July 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH][next] selftests: ALSA: Fix fclose on an already fclosed file pointer
by Colin Ian King 13 Jul '23

13 Jul '23

In the case where a sysfs file cannot be opened the error return path fcloses file pointer fpl, however, fpl has already been closed in the previous stanza. Fix the double fclose by removing it. Fixes: 10b98a4db11a ("selftests: ALSA: Add test for the 'pcmtest' driver") Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- tools/testing/selftests/alsa/test-pcmtest-driver.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/test-pcmtest-driver.c b/tools/testing/selftests/alsa/test-pcmtest-driver.c index 71931b240a83..357adc722cba 100644 --- a/tools/testing/selftests/alsa/test-pcmtest-driver.c +++ b/tools/testing/selftests/alsa/test-pcmtest-driver.c @@ -47,10 +47,8 @@ static int read_patterns(void) sprintf(pf, "/sys/kernel/debug/pcmtest/fill_pattern%d", i); fp = fopen(pf, "r"); - if (!fp) { - fclose(fpl); + if (!fp) return -1; - } fread(patterns[i].buf, 1, patterns[i].len, fp); fclose(fp); } -- 2.39.2

3 2

[PATCH v4] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,6 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; + int extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +377,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +419,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +441,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +452,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +583,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +657,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +706,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +861,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +952,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1246,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..f56bd7d656e9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH 00/13] PCI: Define Intel PCI IDs and use them in drivers
by Amadeusz Sławiński 12 Jul '23

12 Jul '23

PCI IDs for Intel HDA are duplicated across quite a few drivers, due to various configurations and historical reasons. Currently almost all uses of HDA PCI IDs have corresponding comment telling which platform it is. Additionally there are some inconsistencies between drivers about which ID corresponds to which device. Simplify things, by adding PCI IDs to global header and make use of them in drivers. This allows for removal of comments by having IDs themselves being self explanatory. Additionally it allows for removal of existing inconsistencies by having one source of truth. Changes from RFC: - Sort Intel PCI IDs before adding new ones - Fix ordering of new PCI IDs (Andy) - Define all used Intel IDs (Andy) - Add macros for controller type detection (Andy/Bjorn) - Add set of patches changing to use above macro (Andy/Bjorn) - Use PCI_DEVICE_DATA for Intel IDs in sound/pci/hda/hda_intel.c (Andy) - Commit message wording (Andy) - Remove unnecessary tabs (Andy) Amadeusz Sławiński (13): PCI: Sort Intel PCI IDs by number PCI: Add Intel Audio DSP devices to pci_ids.h ALSA: hda: Add controller matching macros ALSA: hda: Use global PCI match macro ALSA: hda/i915: Use global PCI match macro ASoC: Intel: Skylake: Use global PCI match macro ALSA: intel-dsp-config: Convert to PCI device IDs defines ALSA: hda: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: avs: Convert to PCI device IDs defines ASoC: Intel: Skylake: Convert to PCI device IDs defines ASoC: SOF: Intel: Convert to PCI device IDs defines ASoC: Intel: sst: Convert to PCI device IDs defines include/linux/pci_ids.h | 104 +++++-- include/sound/hda_codec.h | 3 - include/sound/hdaudio.h | 27 ++ sound/hda/hdac_i915.c | 7 +- sound/hda/intel-dsp-config.c | 119 ++++---- sound/pci/hda/hda_intel.c | 373 ++++++++++--------------- sound/soc/intel/atom/sst/sst.c | 3 +- sound/soc/intel/atom/sst/sst.h | 1 - sound/soc/intel/atom/sst/sst_pci.c | 4 +- sound/soc/intel/avs/board_selection.c | 10 +- sound/soc/intel/avs/core.c | 16 +- sound/soc/intel/skylake/skl-messages.c | 16 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/intel/skylake/skl.c | 36 +-- sound/soc/sof/intel/pci-apl.c | 9 +- sound/soc/sof/intel/pci-cnl.c | 15 +- sound/soc/sof/intel/pci-icl.c | 12 +- sound/soc/sof/intel/pci-mtl.c | 3 +- sound/soc/sof/intel/pci-skl.c | 6 +- sound/soc/sof/intel/pci-tgl.c | 45 +-- sound/soc/sof/intel/pci-tng.c | 3 +- 21 files changed, 384 insertions(+), 431 deletions(-) -- 2.34.1

7 42

[PATCH v2 0/4] ASoC: rt5677: Refactor GPIO and use device_get_match_data()
by Andy Shevchenko 12 Jul '23

