November 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: soc-core: add the driver component name to the component struc
by Kory Maincent 18 Nov '21

18 Nov '21

The function fmt_single_name is using to sanitize namings, and it return the name set to component->name. This function is returning either the "device_name" or the "device_name.driver_name" for i2c devices. There is no use of the component driver name. If a non i2c driver register two components the function will return the same "device_name" for both components. This could cause unexpected issue, in my case it is a debugfs error which tries to create two directory with the same component name. I have fixed it by prefixing the component name with the driver component name. Signed-off-by: Kory Maincent <kory.maincent(a)bootlin.com> --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcf6be4c4aaa..21ff77b231b8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2342,10 +2342,10 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, const char *snd_drv_name, int *id) { const char *devname = dev_name(dev); - char *found, *name; + char *found, *name, *tmp; unsigned int id1, id2; if (devname == NULL) @@ -2380,6 +2380,14 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (snd_drv_name != NULL) { + /* Add driver component name if present */ + tmp = devm_kasprintf(dev, GFP_KERNEL, "%s.%s", snd_drv_name, name); + devm_kfree(dev, name); + name = devm_kstrdup(dev, tmp, GFP_KERNEL); + devm_kfree(dev, tmp); + } + return name; } @@ -2444,7 +2452,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, dai_drv->name, &dai->id); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2578,7 +2586,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, driver->name, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.25.1

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[PATCH 0/3] ASoC: stm32: add pm runtime support
by Olivier Moysan 18 Nov '21

18 Nov '21

Enable support of pm runtime on STM32 SPDIFRX, I2S and DFSDM drivers to allow power state monitoring. Olivier Moysan (3): ASoC: stm32: i2s: add pm_runtime support ASoC: stm32: dfsdm: add pm_runtime support for audio ASoC: stm32: spdifrx: add pm_runtime support sound/soc/stm/stm32_adfsdm.c | 11 ++++++++--- sound/soc/stm/stm32_i2s.c | 4 ++++ sound/soc/stm/stm32_spdifrx.c | 4 ++++ 3 files changed, 16 insertions(+), 3 deletions(-) -- 2.17.1

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alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HDA Intel at 0xfa9f8000 irq 47)
by GitHub issues - opened 18 Nov '21

18 Nov '21

alsa-project/alsa-utils issue #120 was opened from svarshavchik: I'm using alsa-utils 1.2.5.1 on Fedora 35 (alsa-utils-1.2.5.1-2.fc35.x86_64) There are two audio devices, built-in audio and a USB audio device. After a reboot the default audio output gets reset to the USB audio service: $ pactl get-default-sink alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo After resetting it back to built-in audio: $ pactl set-default-sink alsa_output.pci-0000_00_1b.0.analog-stereo $ pactl get-default-sink alsa_output.pci-0000_00_1b.0.analog-stereo This persists until the next reboot, when it gets reset back to the other device. While perusing system boot messages the following caught my eye: Nov 18 07:05:58 monster.email-scan.com systemd[1]: Started Manage Sound Card State (restore and store). Nov 18 07:05:58 monster.email-scan.com alsactl[3870]: alsactl 1.2.5.1 daemon started Nov 18 07:05:58 monster.email-scan.com alsactl[3870]: alsa-lib parser.c:242:(error_node) UCM is not supported for this HDA model (HDA Intel at 0xfa9f8000 irq 47) Nov 18 07:05:58 monster.email-scan.com alsactl[3870]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 Nov 18 07:05:58 monster.email-scan.com alsactl[3870]: alsa-lib main.c:1405:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -2 The fedora package starts alsa like this: /usr/sbin/alsactl -s -n 19 -c -E ALSA_CONFIG_PATH=/etc/alsa/alsactl.conf --initfile=/lib/alsa/init/00main rdaemon and stops it like this: /usr/sbin/alsactl -s kill save_and_quit Stopping/restarting logs the same errors again. Not sure this is related to the default audio device getting reset on boot. Issue URL : https://github.com/alsa-project/alsa-utils/issues/120 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2] ASoC: codecs: MBHC: Add support for special headset
by Srinivasa Rao Mandadapu 18 Nov '21

