVICE_DSP_FW_ALL_OK || + tas_priv->fw_state == TASDEVICE_RCA_FW_OK)) { + dev_err(tas_priv->dev, "Bin file not loaded\n"); + return -EINVAL; } - return ret; + return 0; } static int tasdevice_hw_params(struct snd_pcm_substream *substream, @@ -541,7 +1392,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", @@ -565,13 +1416,41 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } else { + rc = snd_soc_add_component_controls(codec, + tas2563_snd_controls, + ARRAY_SIZE(tas2563_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } + + rc = tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); - return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); + return rc; } static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_restore); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); @@ -591,8 +1470,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, -- 2.34.1

5 6

[GIT PULL] ASoC fixes for v6.11-rc2
by Mark Brown 09 Aug '24

09 Aug '24

The following changes since commit 8400291e289ee6b2bf9779ff1c83a291501f017b: Linux 6.11-rc1 (2024-07-28 14:19:55 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc2 for you to fetch changes up to 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 Quite a lot of fixes have come in since the merge window, there's some repetitive fixes over the Qualcomm drivers increasing the patch count, along with a large batch of fixes from Cirrus. We also have some quirks and some individual fixes. ---------------------------------------------------------------- Bruno Ancona (1): ASoC: amd: yc: Support mic on HP 14-em0002la Charles Keepax (5): ASoC: cs42l43: Remove redundant semi-colon at end of function ASoC: soc-component: Add new snd_soc_component_get_kcontrol() helpers ASoC: cs35l45: Use new snd_soc_component_get_kcontrol_locked() helper ASoC: cs42l43: Cache shutter IRQ control pointers MAINTAINERS: Update Cirrus Logic parts to linux-sound mailing list Curtis Malainey (1): ASoC: SOF: Remove libraries from topology lookups Francesco Dolcini (1): ASoC: nau8822: Lower debug print priority Jerome Audu (1): ASoC: sti: add missing probe entry for player and reader Jerome Brunet (1): ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT Johan Hovold (1): ASoC: codecs: lpass-macro: fix missing codec version Krzysztof Kozlowski (10): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd934x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd938x: Correct reset GPIO polarity in example ASoC: dt-bindings: qcom,wcd939x: Correct reset GPIO polarity in example Krzysztof Stępniak (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E14 Gen 6 Mark Brown (3): ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports ASoC: fsl_micfil: Check the difference for i.MX8 and ASoC: Extend wm_adsp so cs35l56 can suppress controls Paul Handrigan (1): ASoC: cs530x: Change IN HPF Select kcontrol name Richard Fitzgerald (2): ASoC: cs-amp-lib: Fix NULL pointer crash if efi.get_variable is NULL ASoC: cs35l56: Handle OTP read latency over SoundWire Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms Simon Trimmer (3): ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value Takashi Iwai (1): ASoC: amd: yc: Add quirk entry for OMEN by HP Gaming Laptop 16-n0xxx Zhang Yi (1): ASoC: codecs: ES8326: button detect issue .../devicetree/bindings/sound/qcom,wcd934x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 3 +- .../devicetree/bindings/sound/qcom,wcd939x.yaml | 4 +- MAINTAINERS | 4 +- include/sound/cs35l56.h | 5 ++ include/sound/soc-component.h | 5 ++ sound/soc/amd/yc/acp6x-mach.c | 21 ++++++ sound/soc/codecs/cs-amp-lib.c | 2 +- sound/soc/codecs/cs35l45.c | 11 +--- sound/soc/codecs/cs35l56-sdw.c | 77 ++++++++++++++++++++++ sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 11 ++++ sound/soc/codecs/cs42l43.c | 75 ++++++++++++++++----- sound/soc/codecs/cs42l43.h | 2 + sound/soc/codecs/cs530x.c | 8 +-- sound/soc/codecs/es8326.c | 2 + sound/soc/codecs/lpass-va-macro.c | 2 + sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/wcd937x-sdw.c | 4 +- sound/soc/codecs/wcd938x-sdw.c | 4 +- sound/soc/codecs/wcd939x-sdw.c | 4 +- sound/soc/codecs/wm_adsp.c | 17 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- sound/soc/fsl/fsl_micfil.c | 20 ++++-- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/meson/axg-fifo.c | 26 +++----- sound/soc/soc-component.c | 42 +++++++++--- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 36 files changed, 291 insertions(+), 84 deletions(-)

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[PATCH 0/2] ASoC: codecs: warn on unknown codec version
by Dmitry Baryshkov 09 Aug '24

