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

[PATCH v2] 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 | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 5 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..53451c1b67f8 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 */ @@ -305,7 +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 ret; + int ret, extra; /* need to enable ASRC function for 24MHz mclk rate */ if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && @@ -315,7 +317,16 @@ 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 +376,12 @@ 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 = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra); + } 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 +418,8 @@ 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 +440,8 @@ 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 +451,9 @@ 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 +582,30 @@ 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 +656,17 @@ 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 +705,9 @@ 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 +860,11 @@ 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 +951,12 @@ 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 +1245,14 @@ 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..36ea2bacbe7d 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-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH] ASoC: amd: ps: Fix extraneous error messages
by Mario Limonciello 11 Jul '23

11 Jul '23

On Pink Sardine machines that don't have SdW controllers in use, the property `mipi-sdw-manager-list` won't exist. There is no point in showing an error to a user when this situation is encountered. Furthermore if the machine doesn't have a DMIC connected to the ACP, there may be no platform devices created either. Downgrade the associated message to debug. Fixes: d1351c30ac8a6 ("ASoC: amd: ps: create platform devices based on acp config") Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 5b46dc8573f86..4af3c3665387d 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -257,7 +257,7 @@ static int sdw_amd_scan_controller(struct device *dev) &sdw_manager_bitmap, 1); if (ret) { - dev_err(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); + dev_dbg(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); return -EINVAL; } count = hweight32(sdw_manager_bitmap); @@ -641,7 +641,7 @@ static int snd_acp63_probe(struct pci_dev *pci, ret = get_acp63_device_config(val, pci, adata); /* ACP PCI driver probe should be continued even PDM or SoundWire Devices are not found */ if (ret) { - dev_err(&pci->dev, "get acp device config failed:%d\n", ret); + dev_dbg(&pci->dev, "get acp device config failed:%d\n", ret); goto skip_pdev_creation; } ret = create_acp63_platform_devs(pci, adata, addr); base-commit: 8689f4f2ea561dd080118eeb05c0255ac9542905 -- 2.34.1

2 1

[PATCH] ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode"
by Fabio Estevam 11 Jul '23

11 Jul '23

From: Fabio Estevam <festevam(a)denx.de> This reverts commit ff87d619ac180444db297f043962a5c325ded47b. Andreas reports that on an i.MX8MP-based system where MCLK needs to be used as an input, the MCLK pin is actually an output, despite not having the 'fsl,sai-mclk-direction-output' property present in the devicetree. This is caused by commit ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") that sets FSL_SAI_MCTL_MCLK_EN unconditionally for imx8mm/8mn/8mp/93, causing the MCLK to always be configured as output. FSL_SAI_MCTL_MCLK_EN corresponds to the MOE (MCLK Output Enable) bit of register MCR and the drivers sets it when the 'fsl,sai-mclk-direction-output' devicetree property is present. Revert the commit to allow SAI to use MCLK as input as well. Cc: stable(a)vger.kernel.org Fixes: ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Reported-by: Andreas Henriksson <andreas(a)fatal.se> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl_sai.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..54b4bf3744c6 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -507,12 +507,6 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) savediv / 2 - 1); } - if (sai->soc_data->max_register >= FSL_SAI_MCTL) { - /* SAI is in master mode at this point, so enable MCLK */ - regmap_update_bits(sai->regmap, FSL_SAI_MCTL, - FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); - } - return 0; } -- 2.34.1

3 2

[PATCH] ASoC: qcom: q6apm: do not close GPR port before closing graph
by Srinivas Kandagatla 11 Jul '23

11 Jul '23

Closing GPR port before graph close can result in un handled notifications from DSP, this results in spam of errors from GPR driver as there is no one to handle these notification at that point in time. Fix this by closing GPR port after graph close is finished. Fixes: 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 7bfac9492ab5..5d44d07acd69 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -511,6 +511,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) switch (hdr->opcode) { case DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_WRITE_DONE; mutex_lock(&graph->lock); token = hdr->token & APM_WRITE_TOKEN_MASK; @@ -544,6 +546,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) wake_up(&graph->cmd_wait); break; case DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_READ_DONE; mutex_lock(&graph->lock); rd_done = data->payload; @@ -649,8 +653,9 @@ int q6apm_graph_close(struct q6apm_graph *graph) { struct audioreach_graph *ar_graph = graph->ar_graph; - gpr_free_port(graph->port); + graph->ar_graph = NULL; kref_put(&ar_graph->refcount, q6apm_put_audioreach_graph); + gpr_free_port(graph->port); kfree(graph); return 0; -- 2.25.1

2 1

[PATCH 0/8] ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral
by Johan Hovold 11 Jul '23