12 Jul '23

The code can be simplified with refactored GPIO parts and with use of device_get_match_data(). Besides that couple of additional changes, one for maintenance and one for making IRQ domain agnostic (not being pinned to OF). Changelog v2: - refactored GPIO code in (a new) patch 1 - fixed compilation error in patch 2 (LKP) Andy Shevchenko (4): ASoC: rt5677: Refactor GPIO support code ASoC: rt5677: Use agnostic irq_domain_create_linear() ASoC: rt5677: Use device_get_match_data() ASoC: rt5677: Sort headers alphabetically sound/soc/codecs/rt5677.c | 117 +++++++++++++------------------------- sound/soc/codecs/rt5677.h | 92 +++++------------------------- 2 files changed, 53 insertions(+), 156 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 8

[PATCH] ASoC: tlv320aic32x4: Fix the divide by zero
by Guiting Shen 12 Jul '23

12 Jul '23

The value of register(NDAC,MDAC,NADC,MADC,BCLKN) maybe zero lead to divide by zero in clk_aic32x4_div_recalc_rate().And the rate should be divide by 128 if the value was zero in this function according to the datasheet. Add the macro AIC32X4_DIV_MAX to present the 128 and return 0 if failing to read the value of register. Signed-off-by: Guiting Shen <aarongt.shen(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4-clk.c | 16 +++++++++++----- sound/soc/codecs/tlv320aic32x4.h | 5 +++-- 2 files changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index c116e82f712d..5c0a76a4a106 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -321,7 +321,7 @@ static int clk_aic32x4_div_set_rate(struct clk_hw *hw, unsigned long rate, u8 divisor; divisor = DIV_ROUND_UP(parent_rate, rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; return regmap_update_bits(div->regmap, div->reg, @@ -334,7 +334,7 @@ static int clk_aic32x4_div_determine_rate(struct clk_hw *hw, unsigned long divisor; divisor = DIV_ROUND_UP(req->best_parent_rate, req->rate); - if (divisor > 128) + if (divisor > AIC32X4_DIV_MAX) return -EINVAL; req->rate = DIV_ROUND_UP(req->best_parent_rate, divisor); @@ -345,12 +345,18 @@ static unsigned long clk_aic32x4_div_recalc_rate(struct clk_hw *hw, unsigned long parent_rate) { struct clk_aic32x4 *div = to_clk_aic32x4(hw); - unsigned int val; + int err; + + err = regmap_read(div->regmap, div->reg, &val); + if (err) + return 0; - regmap_read(div->regmap, div->reg, &val); + val &= AIC32X4_DIV_MASK; + if (!val) + val = AIC32X4_DIV_MAX; - return DIV_ROUND_UP(parent_rate, val & AIC32X4_DIV_MASK); + return DIV_ROUND_UP(parent_rate, val); } static const struct clk_ops aic32x4_div_ops = { diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 4de5bd9e8cc5..d6101ce73f80 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -223,8 +223,9 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name); #define AIC32X4_REFPOWERUP_120MS 0x07 /* Common mask and enable for all of the dividers */ -#define AIC32X4_DIVEN BIT(7) -#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIVEN BIT(7) +#define AIC32X4_DIV_MASK GENMASK(6, 0) +#define AIC32X4_DIV_MAX 128 /* Clock Limits */ #define AIC32X4_MAX_DOSR_FREQ 6200000 -- 2.25.1