09 Aug '24

While debugging broken audio issues on some of Qualcomm platforms I stumbled upon the kernel not providing the actual error information. It prints an error from the wsa_macro driver, but the actual issue is in the VA macro driver. Add error message to point to the actual error location. va_macro 3370000.codec: Unknown VA Codec version, ID: 00 / 0f / 00 wsa_macro 3240000.codec: Unsupported Codec version (0) Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> --- Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version sound/soc/codecs/lpass-macro-common.h | 6 ++++++ sound/soc/codecs/lpass-va-macro.c | 4 ++++ 2 files changed, 10 insertions(+) --- base-commit: 668d33c9ff922c4590c58754ab064aaf53c387dd change-id: 20240803-codec-version-db6f65ab51ce Best regards, -- Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org>

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[tiwai-sound:for-next 69/83] sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/83] ALSA: vxpocket: Use standard print API config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090809.TAYiyrqJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090809.TAYiyrqJ-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c:207:8: error: no member named 'dev' in 'struct vx_core' 207 | chip->dev = &link->dev; | ~~~~ ^ >> sound/pcmcia/vx/vxpocket.c:273:12: error: use of undeclared identifier 'pdev'; did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ | p_dev include/linux/dev_printk.h:154:44: note: expanded from macro 'dev_err' 154 | dev_printk_index_wrap(_dev_err, KERN_ERR, dev, dev_fmt(fmt), ##__VA_ARGS__) | ^ include/linux/dev_printk.h:110:11: note: expanded from macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^ sound/pcmcia/vx/vxpocket.c:251:49: note: 'p_dev' declared here 251 | static int vxpocket_probe(struct pcmcia_device *p_dev) | ^ 2 errors generated. vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:for-next 73/83] sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared; did you mean 'chip'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: d41abde894830bfb77252653f606473728e930eb [73/83] ALSA: sparc: Use standard print API config: sparc64-defconfig (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) compiler: sparc64-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090648.J2EAijjH-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from include/linux/device.h:15, from include/linux/platform_device.h:13, from sound/sparc/cs4231.c:21: sound/sparc/cs4231.c: In function 'snd_cs4231_ebus_create': >> sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared (first use in this function); did you mean 'chip'? 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ sound/sparc/cs4231.c:2018:25: note: each undeclared identifier is reported only once for each function it appears in 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ vim +2018 sound/sparc/cs4231.c 1950 1951 static int snd_cs4231_ebus_create(struct snd_card *card, 1952 struct platform_device *op, 1953 int dev) 1954 { 1955 struct snd_cs4231 *chip = card->private_data; 1956 int err; 1957 1958 spin_lock_init(&chip->lock); 1959 spin_lock_init(&chip->c_dma.ebus_info.lock); 1960 spin_lock_init(&chip->p_dma.ebus_info.lock); 1961 mutex_init(&chip->mce_mutex); 1962 mutex_init(&chip->open_mutex); 1963 chip->flags |= CS4231_FLAG_EBUS; 1964 chip->op = op; 1965 memcpy(&chip->image, &snd_cs4231_original_image, 1966 sizeof(snd_cs4231_original_image)); 1967 strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); 1968 chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1969 chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; 1970 chip->c_dma.ebus_info.client_cookie = chip; 1971 chip->c_dma.ebus_info.irq = op->archdata.irqs[0]; 1972 strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); 1973 chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1974 chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; 1975 chip->p_dma.ebus_info.client_cookie = chip; 1976 chip->p_dma.ebus_info.irq = op->archdata.irqs[1]; 1977 1978 chip->p_dma.prepare = _ebus_dma_prepare; 1979 chip->p_dma.enable = _ebus_dma_enable; 1980 chip->p_dma.request = _ebus_dma_request; 1981 chip->p_dma.address = _ebus_dma_addr; 1982 1983 chip->c_dma.prepare = _ebus_dma_prepare; 1984 chip->c_dma.enable = _ebus_dma_enable; 1985 chip->c_dma.request = _ebus_dma_request; 1986 chip->c_dma.address = _ebus_dma_addr; 1987 1988 chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); 1989 chip->p_dma.ebus_info.regs = 1990 of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); 1991 chip->c_dma.ebus_info.regs = 1992 of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); 1993 if (!chip->port || !chip->p_dma.ebus_info.regs || 1994 !chip->c_dma.ebus_info.regs) { 1995 snd_cs4231_ebus_free(chip); 1996 dev_dbg(chip->card->dev, 1997 "cs4231-%d: Unable to map chip registers.\n", dev); 1998 return -EIO; 1999 } 2000 2001 if (ebus_dma_register(&chip->c_dma.ebus_info)) { 2002 snd_cs4231_ebus_free(chip); 2003 dev_dbg(chip->card->dev, 2004 "cs4231-%d: Unable to register EBUS capture DMA\n", 2005 dev); 2006 return -EBUSY; 2007 } 2008 if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) { 2009 snd_cs4231_ebus_free(chip); 2010 dev_dbg(chip->card->dev, 2011 "cs4231-%d: Unable to enable EBUS capture IRQ\n", 2012 dev); 2013 return -EBUSY; 2014 } 2015 2016 if (ebus_dma_register(&chip->p_dma.ebus_info)) { 2017 snd_cs4231_ebus_free(chip); > 2018 dev_dbg(chpi->card->dev, 2019 "cs4231-%d: Unable to register EBUS play DMA\n", 2020 dev); 2021 return -EBUSY; 2022 } 2023 if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) { 2024 snd_cs4231_ebus_free(chip); 2025 dev_dbg(chip->card->dev, 2026 "cs4231-%d: Unable to enable EBUS play IRQ\n", dev); 2027 return -EBUSY; 2028 } 2029 2030 if (snd_cs4231_probe(chip) < 0) { 2031 snd_cs4231_ebus_free(chip); 2032 return -ENODEV; 2033 } 2034 snd_cs4231_init(chip); 2035 2036 err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, 2037 chip, &snd_cs4231_ebus_dev_ops); 2038 if (err < 0) { 2039 snd_cs4231_ebus_free(chip); 2040 return err; 2041 } 2042 2043 return 0; 2044 } 2045 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:for-next 32/83] sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core'
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/83] ALSA: vx_core: Drop unused dev field config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090607.58bm8VK5-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core' 208 | chip->dev = &link->dev; | ~~~~ ^ 1 error generated. vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH v2] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT
by Jerome Brunet 08 Aug '24