11 Jul '23

I've been hitting a race during boot which breaks probe of the sound card on the Lenovo ThinkPad X13s as I've previously reported here: https://lore.kernel.org/all/ZIHMMFtuDtvdpFAZ@hovoldconsulting.com/ The immediate issue appeared to be a probe deferral that was turned into a hard failure, but addressing that in itself only made things worse as it exposed further bugs. I was hoping someone more familiar with the code in question would look into this, but as this affects users of the X13s and breaks audio on my machine every fifth boot or so, I decided to investigate it myself. As expected, the Qualcomm codec drivers are broken and specifically leak resources on component remove, which in turn breaks sound card probe deferrals. The source of the deferral itself appears to be legitimate and was simply due to some audio component not yet having been registered due to random changes in timing during boot. These issues can most easily be reproduced by simply blacklisting the q6apm_dai module and loading it manually after boot. The sound card probe deferral also exposes a bug in the soundwire subsystem, which uses completion structures for signalling that a device has been enumerated on the bus and initialised. The way this is implemented prevents reprobed codec drivers from learning that the soundwire devices are still attached, which causes probe to fail. Included are also two patches that suppresses error messages on component probe deferral to avoid spamming the logs during boot. These patches should preferably all go through the ASoC tree even if merging the soundwire fix separately also works. Note the ASoC tree already has the following related fixes: https://lore.kernel.org/lkml/20230630120318.6571-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230630142717.5314-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230701094723.29379-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230703124701.11734-1-johan+linaro@kernel.org/ Johan Johan Hovold (8): soundwire: fix enumeration completion ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors drivers/soundwire/bus.c | 8 ++--- sound/soc/codecs/wcd-mbhc-v2.c | 57 ++++++++++++++++++++++--------- sound/soc/codecs/wcd934x.c | 12 +++++++ sound/soc/codecs/wcd938x.c | 59 +++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/topology.c | 4 +-- sound/soc/soc-core.c | 6 ++-- sound/soc/soc-topology.c | 10 ++++-- 7 files changed, 122 insertions(+), 34 deletions(-) -- 2.39.3

7 22

[PATCH] ASoC: codecs: aw88395: Add of_match_table
by Konrad Dybcio 11 Jul '23

11 Jul '23

The i2c_match_id entry is not enough to get the device probing with DT. Add an of match table in accordance with the bindings. Fixes: 62fc25fbab5f ("ASoC: codecs: Add i2c and codec registration for aw88395 and their associated operation functions") Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- sound/soc/codecs/aw88395/aw88395.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/aw88395/aw88395.c b/sound/soc/codecs/aw88395/aw88395.c index 9dcd75dd799a..5ff01d3c31c8 100644 --- a/sound/soc/codecs/aw88395/aw88395.c +++ b/sound/soc/codecs/aw88395/aw88395.c @@ -566,9 +566,18 @@ static const struct i2c_device_id aw88395_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aw88395_i2c_id); +#ifdef CONFIG_OF +static const struct of_device_id aw88395_of_match[] = { + { .compatible = "awinic,aw88395" }, + { }, +}; +MODULE_DEVICE_TABLE(of, aw88395_of_match); +#endif + static struct i2c_driver aw88395_i2c_driver = { .driver = { .name = AW88395_I2C_NAME, + .of_match_table = of_match_ptr(aw88395_of_match), }, .probe = aw88395_i2c_probe, .id_table = aw88395_i2c_id, --- base-commit: 8e4b7f2f3d6071665b1dfd70786229c8a5d6c256 change-id: 20230711-topic-aw88395_of_tbl-68b16c5860a5 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

1 0

[PATCH v3 00/14] spi: Header and core clean up and refactoring
by Andy Shevchenko 11 Jul '23