1 0

[PATCH v2 00/37] ALSA: Add MIDI 2.0 support
by Takashi Iwai 12 Jul '23

12 Jul '23

Hi, this is a revised patch set for adding the support of MIDI 2.0 functionality. Only a few minor updates since v1 as below: * Bump CTL protocol version to 2.0.9 * Add two more ioctls for control API to receive UMP EP and block info * Minor compile warning fixes * Documentation fix and addition of two new ioctls * Put acks The latest code is found in topic/midi20 branch of sound git tree. And below is the cover letter of v1, which is still valid: Hi, this is a (largish) patch set for adding the support of MIDI 2.0 functionality, mainly targeted for USB devices. MIDI 2.0 is a complete overhaul of the 40-years old MIDI 1.0. Unlike MIDI 1.0 byte stream, MIDI 2.0 uses packets in 32bit words for Universal MIDI Packet (UMP) protocol. It supports both MIDI 1.0 commands for compatibility and the extended MIDI 2.0 commands for higher resolutions and more functions. For supporting the UMP, the patch set extends the existing ALSA rawmidi and sequencer interfaces, and adds the USB MIDI 2.0 support to the standard USB-audio driver. The rawmidi for UMP has a different device name (/dev/snd/umpC*D*) and it reads/writes UMP packet data in 32bit CPU-native endianness. For the old MIDI 1.0 applications, the legacy rawmidi interface is provided, too. As default, USB-audio driver will take the alternate setting for MIDI 2.0 interface, and the compatibility with MIDI 1.0 is provided via the rawmidi common layer. However, user may let the driver falling back to the old MIDI 1.0 interface by a module option, too. A UMP-capable rawmidi device can create the corresponding ALSA sequencer client(s) to support the UMP Endpoint and UMP Group connections. As a nature of ALSA sequencer, arbitrary connections between clients/ports are allowed, and the ALSA sequencer core performs the automatic conversions for the connections between a new UMP sequencer client and a legacy MIDI 1.0 sequencer client. It allows the existing application to use MIDI 2.0 devices without changes. The MIDI-CI, which is another major extension in MIDI 2.0, isn't covered by this patch set. It would be implemented rather in user-space. Roughly speaking, the first half of this patch set is for extending the rawmidi and USB-audio, and the second half is for extending the ALSA sequencer interface. The patch set is based on 6.4-rc2 kernel, but all patches can be cleanly applicable on 6.2 and 6.3 kernels, too (while 6.1 and older kernels would need minor adjustment for uapi header changes). The updates for alsa-lib and alsa-utils will follow shortly later. The author thanks members of MIDI Association OS/API Working Group, especially Andrew Mee, for great helps for the initial design and debugging / testing the drivers. Takashi --- Takashi Iwai (37): ALSA: rawmidi: Pass rawmidi directly to snd_rawmidi_kernel_open() ALSA: rawmidi: Add ioctl callback to snd_rawmidi_global_ops ALSA: rawmidi: UMP support ALSA: rawmidi: Skip UMP devices at SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE ALSA: ump: Add ioctls to inquiry UMP EP and Block info via control API ALSA: ump: Additional proc output ALSA: usb-audio: Manage number of rawmidis globally ALSA: usb-audio: Define USB MIDI 2.0 specs ALSA: usb-audio: USB MIDI 2.0 UMP support ALSA: usb-audio: Get UMP EP name string from USB interface ALSA: usb-audio: Trim superfluous "MIDI" suffix from UMP EP name ALSA: usb-audio: Create UMP blocks from USB MIDI GTBs ALSA: ump: Redirect rawmidi substream access via own helpers ALSA: ump: Add legacy raw MIDI support ALSA: usb-audio: Enable the legacy raw MIDI support ALSA: usb-audio: Inform inconsistent protocols in GTBs ALSA: seq: Clear padded bytes at expanding events ALSA: seq: Add snd_seq_expand_var_event_at() helper ALSA: seq: Treat snd_seq_client object directly in client drivers ALSA: seq: Drop dead code for the old broadcast support ALSA: seq: Check the conflicting port at port creation ALSA: seq: Check validity before creating a port object ALSA: seq: Prohibit creating ports with special numbers ALSA: seq: Introduce SNDRV_SEQ_IOCTL_USER_PVERSION ioctl ALSA: seq: Add UMP support ALSA: seq: Add port inactive flag ALSA: seq: Support MIDI 2.0 UMP Endpoint port ALSA: seq: Add port direction to snd_seq_port_info ALSA: seq: Add UMP group number to snd_seq_port_info ALSA: seq: Automatic conversion of UMP events ALSA: seq: Allow suppressing UMP conversions ALSA: seq: Bind UMP device ALSA: seq: ump: Create UMP Endpoint port for broadcast ALSA: seq: Add ioctls for client UMP info query and setup ALSA: seq: Print UMP Endpoint and Block information in proc outputs ALSA: seq: Add UMP group filter ALSA: docs: Add MIDI 2.0 documentation Documentation/sound/designs/index.rst | 1 + Documentation/sound/designs/midi-2.0.rst | 351 +++++++ include/linux/usb/midi-v2.h | 94 ++ include/sound/asequencer.h | 4 + include/sound/rawmidi.h | 15 +- include/sound/seq_device.h | 1 + include/sound/seq_kernel.h | 10 + include/sound/ump.h | 175 ++++ include/sound/ump_msg.h | 540 ++++++++++ include/uapi/sound/asequencer.h | 83 +- include/uapi/sound/asound.h | 62 +- sound/core/Kconfig | 13 + sound/core/Makefile | 3 + sound/core/rawmidi.c | 264 +++-- sound/core/rawmidi_compat.c | 4 + sound/core/seq/Kconfig | 14 + sound/core/seq/Makefile | 3 + sound/core/seq/seq_clientmgr.c | 577 +++++++---- sound/core/seq/seq_clientmgr.h | 27 +- sound/core/seq/seq_compat.c | 3 + sound/core/seq/seq_dummy.c | 9 + sound/core/seq/seq_memory.c | 98 +- sound/core/seq/seq_memory.h | 19 +- sound/core/seq/seq_midi.c | 12 +- sound/core/seq/seq_ports.c | 47 +- sound/core/seq/seq_ports.h | 23 +- sound/core/seq/seq_ump_client.c | 464 +++++++++ sound/core/seq/seq_ump_convert.c | 1203 ++++++++++++++++++++++ sound/core/seq/seq_ump_convert.h | 22 + sound/core/seq/seq_virmidi.c | 1 + sound/core/ump.c | 668 ++++++++++++ sound/core/ump_convert.c | 520 ++++++++++ sound/core/ump_convert.h | 43 + sound/usb/Kconfig | 11 + sound/usb/Makefile | 1 + sound/usb/card.c | 12 +- sound/usb/midi.c | 7 +- sound/usb/midi.h | 5 +- sound/usb/midi2.c | 1190 +++++++++++++++++++++ sound/usb/midi2.h | 33 + sound/usb/quirks.c | 8 +- sound/usb/usbaudio.h | 2 + 42 files changed, 6306 insertions(+), 336 deletions(-) create mode 100644 Documentation/sound/designs/midi-2.0.rst create mode 100644 include/linux/usb/midi-v2.h create mode 100644 include/sound/ump.h create mode 100644 include/sound/ump_msg.h create mode 100644 sound/core/seq/seq_ump_client.c create mode 100644 sound/core/seq/seq_ump_convert.c create mode 100644 sound/core/seq/seq_ump_convert.h create mode 100644 sound/core/ump.c create mode 100644 sound/core/ump_convert.c create mode 100644 sound/core/ump_convert.h create mode 100644 sound/usb/midi2.c create mode 100644 sound/usb/midi2.h -- 2.35.3