08 Aug '24

With PREEMPT_RT enabled a spinlock_t becomes a sleeping lock. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the primary handler won't be force-threaded and runs always in hardirq context. This is a problem because spinlock_t requires a preemptible context on PREEMPT_RT. In this particular instance, regmap mmio uses spinlock_t to protect the register access and IRQF_ONESHOT is set on the IRQ. In this case, it is actually better to do everything in threaded handler and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@sa… Suggested-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..75909196b769 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the threaded irq handler only with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- 2.43.0

3 2

[PATCH 1/4] ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example
by Krzysztof Kozlowski 08 Aug '24

08 Aug '24

The reset GPIO of WCD9370/WCD9375 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Cc: Alexey Klimov <alexey.klimov(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml index de397d879acc..f94203798f24 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml @@ -42,7 +42,7 @@ examples: pinctrl-names = "default", "sleep"; pinctrl-0 = <&wcd_reset_n>; pinctrl-1 = <&wcd_reset_n_sleep>; - reset-gpios = <&tlmm 83 GPIO_ACTIVE_HIGH>; + reset-gpios = <&tlmm 83 GPIO_ACTIVE_LOW>; vdd-buck-supply = <&vreg_l17b_1p8>; vdd-rxtx-supply = <&vreg_l18b_1p8>; vdd-px-supply = <&vreg_l18b_1p8>; -- 2.43.0

2 4

[PATCH v2 0/3] ASoC: tas*: Fix up GPIO usage
by Linus Walleij 08 Aug '24

08 Aug '24

The TI TAS drivers use some legacy GPIO code and headers, this series fixes it up. The TAS2781 is a special case since it adds a handful of lines of deviating code to reconfigure a GPIO line for IRQ mode and then never actually use the IRQ obtained in the code. Is the line used by autonomous hardware? I'm puzzled by this. Anyways the patch suggest how to solve this properly by fixing the parent irqchip and I'm happy to help. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased onto v6.11-rc1 - Link to v1: https://lore.kernel.org/r/20240701-asoc-tas-gpios-v1-0-d69ec5d79939@linaro.… --- Linus Walleij (3): ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 26 ++++---------------------- 9 files changed, 6 insertions(+), 37 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240701-asoc-tas-gpios-5c051d80d768 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 4

[PATCH] ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value
by Simon Trimmer 08 Aug '24

08 Aug '24

Device tuning files made with early revision tooling may contain configuration that can unmask IRQ signals that are owned by the host. Adding a safe default to the regmap patch ensures that the hardware matches the driver expectations. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..bd74fef33d49 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -36,6 +36,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, + { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, -- 2.43.0

2 1

[PATCH 0/2] ASoC: codecs: ES8326: button and input issue
by Zhang Yi 08 Aug '24

08 Aug '24

TO solve button and input issue, We have modified parts of the driver related to button and input. Zhang Yi (2): ASoC: codecs: ES8326: button detect issue ASoC: codecs: ES8326: input issue after init sound/soc/codecs/es8326.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.17.1

