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

Why Choose a Dedicated Server for Your Business?
by David Menk 29 Oct '24

29 Oct '24

Hello everyone, I wanted to share some insights on why dedicated server hosting can be a game-changer for businesses, especially those with high-traffic websites or specific performance needs. What is a Dedicated Server? A dedicated server is a physical server that's exclusively used by one client. Unlike shared hosting, where resources are split between multiple users, a dedicated server gives you complete control over CPU, RAM, storage, and bandwidth, making it ideal for websites or applications that need consistent performance, security, and reliability. Benefits of Using a Dedicated Server: Performance and Speed: With dedicated resources, your website or application can handle high volumes of traffic without slowing down. This is especially important for e-commerce sites or any business where speed impacts the user experience. Enhanced Security: Dedicated servers offer stronger security, as you're not sharing the server with other users. You can implement custom security measures tailored to your business, helping to protect against potential breaches and data loss. Full Control and Customization: Since the server is dedicated to you, you have complete control over the operating system, software, and applications you want to run. This level of customization can be essential for companies with specific software requirements. Scalability: As your business grows, dedicated servers can be upgraded with additional resources to meet your increasing demands, providing flexibility without the need to migrate to another server. Better Reliability: Dedicated servers are less prone to outages or downtime compared to shared hosting, providing a stable environment that’s crucial for businesses that rely on 24/7 availability. Is a Dedicated Server Right for You? If your website or application has high traffic, complex applications, or sensitive data, a dedicated server could be the right choice. It's a bit more of an investment, but the control, performance, and reliability often pay off in the long run. If anyone has experience with dedicated servers or is considering making the switch, I’d love to hear your thoughts and any advice you might have! Looking forward to the discussion!

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[PATCH] ALSA: usb-audio: Add Pioneer DJ/AlphaTheta DJM-A9 Mixer
by Sarah Grant 29 Oct '24