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ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - edited 12 Jul '23

12 Jul '23

alsa-project/alsa-ucm-conf issue #334 was edited from Shmob1: I have an [ASRock Z790 PG-ITX/TB4](https://pg.asrock.com/MB/Intel/Z790%20PG-ITXTB4/index.asp) which has the `ALC4082` Realtek chipset connected over USB, that has been mentioned in a number of issues, most recently #229. It seems the configuration changes in that issue (notably, https://github.com/hwchong/alsa-ucm-conf/commit/3ec7e6e2a61240764b77c30892f…) work on my board as well, with a small modification. Unlike the Taichi boards, my Realtek audio driver is at `USB26ce:0a08`, not `USB26ce:0a06`. Making the necessary regex changes in `ALC4080-HiFi.conf` and `USB-Audio.conf` enables functionality on my board. I am happy to make a PR to merge these changes if pertinent. Thanks for the great work! Update: I am unable to get more than 2 channels through spdif either using AC3 or DTS Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[RESEND 0/2] ASoC: mt8188: add memory-region support
by Trevor Wu 12 Jul '23

12 Jul '23

This series introduces support for memory-region, allowing afe memif to utilize the region specified in dts. Trevor Wu (2): ASoC: mediatek: mt8188: add memory-region support ASoC: dt-bindings: mediatek,mt8188-afe: add memory-region .../devicetree/bindings/sound/mediatek,mt8188-afe.yaml | 7 +++++++ sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++++-- 2 files changed, 13 insertions(+), 2 deletions(-) -- 2.18.0

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[PATCH 1/4] ASoC: codecs: es8326: Add es8326_mute function
by Zhu Ning 12 Jul '23