2 3

[PATCH] ASoC: amd: acp: add ZSC control register programming sequence
by Vijendar Mukunda 08 Aug '24

08 Aug '24

Add ZSC Control register programming sequence for ACP D0 and D3 state transitions for ACP7.0 onwards. This will allow ACP to enter low power state when ACP enters D3 state. When ACP enters D0 State, ZSC control should be disabled. Tested-by: Leo Li <sunpeng.li(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 5 +++++ sound/soc/amd/acp/amd.h | 2 ++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 4422cec81e3c..04bd605fdce3 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -321,6 +321,8 @@ int acp_init(struct acp_chip_info *chip) pr_err("ACP reset failed\n"); return ret; } + if (chip->acp_rev >= ACP70_DEV) + writel(0, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_init, SND_SOC_ACP_COMMON); @@ -336,6 +338,9 @@ int acp_deinit(struct acp_chip_info *chip) if (chip->acp_rev != ACP70_DEV) writel(0, chip->base + ACP_CONTROL); + + if (chip->acp_rev >= ACP70_DEV) + writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_deinit, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 87a4813783f9..c095a34a7229 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -103,6 +103,8 @@ #define ACP70_PGFSM_CONTROL ACP6X_PGFSM_CONTROL #define ACP70_PGFSM_STATUS ACP6X_PGFSM_STATUS +#define ACP_ZSC_DSP_CTRL 0x0001014 +#define ACP_ZSC_STS 0x0001018 #define ACP_SOFT_RST_DONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 0xffffffff -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver
by Vijendar Mukunda 08 Aug '24

08 Aug '24

As module device table added for AMD SoundWire machine driver MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD SoundWire machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 3419675e45a9..f7e4af850309 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -738,5 +738,4 @@ module_platform_driver(sof_sdw_driver); MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); MODULE_AUTHOR("Vijendar Mukunda <Vijendar.Mukunda(a)amd.com"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:amd_sof_sdw"); MODULE_IMPORT_NS(SND_SOC_SDW_UTILS); -- 2.34.1

4 3

[PATCH] ASoC: cs35l56: Use regmap_read_bypassed() to wake the device
by Simon Trimmer 08 Aug '24

08 Aug '24

Now that regmap_read_bypassed() has been added to the kernel it is preferable to wake the device with a read rather than a write as the utility function can be called at a time before the device has been identified. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..91b3c1c8575c 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -450,32 +450,23 @@ static const struct reg_sequence cs35l56_hibernate_seq[] = { REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_ALLOW_AUTO_HIBERNATE), }; -static const struct reg_sequence cs35l56_hibernate_wake_seq[] = { - REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_WAKEUP), -}; - static void cs35l56_issue_wake_event(struct cs35l56_base *cs35l56_base) { + unsigned int val; + /* * Dummy transactions to trigger I2C/SPI auto-wake. Issue two * transactions to meet the minimum required time from the rising edge * to the last falling edge of wake. * - * It uses bypassed write because we must wake the chip before + * It uses bypassed read because we must wake the chip before * disabling regmap cache-only. - * - * This can NAK on I2C which will terminate the write sequence so the - * single-write sequence is issued twice. */ - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); usleep_range(CS35L56_WAKE_HOLD_TIME_US, 2 * CS35L56_WAKE_HOLD_TIME_US); - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); cs35l56_wait_control_port_ready(); } -- 2.43.0