29 Oct '24

This also includes a change to the capture level as they differ with this device, as well as additional values for prefader channels. Acked-by: Olivia Mackintosh <livvy(a)base.nu> Signed-off-by: Sarah Grant <s(a)srd.tw> --- sound/usb/mixer_quirks.c | 68 +++++++++++++++++++++++++++++++++++++--- sound/usb/quirks-table.h | 57 +++++++++++++++++++++++++++++++++ 2 files changed, 121 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2a9594f34dac..15c300eddf53 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3498,7 +3498,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) } /* - * Pioneer DJ DJM Mixers + * Pioneer DJ / AlphaTheta DJM Mixers * * These devices generally have options for soft-switching the playback and * capture sources in addition to the recording level. Although different @@ -3515,17 +3515,23 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_CAP_CDLINE 0x01 #define SND_DJM_CAP_DIGITAL 0x02 #define SND_DJM_CAP_PHONO 0x03 +#define SND_DJM_CAP_PREFADER 0x05 #define SND_DJM_CAP_PFADER 0x06 #define SND_DJM_CAP_XFADERA 0x07 #define SND_DJM_CAP_XFADERB 0x08 #define SND_DJM_CAP_MIC 0x09 #define SND_DJM_CAP_AUX 0x0d #define SND_DJM_CAP_RECOUT 0x0a +#define SND_DJM_CAP_RECOUT_NOMIC 0x0e #define SND_DJM_CAP_NONE 0x0f #define SND_DJM_CAP_CH1PFADER 0x11 #define SND_DJM_CAP_CH2PFADER 0x12 #define SND_DJM_CAP_CH3PFADER 0x13 #define SND_DJM_CAP_CH4PFADER 0x14 +#define SND_DJM_CAP_CH1PREFADER 0x31 +#define SND_DJM_CAP_CH2PREFADER 0x32 +#define SND_DJM_CAP_CH3PREFADER 0x33 +#define SND_DJM_CAP_CH4PREFADER 0x34 // Playback types #define SND_DJM_PB_CH1 0x00 @@ -3551,6 +3557,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_900NXS2_IDX 0x3 #define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_450_IDX 0x5 +#define SND_DJM_A9_IDX 0x6 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3579,7 +3586,7 @@ struct snd_djm_ctl { u16 wIndex; }; -static const char *snd_djm_get_label_caplevel(u16 wvalue) +static const char *snd_djm_get_label_caplevel_common(u16 wvalue) { switch (wvalue) { case 0x0000: return "-19dB"; @@ -3590,6 +3597,20 @@ static const char *snd_djm_get_label_caplevel(u16 wvalue) } }; +// The DJM-A9 has different capture levels than other, older models +static const char *snd_djm_get_label_caplevel_a9(u16 wvalue) +{ + switch (wvalue) { + case 0x0000: return "+15dB"; + case 0x0100: return "+12dB"; + case 0x0200: return "+9dB"; + case 0x0300: return "+6dB"; + case 0x0400: return "+3dB"; + case 0x0500: return "0dB"; + default: return NULL; + } +}; + static const char *snd_djm_get_label_cap_common(u16 wvalue) { switch (wvalue & 0x00ff) { @@ -3602,8 +3623,13 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_XFADERB: return "Cross Fader B"; case SND_DJM_CAP_MIC: return "Mic"; case SND_DJM_CAP_RECOUT: return "Rec Out"; + case SND_DJM_CAP_RECOUT_NOMIC: return "Rec Out without Mic"; case SND_DJM_CAP_AUX: return "Aux"; case SND_DJM_CAP_NONE: return "None"; + case SND_DJM_CAP_CH1PREFADER: return "Pre Fader Ch1"; + case SND_DJM_CAP_CH2PREFADER: return "Pre Fader Ch2"; + case SND_DJM_CAP_CH3PREFADER: return "Pre Fader Ch3"; + case SND_DJM_CAP_CH4PREFADER: return "Pre Fader Ch4"; case SND_DJM_CAP_CH1PFADER: return "Post Fader Ch1"; case SND_DJM_CAP_CH2PFADER: return "Post Fader Ch2"; case SND_DJM_CAP_CH3PFADER: return "Post Fader Ch3"; @@ -3623,6 +3649,14 @@ static const char *snd_djm_get_label_cap_850(u16 wvalue) } }; +static const char *snd_djm_get_label_caplevel(u8 device_idx, u16 wvalue) +{ + switch (device_idx) { + case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_a9(wvalue); + default: return snd_djm_get_label_caplevel_common(wvalue); + } +}; + static const char *snd_djm_get_label_cap(u8 device_idx, u16 wvalue) { switch (device_idx) { @@ -3644,7 +3678,7 @@ static const char *snd_djm_get_label_pb(u16 wvalue) static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) { switch (windex) { - case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(wvalue); + case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(device_idx, wvalue); case SND_DJM_WINDEX_CAP: return snd_djm_get_label_cap(device_idx, wvalue); case SND_DJM_WINDEX_PB: return snd_djm_get_label_pb(wvalue); default: return NULL; @@ -3653,7 +3687,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300 }; + 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; // DJM-250MK2 @@ -3795,6 +3829,28 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { }; +// DJM-A9 +static const u16 snd_djm_opts_a9_cap1[] = { + 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, + 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; +static const u16 snd_djm_opts_a9_cap2[] = { + 0x0201, 0x0202, 0x0203, 0x0205, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a, 0x020e }; +static const u16 snd_djm_opts_a9_cap3[] = { + 0x0301, 0x0302, 0x0303, 0x0305, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a, 0x030e }; +static const u16 snd_djm_opts_a9_cap4[] = { + 0x0401, 0x0402, 0x0403, 0x0405, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a, 0x040e }; +static const u16 snd_djm_opts_a9_cap5[] = { + 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; + +static const struct snd_djm_ctl snd_djm_ctls_a9[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", a9_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", a9_cap5, 2, SND_DJM_WINDEX_CAP) +}; + static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), @@ -3802,6 +3858,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), + [SND_DJM_A9_IDX] = SND_DJM_DEVICE(a9), }; @@ -4076,6 +4133,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ err = snd_djm_controls_create(mixer, SND_DJM_900NXS2_IDX); break; + case USB_ID(0x2b73, 0x003c): /* Pioneer DJ / AlphaTheta DJM-A9 */ + err = snd_djm_controls_create(mixer, SND_DJM_A9_IDX); + break; } return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 24c981c9b240..b758655b948c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3119,6 +3119,63 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * Pioneer DJ / AlphaTheta DJM-A9 + * 10 channels playback & 12 channels capture @ 44.1/48/96kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x003c), + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + QUIRK_COMPOSITE_END + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * -- 2.40.1