11 Jul '23

Various cleanups and refactorings of the SPI header and core parts united in a single series. It also touches drivers under SPI subsystem folder on the pure renaming purposes of some constants. No functional change intended (with some subtle shortcuts which are explained in the respective commit messages). Changelog v3: - dropped controversial used to be patches 3,4,8,10 (Mark) - amended many commit messages (Mark) - added tag to patch 1 (AngeloGioacchino) - split used to be patch 2 to patches 2 & 3 for better review (Mark) - rewritten used to be patch 5 to patches 4 & 5 (Mark, Sebastian) - added new patch 7 - fixed typos and added tag to patch 12 (Serge) v2: 20230710154932.68377-1-andriy.shevchenko(a)linux.intel.com Changelog v2: - added new patches 3,4,5,10,13,14 - massaged comment and kernel doc in patch 9 - split used to be patch 4 to patches 11,12 - covered a few things in SPI core in patch 15 - amended commit message for above (Mark) - reshuffled patches in the series for better logical grouping Andy Shevchenko (14): spi: Remove unneeded OF node NULL checks spi: Deduplicate IDR allocation code in spi_register_controller() spi: Refactor bus number assigning code in spi_register_controller() spi: Remove code duplication in spi_add_device*() spi: Kill spi_add_device_locked() spi: Use sysfs_emit() to instead of s*printf() spi: Use BITS_TO_BYTES() spi: Sort headers alphabetically spi: Use struct_size() helper spi: Get rid of old SPI_MASTER_NO_TX & SPI_MASTER_NO_RX spi: Get rid of old SPI_MASTER_MUST_TX & SPI_MASTER_MUST_RX spi: Rename SPI_MASTER_GPIO_SS to SPI_CONTROLLER_GPIO_SS spi: Convert to SPI_CONTROLLER_HALF_DUPLEX spi: Fix spelling typos and acronyms capitalization drivers/spi/spi-amd.c | 2 +- drivers/spi/spi-at91-usart.c | 2 +- drivers/spi/spi-ath79.c | 2 +- drivers/spi/spi-atmel.c | 4 +- drivers/spi/spi-bitbang-txrx.h | 16 +-- drivers/spi/spi-bitbang.c | 8 +- drivers/spi/spi-cavium-thunderx.c | 2 +- drivers/spi/spi-davinci.c | 2 +- drivers/spi/spi-dw-core.c | 2 +- drivers/spi/spi-falcon.c | 2 +- drivers/spi/spi-fsl-lpspi.c | 2 +- drivers/spi/spi-gpio.c | 10 +- drivers/spi/spi-imx.c | 2 +- drivers/spi/spi-lp8841-rtc.c | 10 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-mt65xx.c | 2 +- drivers/spi/spi-mxs.c | 2 +- drivers/spi/spi-omap-uwire.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pci1xxxx.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-qcom-qspi.c | 2 +- drivers/spi/spi-rb4xx.c | 2 +- drivers/spi/spi-rockchip-sfc.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-sifive.c | 2 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-sprd-adi.c | 2 +- drivers/spi/spi-stm32.c | 2 +- drivers/spi/spi-ti-qspi.c | 2 +- drivers/spi/spi-xcomm.c | 2 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 211 +++++++++++++----------------- include/linux/spi/spi.h | 173 ++++++++++++------------ include/trace/events/spi.h | 2 +- 36 files changed, 238 insertions(+), 252 deletions(-) -- 2.40.0.1.gaa8946217a0b

2 15

[PATCH 1/2] ALSA: emu10k1: make E-MU dock monitoring interrupt-driven
by Oswald Buddenhagen 11 Jul '23