2 1

[tiwai-sound:for-next 66/82] sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: f8466d91f36d97e237ea1b3c6f19de980054a1b6 [66/82] ALSA: usx2y: Use standard print API config: x86_64-randconfig-161-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090249.8LE9qrae-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/usb/usx2y/us122l.c:15: >> sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used [-Wunused-but-set-variable] 668 | int frame = 0, iters = 0; | ^ 1 warning generated. vim +/iters +668 sound/usb/usx2y/usb_stream.c 030a07e441296c Karsten Wiese 2008-07-30 664 030a07e441296c Karsten Wiese 2008-07-30 665 int usb_stream_start(struct usb_stream_kernel *sk) 030a07e441296c Karsten Wiese 2008-07-30 666 { 030a07e441296c Karsten Wiese 2008-07-30 667 struct usb_stream *s = sk->s; 030a07e441296c Karsten Wiese 2008-07-30 @668 int frame = 0, iters = 0; 030a07e441296c Karsten Wiese 2008-07-30 669 int u, err; 030a07e441296c Karsten Wiese 2008-07-30 670 int try = 0; 030a07e441296c Karsten Wiese 2008-07-30 671 030a07e441296c Karsten Wiese 2008-07-30 672 if (s->state != usb_stream_stopped) 030a07e441296c Karsten Wiese 2008-07-30 673 return -EAGAIN; 030a07e441296c Karsten Wiese 2008-07-30 674 030a07e441296c Karsten Wiese 2008-07-30 675 subs_set_complete(sk->inurb, i_capture_start); 030a07e441296c Karsten Wiese 2008-07-30 676 subs_set_complete(sk->outurb, i_playback_start); 030a07e441296c Karsten Wiese 2008-07-30 677 memset(sk->write_page, 0, s->write_size); 030a07e441296c Karsten Wiese 2008-07-30 678 dotry: 030a07e441296c Karsten Wiese 2008-07-30 679 s->insize_done = 0; 030a07e441296c Karsten Wiese 2008-07-30 680 s->idle_insize = 0; 030a07e441296c Karsten Wiese 2008-07-30 681 s->idle_outsize = 0; 030a07e441296c Karsten Wiese 2008-07-30 682 s->sync_packet = -1; 030a07e441296c Karsten Wiese 2008-07-30 683 s->inpacket_head = -1; 030a07e441296c Karsten Wiese 2008-07-30 684 sk->iso_frame_balance = 0; 030a07e441296c Karsten Wiese 2008-07-30 685 ++try; 030a07e441296c Karsten Wiese 2008-07-30 686 for (u = 0; u < 2; u++) { 030a07e441296c Karsten Wiese 2008-07-30 687 struct urb *inurb = sk->inurb[u]; 030a07e441296c Karsten Wiese 2008-07-30 688 struct urb *outurb = sk->outurb[u]; 4c0a58ef36f3de Takashi Iwai 2021-05-17 689 030a07e441296c Karsten Wiese 2008-07-30 690 playback_prep_freqn(sk, outurb); 030a07e441296c Karsten Wiese 2008-07-30 691 inurb->number_of_packets = outurb->number_of_packets; 030a07e441296c Karsten Wiese 2008-07-30 692 inurb->transfer_buffer_length = 030a07e441296c Karsten Wiese 2008-07-30 693 inurb->number_of_packets * 030a07e441296c Karsten Wiese 2008-07-30 694 inurb->iso_frame_desc[0].length; d0f3a2eb906256 Karsten Wiese 2011-12-30 695 030a07e441296c Karsten Wiese 2008-07-30 696 if (u == 0) { 030a07e441296c Karsten Wiese 2008-07-30 697 int now; 030a07e441296c Karsten Wiese 2008-07-30 698 struct usb_device *dev = inurb->dev; 4c0a58ef36f3de Takashi Iwai 2021-05-17 699 030a07e441296c Karsten Wiese 2008-07-30 700 frame = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 701 do { 030a07e441296c Karsten Wiese 2008-07-30 702 now = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 703 ++iters; 030a07e441296c Karsten Wiese 2008-07-30 704 } while (now > -1 && now == frame); 030a07e441296c Karsten Wiese 2008-07-30 705 } 030a07e441296c Karsten Wiese 2008-07-30 706 err = usb_submit_urb(inurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 707 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 708 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 709 "%s: usb_submit_urb(sk->inurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 710 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 711 return err; 030a07e441296c Karsten Wiese 2008-07-30 712 } 030a07e441296c Karsten Wiese 2008-07-30 713 err = usb_submit_urb(outurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 714 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 715 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 716 "%s: usb_submit_urb(sk->outurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 717 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 718 return err; 030a07e441296c Karsten Wiese 2008-07-30 719 } d0f3a2eb906256 Karsten Wiese 2011-12-30 720 030a07e441296c Karsten Wiese 2008-07-30 721 if (inurb->start_frame != outurb->start_frame) { f8466d91f36d97 Takashi Iwai 2024-08-07 722 dev_dbg(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 723 "%s: u[%i] start_frames differ in:%u out:%u\n", f8466d91f36d97 Takashi Iwai 2024-08-07 724 __func__, u, inurb->start_frame, outurb->start_frame); 030a07e441296c Karsten Wiese 2008-07-30 725 goto check_retry; 030a07e441296c Karsten Wiese 2008-07-30 726 } 030a07e441296c Karsten Wiese 2008-07-30 727 } 030a07e441296c Karsten Wiese 2008-07-30 728 try = 0; 030a07e441296c Karsten Wiese 2008-07-30 729 check_retry: 030a07e441296c Karsten Wiese 2008-07-30 730 if (try) { 030a07e441296c Karsten Wiese 2008-07-30 731 usb_stream_stop(sk); 030a07e441296c Karsten Wiese 2008-07-30 732 if (try < 5) { 030a07e441296c Karsten Wiese 2008-07-30 733 msleep(1500); f8466d91f36d97 Takashi Iwai 2024-08-07 734 dev_dbg(&sk->dev->dev, "goto dotry;\n"); 030a07e441296c Karsten Wiese 2008-07-30 735 goto dotry; 030a07e441296c Karsten Wiese 2008-07-30 736 } f8466d91f36d97 Takashi Iwai 2024-08-07 737 dev_warn(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 738 "%s: couldn't start all urbs on the same start_frame.\n", f8466d91f36d97 Takashi Iwai 2024-08-07 739 __func__); 030a07e441296c Karsten Wiese 2008-07-30 740 return -EFAULT; 030a07e441296c Karsten Wiese 2008-07-30 741 } 030a07e441296c Karsten Wiese 2008-07-30 742 030a07e441296c Karsten Wiese 2008-07-30 743 sk->idle_inurb = sk->inurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 744 sk->idle_outurb = sk->outurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 745 sk->completed_inurb = sk->inurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 746 sk->completed_outurb = sk->outurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 747 :::::: The code at line 668 was first introduced by commit :::::: 030a07e441296c372f946cd4065b5d831d8dc40c ALSA: Add USB US122L driver :::::: TO: Karsten Wiese <fzu(a)wemgehoertderstaat.de> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 69/82] sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared; did you mean 'p_dev'?
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/82] ALSA: vxpocket: Use standard print API config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090110.t0mWbTyh-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': sound/pcmcia/vx/vxpocket.c:207:13: error: 'struct vx_core' has no member named 'dev' 207 | chip->dev = &link->dev; | ^~ In file included from include/linux/device.h:15, from include/sound/core.h:10, from sound/pcmcia/vx/vxpocket.c:13: sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_probe': >> sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared (first use in this function); did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ sound/pcmcia/vx/vxpocket.c:273:26: note: each undeclared identifier is reported only once for each function it appears in 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ALSA: snd-usb-caiaq: use snd_pcm_rate_to_rate_bit
by Jerome Brunet 08 Aug '24