12 Jul '23

The internal analog power and hp Vref of es8326 should always be on to reduce pop noise. The HP_VOL and HP_CAL are moved to es8326_mute function so they are turned on at last and turned off at first. Also, the calibration should be done manually once during start-up to reduce DC offset on headphone. Signed-off-by: Zhu Ning <zhuning0077(a)gmail.com> --- sound/soc/codecs/es8326.c | 75 ++++++++++++++++----------------------- sound/soc/codecs/es8326.h | 5 +-- 2 files changed, 34 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index ed348bbf5f49..9190a8779334 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -38,6 +38,9 @@ struct es8326_priv { u8 interrupt_clk; bool jd_inverted; unsigned int sysclk; + + bool calibrated; + int version; }; static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); @@ -121,33 +124,12 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { /* Analog Power Supply*/ SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), - SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), - /* Headphone Charge Pump and Output */ - SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, - 3, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, - 2, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, - 1, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, - 0, 1, NULL, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, - ES8326_HPOR_SHIFT, 7, 7, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, - 0, 7, 7, 0), - SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -166,34 +148,12 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"I2S OUT", NULL, "ADC L"}, {"I2S OUT", NULL, "ADC R"}, - {"I2S OUT", NULL, "Analog Power"}, - {"I2S OUT", NULL, "ADC Vref"}, - {"I2S OUT", NULL, "Vref Power"}, - {"I2S OUT", NULL, "IBias Power"}, - {"I2S IN", NULL, "Analog Power"}, - {"I2S IN", NULL, "DAC Vref"}, - {"I2S IN", NULL, "Vref Power"}, - {"I2S IN", NULL, "IBias Power"}, - {"Right DAC", NULL, "I2S IN"}, {"Left DAC", NULL, "I2S IN"}, {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, - {"HPOR", NULL, "HPOR Cal"}, - {"HPOL", NULL, "HPOL Cal"}, - {"HPOR", NULL, "HPOR Supply"}, - {"HPOL", NULL, "HPOL Supply"}, - {"HPOL", NULL, "Headphone Charge Pump"}, - {"HPOR", NULL, "Headphone Charge Pump"}, - {"HPOL", NULL, "Headphone Driver Bias"}, - {"HPOR", NULL, "Headphone Driver Bias"}, - {"HPOL", NULL, "Headphone LDO"}, - {"HPOR", NULL, "Headphone LDO"}, - {"HPOL", NULL, "Headphone Reference"}, - {"HPOR", NULL, "Headphone Reference"}, - {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; @@ -419,6 +379,31 @@ static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) +{ + struct snd_soc_component *component = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + if (mute) { + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x00); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ES8326_MUTE); + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xf0); + } else { + if (!es8326->calibrated) { + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x88); + msleep(30); + es8326->calibrated = true; + } + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa0); + regmap_write(es8326->regmap, ES8326_HP_VOL, 0x00); + regmap_write(es8326->regmap, ES8326_HP_CAL, 0x77); + regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, + ES8326_MUTE_MASK, ~(ES8326_MUTE)); + } + return 0; +} + static int es8326_set_bias_level(struct snd_soc_component *codec, enum snd_soc_bias_level level) { @@ -469,6 +454,8 @@ static const struct snd_soc_dai_ops es8326_ops = { .hw_params = es8326_pcm_hw_params, .set_fmt = es8326_set_dai_fmt, .set_sysclk = es8326_set_dai_sysclk, + .mute_stream = es8326_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver es8326_dai = { @@ -691,7 +678,7 @@ static int es8326_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&es8326->jack_detect_work); es8326_disable_micbias(component); - + es8326->calibrated = false; regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); regcache_mark_dirty(es8326->regmap); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 8e5ffe5ee10d..b837ed8c4b39 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -9,8 +9,6 @@ #ifndef _ES8326_H #define _ES8326_H -#define CONFIG_HHTECH_MINIPMP 1 - /* ES8326 register space */ #define ES8326_RESET 0x00 #define ES8326_CLK_CTL 0x01 @@ -94,6 +92,8 @@ #define ES8326_PWRUP_SEQ_EN (1 << 5) #define ES8326_CODEC_RESET (0x0f << 0) #define ES8326_CSM_OFF (0 << 7) +#define ES8326_MUTE_MASK (3 << 0) +#define ES8326_MUTE (3 << 0) /* ES8326_CLK_CTL */ #define ES8326_CLK_ON (0x7f << 0) @@ -180,3 +180,4 @@ #define ES8326_VERSION_B (1 << 0) #endif + -- 2.34.1

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[PATCH 0/3] regmap: Fix REGMAP selections
by Geert Uytterhoeven 12 Jul '23

12 Jul '23

Hi all, This patch series fixes several missing selects related to REGMAP. The first patch fixes a serious thinko I made when making the regmap KUnit test depend on REGMAP. The last two patches fix missing selects I discovered when diving deeper into the REGMAP_* => REGMAP logic. Thanks for your comments! Geert Uytterhoeven (3): regmap: Replace "default y if" by select regmap: REGMAP_SLIMBUS should select REGMAP ASoC: codecs: SND_SOC_WCD934X should select REGMAP_IRQ drivers/base/regmap/Kconfig | 16 +++++++++++++++- drivers/mfd/Kconfig | 1 - sound/soc/codecs/Kconfig | 1 + 3 files changed, 16 insertions(+), 2 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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