18 Nov '21

Update MBHC driver to support special headset such as apple and huwawei headsets. Changes Since V1: -- Fix typo errors. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Co-developed-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd-mbhc-v2.c | 75 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 71 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 405128c..d6545e4 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1022,6 +1022,56 @@ static int wcd_mbhc_get_plug_from_adc(struct wcd_mbhc *mbhc, int adc_result) return plug_type; } +static int wcd_mbhc_get_spl_hs_thres(struct wcd_mbhc *mbhc) +{ + int hs_threshold, micbias_mv; + + micbias_mv = wcd_mbhc_get_micbias(mbhc); + if (mbhc->cfg->hs_thr && mbhc->cfg->micb_mv != WCD_MBHC_ADC_MICBIAS_MV) { + if (mbhc->cfg->micb_mv == micbias_mv) + hs_threshold = mbhc->cfg->hs_thr; + else + hs_threshold = (mbhc->cfg->hs_thr * micbias_mv) / mbhc->cfg->micb_mv; + } else { + hs_threshold = ((WCD_MBHC_ADC_HS_THRESHOLD_MV * micbias_mv) / + WCD_MBHC_ADC_MICBIAS_MV); + } + return hs_threshold; +} + +static bool wcd_mbhc_check_for_spl_headset(struct wcd_mbhc *mbhc) +{ + bool is_spl_hs = false; + int output_mv, hs_threshold, hph_threshold; + + if (!mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic) + return false; + + /* Bump up MIC_BIAS2 to 2.7V */ + mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic(mbhc->component, MIC_BIAS_2, true); + usleep_range(10000, 10100); + + output_mv = wcd_measure_adc_once(mbhc, MUX_CTL_IN2P); + hs_threshold = wcd_mbhc_get_spl_hs_thres(mbhc); + hph_threshold = wcd_mbhc_adc_get_hph_thres(mbhc); + + if (output_mv > hs_threshold || output_mv < hph_threshold) { + if (mbhc->force_linein == true) + is_spl_hs = false; + } else { + is_spl_hs = true; + } + + /* Back MIC_BIAS2 to 1.8v if the type is not special headset */ + if (!is_spl_hs) { + mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic(mbhc->component, MIC_BIAS_2, false); + /* Add 10ms delay for micbias to settle */ + usleep_range(10000, 10100); + } + + return is_spl_hs; +} + static void wcd_correct_swch_plug(struct work_struct *work) { struct wcd_mbhc *mbhc; @@ -1029,12 +1079,14 @@ static void wcd_correct_swch_plug(struct work_struct *work) enum wcd_mbhc_plug_type plug_type = MBHC_PLUG_TYPE_INVALID; unsigned long timeout; int pt_gnd_mic_swap_cnt = 0; - int output_mv, cross_conn, hs_threshold, try = 0; + int output_mv, cross_conn, hs_threshold, try = 0, micbias_mv; + bool is_spl_hs = false; bool is_pa_on; mbhc = container_of(work, struct wcd_mbhc, correct_plug_swch); component = mbhc->component; + micbias_mv = wcd_mbhc_get_micbias(mbhc); hs_threshold = wcd_mbhc_adc_get_hs_thres(mbhc); /* Mask ADC COMPLETE interrupt */ @@ -1097,6 +1149,16 @@ static void wcd_correct_swch_plug(struct work_struct *work) plug_type = wcd_mbhc_get_plug_from_adc(mbhc, output_mv); is_pa_on = wcd_mbhc_read_field(mbhc, WCD_MBHC_HPH_PA_EN); + if ((output_mv > hs_threshold) && (!is_spl_hs)) { + is_spl_hs = wcd_mbhc_check_for_spl_headset(mbhc); + output_mv = wcd_measure_adc_once(mbhc, MUX_CTL_IN2P); + + if (is_spl_hs) { + hs_threshold = (hs_threshold * wcd_mbhc_get_micbias(mbhc)) / + micbias_mv; + } + } + if ((output_mv <= hs_threshold) && !is_pa_on) { /* Check for cross connection*/ cross_conn = wcd_check_cross_conn(mbhc); @@ -1122,14 +1184,19 @@ static void wcd_correct_swch_plug(struct work_struct *work) } } - if (output_mv > hs_threshold) /* cable is extension cable */ + /* cable is extension cable */ + if (output_mv > hs_threshold || mbhc->force_linein == true) plug_type = MBHC_PLUG_TYPE_HIGH_HPH; } wcd_mbhc_bcs_enable(mbhc, plug_type, true); - if (plug_type == MBHC_PLUG_TYPE_HIGH_HPH) - wcd_mbhc_write_field(mbhc, WCD_MBHC_ELECT_ISRC_EN, 1); + if (plug_type == MBHC_PLUG_TYPE_HIGH_HPH) { + if (is_spl_hs) + plug_type = MBHC_PLUG_TYPE_HEADSET; + else + wcd_mbhc_write_field(mbhc, WCD_MBHC_ELECT_ISRC_EN, 1); + } wcd_mbhc_write_field(mbhc, WCD_MBHC_ADC_MODE, 0); wcd_mbhc_write_field(mbhc, WCD_MBHC_ADC_EN, 0); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] ALSA: hda/realtek: Fix LED on HP ProBook 435 G7
by Takashi Iwai 18 Nov '21