08 Aug '24

Use snd_pcm_rate_to_rate_bit() helper provided by Alsa instead re-implementing it. This reduce code duplication and helps when changing some Alsa definition is necessary. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/usb/caiaq/audio.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4981753652a7..e62a4ea1d19c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -174,14 +174,6 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) return 0; } -/* this should probably go upstream */ -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 -#error "Change this table" -#endif - -static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; - static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { int bytes_per_sample, bpp, ret, i; @@ -233,10 +225,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ - for (i=0; i < ARRAY_SIZE(rates); i++) - if (runtime->rate == rates[i]) - cdev->pcm_info.rates = 1 << i; - + cdev->pcm_info.rates = snd_pcm_rate_to_rate_bit(runtime->rate); snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; -- 2.43.0

2 1

[tiwai-sound:for-next 32/79] sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev'
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b88d0890ed9adab413ea991fac90842688e1017 commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/79] ALSA: vx_core: Drop unused dev field config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090016.kW0TA6fc-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': >> sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev' 208 | chip->dev = &link->dev; | ^~ vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 08 Aug '24

08 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support" Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: SOF: Intel: add initial support for PTL ASoC: Intel: soc-acpi: add PTL match tables include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 121 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 +++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 ++- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 9

[PATCH 2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 07 Aug '24

07 Aug '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a51..757ade6373ed 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1095,6 +1095,11 @@ int cs35l56_system_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_resume); +static int cs35l56_control_add_nop(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl) +{ + return 0; +} + static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) { struct wm_adsp *dsp; @@ -1117,6 +1122,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp->fw = 12; dsp->wmfw_optional = true; + /* + * None of the firmware controls need to be exported so add a no-op + * callback that suppresses creating an ALSA control. + */ + dsp->control_add = &cs35l56_control_add_nop; + dev_dbg(cs35l56->base.dev, "DSP system name: '%s'\n", dsp->system_name); ret = wm_halo_init(dsp); -- 2.43.0

1 0

[PATCH alsa-lib] seq: Add snd_seq_{get|set}_ump_is_midi1() API functions
by Takashi Iwai 07 Aug '24