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[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 29 Oct '24

29 Oct '24

Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

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[PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 29 Oct '24

29 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… 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. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format 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_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH] ASoC: codecs: ES8326: Modify the configuration of and micbias
by Zhang Yi 28 Oct '24

28 Oct '24

Because we designed a new version of ES8326, the configuration of micbias needed to be modified.We tested the new driver, on both the new version and the old one. It works well. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..aa3e364827c8 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -614,6 +614,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x0F); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + } } } else { if (!es8326->calibrated) { @@ -640,6 +644,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) ES8326_MUTE_MASK, ~(ES8326_MUTE)); } else { msleep(300); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); + } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x00); } @@ -821,7 +829,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); - if ((es8326->jack_remove_retry == 1) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 1) && (es8326->version < ES8326_VERSION_B)) { if (iface & ES8326_HPINSERT_FLAG) es8326->jack_remove_retry = 2; else @@ -859,7 +867,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ - if ((es8326->jack_remove_retry == 0) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 0) && (es8326->version < ES8326_VERSION_B)) { es8326->jack_remove_retry = 1; dev_dbg(comp->dev, "remove event check, invert HPJACK_POL, cnt = %d\n", es8326->jack_remove_retry); @@ -954,7 +962,7 @@ static int es8326_calibrate(struct snd_soc_component *component) regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); es8326->version = reg; - if ((es8326->version == ES8326_VERSION_B) && (es8326->calibrated == false)) { + if ((es8326->version >= ES8326_VERSION_B) && (es8326->calibrated == false)) { dev_dbg(component->dev, "ES8326_VERSION_B, calibrating\n"); regmap_write(es8326->regmap, ES8326_CLK_INV, 0xc0); regmap_write(es8326->regmap, ES8326_CLK_DIV1, 0x03); @@ -1047,7 +1055,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x80 | - ((es8326->version == ES8326_VERSION_B) ? + ((es8326->version >= ES8326_VERSION_B) ? (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol) : (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol | 0x04))); usleep_range(5000, 10000); @@ -1073,6 +1081,10 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + } msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 28 Oct '24

28 Oct '24

From: pc1598 <piyushchouhan1598(a)gmail.com> Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

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[tiwai-sound:topic/compress-accel 1/1] sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used
by kernel test robot 25 Oct '24

25 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/compress-accel head: 04177158cf98a79744937893b100020d77e6f9ac commit: 04177158cf98a79744937893b100020d77e6f9ac [1/1] ALSA: compress_offload: introduce accel operation mode config: arm-randconfig-001-20241026 (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-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/202410260257.PNGCF9yq-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/compress_offload.c: In function 'snd_compr_poll': >> sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] 409 | struct snd_compr_runtime *runtime; | ^~~~~~~ vim +/runtime +409 sound/core/compress_offload.c 404 405 static __poll_t snd_compr_poll(struct file *f, poll_table *wait) 406 { 407 struct snd_compr_file *data = f->private_data; 408 struct snd_compr_stream *stream; > 409 struct snd_compr_runtime *runtime; 410 size_t avail; 411 __poll_t retval = 0; 412 413 if (snd_BUG_ON(!data)) 414 return EPOLLERR; 415 416 stream = &data->stream; 417 runtime = stream->runtime; 418 419 guard(mutex)(&stream->device->lock); 420 421 switch (stream->runtime->state) { 422 case SNDRV_PCM_STATE_OPEN: 423 case SNDRV_PCM_STATE_XRUN: 424 return snd_compr_get_poll(stream) | EPOLLERR; 425 default: 426 break; 427 } 428 429 poll_wait(f, &stream->runtime->sleep, wait); 430 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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Help to solve pop when sound loops
by Charles Mulder 25 Oct '24

25 Oct '24

Hi. I hope this is the correct place to ask for help relating to ALSA's C lib? I'm working on a wavetable project for college. Please see attached. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on buffer > period > frames, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C -- Sent with Tuta; enjoy secure & ad-free emails: https://tuta.com

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Help to solve pop when sound loops
by GitHub issues - edited 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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Help to address pop at end of buffer
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was opened from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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