18 Nov '21

HP ProBook 435 G7 (SSID 103c:8735) needs the similar quirk as another HP ProBook for enabling the mute and the mic-mute LEDs. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215021 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 701c80ed83dc..9ce7457533c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8665,6 +8665,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), -- 2.31.1

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[PATCH 0/4] New debug feature: IPC message injector
by Daniel Baluta 17 Nov '21

17 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> With the new SND_SOC_SOF_DEBUG_IPC_MSG_INJECTOR it is going to be possible to inject arbitrary messages via the debugfs/sof/ipc_msg_inject file and get the reply from the same file as a binary. The main use of this feature is to stress test the firmware's robustness against maliciously (or erroneous) crafted messages. We also receive firmware crash reports with only a binary of the sent message which caused the firmware crash. Before adding the new debug feature we needed to make sure that the IPC reply handling is working correctly for cases when we don't know beforehand the size of the reply (since we don't know what kind of message we are sending). Peter Ujfalusi (4): ASoC: SOF: utils: Add generic function to get the reply for a tx message ASoC: SOF: imx: Use the generic helper to get the reply ASoC: SOF: intel: Use the generic helper to get the reply ASoC: SOF: debug: Add support for IPC message injection sound/soc/sof/Kconfig | 8 +++ sound/soc/sof/debug.c | 107 ++++++++++++++++++++++++++++++++++ sound/soc/sof/imx/imx8.c | 37 +----------- sound/soc/sof/imx/imx8m.c | 37 +----------- sound/soc/sof/intel/atom.c | 43 +------------- sound/soc/sof/intel/bdw.c | 43 +------------- sound/soc/sof/intel/hda-ipc.c | 29 +-------- sound/soc/sof/ipc.c | 61 +++++++++++++++++++ sound/soc/sof/sof-priv.h | 10 ++++ 9 files changed, 192 insertions(+), 183 deletions(-) -- 2.27.0

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[PATCH v2 0/4] ASoC: cs42l42: Remove redundant code
by Richard Fitzgerald 17 Nov '21

17 Nov '21

- Remove field writes in regmap_update_bits() calls where the code intends that the field value is not changed. - Remove unnecessary complexity from reporting jack unplug event - Remove a PLL config value member that was introduced in a bugfix but made redundant by a later bugfix. Richard Fitzgerald (4): ASoC: cs42l42: Remove redundant writes to DETECT_MODE ASoC: cs42l42: Remove redundant writes to RS_PLUG/RS_UNPLUG masks ASoC: cs42l42: Simplify reporting of jack unplug ASoC: cs42l42: Remove redundant pll_divout member sound/soc/codecs/cs42l42.c | 42 +++++++----------------------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 7 insertions(+), 36 deletions(-) -- 2.11.0