07 Aug '24

Implement the API function calls corresponding to the new sequencer port flag bit that has been added recently to the kernel. A UMP MIDI 2.0 device allow to have an optional MIDI 1.0 port while speaking in MIDI 2.0 protocol for other UMP Groups. The new seq port flag indicates that. This is rather a minor difference, and since ALSA sequencer core covers the all conversions, usually you don't have to worry about it at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 2 ++ include/sound/uapi/asequencer.h | 2 ++ src/Versions.in.in | 2 ++ src/seq/seq.c | 29 +++++++++++++++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/seq.h b/include/seq.h index 4f03df438288..2eee95a643b4 100644 --- a/include/seq.h +++ b/include/seq.h @@ -318,6 +318,7 @@ int snd_seq_port_info_get_timestamp_real(const snd_seq_port_info_t *info); int snd_seq_port_info_get_timestamp_queue(const snd_seq_port_info_t *info); int snd_seq_port_info_get_direction(const snd_seq_port_info_t *info); int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info); +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info); void snd_seq_port_info_set_client(snd_seq_port_info_t *info, int client); void snd_seq_port_info_set_port(snd_seq_port_info_t *info, int port); @@ -334,6 +335,7 @@ void snd_seq_port_info_set_timestamp_real(snd_seq_port_info_t *info, int realtim void snd_seq_port_info_set_timestamp_queue(snd_seq_port_info_t *info, int queue); void snd_seq_port_info_set_direction(snd_seq_port_info_t *info, int direction); void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group); +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1); int snd_seq_create_port(snd_seq_t *handle, snd_seq_port_info_t *info); int snd_seq_delete_port(snd_seq_t *handle, int port); diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index 923dfdddfc76..b3e9df5894af 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -477,6 +477,8 @@ struct snd_seq_remove_events { #define SNDRV_SEQ_PORT_FLG_TIMESTAMP (1<<1) #define SNDRV_SEQ_PORT_FLG_TIME_REAL (1<<2) +#define SNDRV_SEQ_PORT_FLG_IS_MIDI1 (1<<3) /* Keep MIDI 1.0 protocol */ + /* port direction */ #define SNDRV_SEQ_PORT_DIR_UNKNOWN 0 #define SNDRV_SEQ_PORT_DIR_INPUT 1 diff --git a/src/Versions.in.in b/src/Versions.in.in index 90849277c983..298e610c2099 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -209,6 +209,8 @@ ALSA_1.2.13 { @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; @SYMBOL_PREFIX@snd_seq_has_tempo_base; + @SYMBOL_PREFIX@snd_seq_port_info_get_ump_is_midi1; + @SYMBOL_PREFIX@snd_seq_port_info_set_ump_is_midi1; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index d0ecedf0838f..347ff45544fa 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -2431,6 +2431,19 @@ int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info) return info->ump_group; } +/** + * \brief Get the status of the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \return 1 if it's an optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_set_ump_is_midi1() + */ +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info) +{ + assert(info); + return !!(info->flags & SNDRV_SEQ_PORT_FLG_IS_MIDI1); +} + /** * \brief Set the client id of a port_info container * \param info port_info container @@ -2635,6 +2648,22 @@ void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group) info->ump_group = ump_group; } +/** + * \brief Set the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \param is_midi1 non-zero for MIDI 1.0 port in MIDI 2.0 EP + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_get_ump_is_midi1() + */ +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1) +{ + assert(info); + if (is_midi1) + info->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; + else + info->flags &= ~SNDRV_SEQ_PORT_FLG_IS_MIDI1; +} + /** * \brief create a sequencer port on the current client * \param seq sequencer handle -- 2.43.0

1 0

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 07 Aug '24

07 Aug '24

Hi, > Based on the inputs/suggestions from Tudor, i am planning to add a new > layer between the SPI-NOR and MTD layers to support stacked and parallel > configurations. This new layer will be part of the spi-nor and located in > mtd/spi-nor/ Will AMD submit to maintain this layer? What happens if the maintainer will leave AMD? TBH, personally, I don't like to maintain such a niche feature. I'd really like to see some use cases and performance reports for this, like actual boards (and no evaluation boards don't count). Why wouldn't someone just use an octal flash? And as already mentioned there is also mtdcat, which seems to duplicate some features? -michael

5 13

[PATCH v1] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 07 Aug '24

07 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 49bd7097d..4dc3350d5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,21 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; - int i, j, rc; + int i, j, rc, data; + int offset = 0; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add Framework Laptop 13 (Intel Core Ultra) to quirks
by Dustin L. Howett 07 Aug '24

07 Aug '24

The Framework Laptop 13 (Intel Core Ultra) has an ALC285 that ships in a similar configuration to the ALC295 in previous models. It requires the same quirk for headset detection. --- Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1645d21d422f..480e82df7a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10678,6 +10678,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. --- base-commit: de9c2c66ad8e787abec7c9d7eff4f8c3cdd28aed change-id: 20240606-alsa-hda-realtek-add-framework-laptop-13-intel-core-ultra-to-quirks-2af5081e2f2c Best regards, -- Dustin L. Howett <dustin(a)howett.net>