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[PATCH v2 01/10] ASoC: wm_adsp: Remove the wmfw_add_ctl helper function
by Charles Keepax 17 Nov '21

17 Nov '21

The helper function wmfw_add_ctl is only called from one place and that place is a function with only 2 lines of code. Merge the helper function into the work function to simplify the code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 33 +++++++++------------------------ 1 file changed, 9 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d4f0d72cbcc80..404717e30f44d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -537,15 +537,20 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) return out; } -static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) +static void wm_adsp_ctl_work(struct work_struct *work) { + struct wm_coeff_ctl *ctl = container_of(work, + struct wm_coeff_ctl, + work); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + struct wm_adsp *dsp = container_of(cs_ctl->dsp, + struct wm_adsp, + cs_dsp); struct snd_kcontrol_new *kcontrol; - int ret; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); if (!kcontrol) - return -ENOMEM; + return; kcontrol->name = ctl->name; kcontrol->info = wm_coeff_info; @@ -571,29 +576,9 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) break; } - ret = snd_soc_add_component_controls(dsp->component, kcontrol, 1); - if (ret < 0) - goto err_kcontrol; + snd_soc_add_component_controls(dsp->component, kcontrol, 1); kfree(kcontrol); - - return 0; - -err_kcontrol: - kfree(kcontrol); - return ret; -} - -static void wm_adsp_ctl_work(struct work_struct *work) -{ - struct wm_coeff_ctl *ctl = container_of(work, - struct wm_coeff_ctl, - work); - struct wm_adsp *dsp = container_of(ctl->cs_ctl->dsp, - struct wm_adsp, - cs_dsp); - - wmfw_add_ctl(dsp, ctl); } static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) -- 2.11.0

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[PATCH] ASoC: stm32: sai: increase channels_max limit
by Olivier Moysan 17 Nov '21

17 Nov '21

From: Olivier Moysan <olivier.moysan(a)st.com> The SAI peripheral supports up to 16 channels in TDM mode (8L+8R). The driver currently supports TDM over two channels. Increase SAI DAI playback/record channels_max, to also allow up to 16 channels in TDM mode. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 9c3b8e209656..95cd38a502bb 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1294,7 +1294,7 @@ static struct snd_soc_dai_driver stm32_sai_playback_dai = { .id = 1, /* avoid call to fmt_single_name() */ .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 16, .rate_min = 8000, .rate_max = 192000, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -1312,7 +1312,7 @@ static struct snd_soc_dai_driver stm32_sai_capture_dai = { .id = 1, /* avoid call to fmt_single_name() */ .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 16, .rate_min = 8000, .rate_max = 192000, .rates = SNDRV_PCM_RATE_CONTINUOUS, -- 2.17.1

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[PATCH] ASoC: stm32: i2s: fix 32 bits channel length without mclk
by Olivier Moysan 17 Nov '21

17 Nov '21

Fix divider calculation in the case of 32 bits channel configuration, when no master clock is used. Fixes: e4e6ec7b127c ("ASoC: stm32: Add I2S driver") Signed-off-by: Olivier Moysan <olivier.moysan(a)foss.st.com> --- sound/soc/stm/stm32_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 6254bacad6eb..717f45a83445 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -700,7 +700,7 @@ static int stm32_i2s_configure_clock(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - nb_bits = frame_len * ((cgfr & I2S_CGFR_CHLEN) + 1); + nb_bits = frame_len * (FIELD_GET(I2S_CGFR_CHLEN, cgfr) + 1); ret = stm32_i2s_calc_clk_div(i2s, i2s_clock_rate, (nb_bits * rate)); if (ret) -- 2.17.1

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