2 1

[PATCH v2] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest asoc/for-6.12 branch include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9ddc..1edcb8d6f6eea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2388,14 +2388,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2525,26 +2517,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", snd_pcm_direction_name(stream), fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2558,10 +2536,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2605,23 +2580,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

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Re: [PATCH 0/2] ASoC: Extend wm_adsp so cs35l56 can suppress controls
by Mark Brown 06 Aug '24

06 Aug '24

On Mon, 05 Aug 2024 10:27:19 +0000, Simon Trimmer wrote: > This pair of patches extend wm_adsp to add a callback that can be used > to control whether ALSA controls are added and then tweak cs35l56 to use > it to suppress controls made from firmware coefficients. > > Simon Trimmer (2): > ASoC: wm_adsp: Add control_add callback and export > wm_adsp_control_add() > ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() commit: 45b4acab4cac79503663f0a4be9eb3752db04d4b [2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients commit: 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de 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

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[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

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[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 122 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

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[PATCH] ALSA: usb-audio: Add input gain and master output mixer controls for RME Babyface Pro
by Stefan Stistrup 06 Aug '24

06 Aug '24

Add missing input gain and master output mixer controls for RME Babyface Pro This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup <sstistrup(a)gmail.com> --- sound/usb/mixer_quirks.c | 162 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 161 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d771..26298e06b838 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,113 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + } else { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + } + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2906,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2938,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2997,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) { + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + } else { + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + } + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, -- 2.45.2

2 2

[RFC PATCH 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Aug '24

06 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. Shengjiu Wang (6): ALSA: compress: add Sample Rate Converter codec support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 + include/uapi/sound/compress_params.h | 9 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 769 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 10 files changed, 1284 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH v1] ASoC: meson: axg-fifo: set option to use raw spinlock
by Arseniy Krasnov 06 Aug '24

06 Aug '24

Raw spinlock is needed here, because with enabled PREEMPT_RT, spinlock_t become preemptible, but this regmap lock could be acquired in IRQ handler. Found by lockdep: [ ] ============================= [ ] [ BUG: Invalid wait context ] [ ] 6.9.9-sdkernel #1 Tainted: G O [ ] ----------------------------- [ ] aplay/413 is trying to lock: [ ] ffff000003930018 (axg_fifo:356:(&axg_fifo_regmap_cfg)->lock){....}-{3:3},c [ ] other info that might help us debug this: [ ] context-{2:2} [ ] no locks held by aplay/413. [ ] stack backtrace: [ ] CPU: 0 PID: 413 Comm: aplay Tainted: G O 6.9.9-kernel #1 [ ] Hardware name: SberDevices SberBoom Mini (DT) [ ] Call trace: [ ] dump_backtrace+0x98/0xf0 [ ] show_stack+0x18/0x24 [ ] dump_stack_lvl+0x90/0xd0 [ ] dump_stack+0x18/0x24 [ ] __lock_acquire+0x9dc/0x1f10 [ ] lock_acquire.part.0+0xe8/0x228 [ ] lock_acquire+0x68/0x84 [ ] _raw_spin_lock_irqsave+0x60/0x88 [ ] regmap_lock_spinlock+0x18/0x2c [ ] regmap_read+0x3c/0x78 [ ] axg_fifo_pcm_irq_block+0x4c/0xc8 [ ] __handle_irq_event_percpu+0xa4/0x2f8 [ ] handle_irq_event+0x4c/0xbc [ ] handle_fasteoi_irq+0xa4/0x23c [ ] generic_handle_domain_irq+0x2c/0x44 [ ] gic_handle_irq+0x40/0xc4 [ ] call_on_irq_stack+0x24/0x4c [ ] do_interrupt_handler+0x80/0x84 [ ] el0_interrupt+0x5c/0x124 [ ] __el0_irq_handler_common+0x18/0x24 [ ] el0t_32_irq_handler+0x10/0x1c [ ] el0t_32_irq+0x194/0x198 Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-fifo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index ecb3eb7a9723d..a22298f74b35a 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -328,6 +328,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .val_bits = 32, .reg_stride = 4, .max_register = FIFO_CTRL2, + .use_raw_spinlock = true, }; int axg_fifo_probe(struct platform_device *pdev) -- 2.30.1

4 12

[PATCH] ASoC: codecs: lpass-macro: fix missing codec version
by Johan Hovold 05 Aug '24

05 Aug '24

Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- Cc: regressions(a)lists.linux.dev #regzbot introduced: 378918d59181 sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) -- 2.44.2

3 2