11 Jul '23

... instead of using a one-second polling timer. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- include/sound/emu10k1.h | 4 +-- sound/pci/emu10k1/emu10k1.c | 8 +---- sound/pci/emu10k1/emu10k1_main.c | 51 ++++++++++++++++++-------------- sound/pci/emu10k1/io.c | 2 ++ sound/pci/emu10k1/irq.c | 7 +++++ 5 files changed, 40 insertions(+), 32 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 386a5f3be3e0..43c097952c3c 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1678,8 +1678,7 @@ struct snd_emu1010 { unsigned int clock_fallback; unsigned int optical_in; /* 0:SPDIF, 1:ADAT */ unsigned int optical_out; /* 0:SPDIF, 1:ADAT */ - struct delayed_work firmware_work; - u32 last_reg; + struct work_struct firmware_work; }; struct snd_emu10k1 { @@ -1761,6 +1760,7 @@ struct snd_emu10k1 { void (*capture_efx_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*spdif_interrupt)(struct snd_emu10k1 *emu, unsigned int status); void (*dsp_interrupt)(struct snd_emu10k1 *emu); + void (*gpio_interrupt)(struct snd_emu10k1 *emu); void (*p16v_interrupt)(struct snd_emu10k1 *emu); struct snd_pcm_substream *pcm_capture_substream; diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 23adace1b969..1a13c086e86c 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -176,9 +176,6 @@ static int snd_card_emu10k1_probe(struct pci_dev *pci, if (err < 0) return err; - if (emu->card_capabilities->emu_model) - schedule_delayed_work(&emu->emu1010.firmware_work, 0); - pci_set_drvdata(pci, card); dev++; return 0; @@ -194,7 +191,7 @@ static int snd_emu10k1_suspend(struct device *dev) emu->suspend = 1; - cancel_delayed_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.firmware_work); snd_ac97_suspend(emu->ac97); @@ -224,9 +221,6 @@ static int snd_emu10k1_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); - if (emu->card_capabilities->emu_model) - schedule_delayed_work(&emu->emu1010.firmware_work, 0); - return 0; } diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 58ed72de6403..661164dbf547 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -391,7 +391,10 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) } #endif - snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); + if (emu->card_capabilities->emu_model) + snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE | INTE_A_GPIOENABLE); + else + snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); } int snd_emu10k1_done(struct snd_emu10k1 *emu) @@ -745,63 +748,61 @@ static void emu1010_firmware_work(struct work_struct *work) int err; emu = container_of(work, struct snd_emu10k1, - emu1010.firmware_work.work); + emu1010.firmware_work); if (emu->card->shutdown) return; #ifdef CONFIG_PM_SLEEP if (emu->suspend) return; #endif - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */ snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ /* Return to Audio Dock programming mode */ dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw); if (err < 0) - goto next; - + return; snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); - dev_info(emu->card->dev, - "emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x%x\n", tmp); - /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, &tmp); dev_info(emu->card->dev, "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", tmp); if ((tmp & 0x1f) != 0x15) { /* FPGA failed to be programmed */ dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", tmp); - goto next; + return; } dev_info(emu->card->dev, "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); dev_info(emu->card->dev, "Audio Dock ver: %u.%u\n", tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); - } else if (!reg && emu->emu1010.last_reg) { + } +} + +static void emu1010_interrupt(struct snd_emu10k1 *emu) +{ + u32 sts; + + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &sts); + if (sts & EMU_HANA_IRQ_DOCK_LOST) { /* Audio Dock removed */ dev_info(emu->card->dev, "emu1010: Audio Dock detached\n"); /* The hardware auto-mutes all, so we unmute again */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (sts & EMU_HANA_IRQ_DOCK) { + schedule_work(&emu->emu1010.firmware_work); } - - next: - emu->emu1010.last_reg = reg; - if (!emu->card->shutdown) - schedule_delayed_work(&emu->emu1010.firmware_work, - msecs_to_jiffies(1000)); } /* @@ -870,6 +871,8 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &reg); dev_info(emu->card->dev, "emu1010: Card options = 0x%x\n", reg); + if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) + schedule_work(&emu->emu1010.firmware_work); if (emu->card_capabilities->no_adat) { emu->emu1010.optical_in = 0; /* IN_SPDIF */ emu->emu1010.optical_out = 0; /* OUT_SPDIF */ @@ -895,10 +898,12 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) /* MIDI routing */ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, EMU_HANA_MIDI_INA_FROM_HAMOA | EMU_HANA_MIDI_INB_FROM_DOCK2); snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, EMU_HANA_MIDI_OUT_DOCK2 | EMU_HANA_MIDI_OUT_SYNC2); - /* IRQ Enable: All on */ - /* snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x0f); */ - /* IRQ Enable: All off */ - snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); + + emu->gpio_interrupt = emu1010_interrupt; + // Note: The Audigy INTE is set later + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, + EMU_HANA_IRQ_DOCK | EMU_HANA_IRQ_DOCK_LOST); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &reg); // Clear pending IRQs emu->emu1010.clock_source = 1; /* 48000 */ emu->emu1010.clock_fallback = 1; /* 48000 */ @@ -938,7 +943,7 @@ static void snd_emu10k1_free(struct snd_card *card) /* Disable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } - cancel_delayed_work_sync(&emu->emu1010.firmware_work); + cancel_work_sync(&emu->emu1010.firmware_work); release_firmware(emu->firmware); release_firmware(emu->dock_fw); snd_util_memhdr_free(emu->memhdr); @@ -1517,7 +1522,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->irq = -1; emu->synth = NULL; emu->get_synth_voice = NULL; - INIT_DELAYED_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); + INIT_WORK(&emu->emu1010.firmware_work, emu1010_firmware_work); /* read revision & serial */ emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index a0d66ce3ee83..c695cb863e5e 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -302,6 +302,8 @@ static void snd_emu1010_fpga_read_locked(struct snd_emu10k1 *emu, u32 reg, u32 * { // The higest input pin is used as the designated interrupt trigger, // so it needs to be masked out. + // But note that any other input pin change will also cause an IRQ, + // so using this function often causes an IRQ as a side effect. u32 mask = emu->card_capabilities->ca0108_chip ? 0x1f : 0x7f; if (snd_BUG_ON(reg > 0x3f)) return; diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c index a813ef8c2f8d..8573248dd799 100644 --- a/sound/pci/emu10k1/irq.c +++ b/sound/pci/emu10k1/irq.c @@ -149,6 +149,13 @@ irqreturn_t snd_emu10k1_interrupt(int irq, void *dev_id) outl(0, emu->port + INTE2); status &= ~IPR_P16V; } + if (status & IPR_A_GPIO) { + if (emu->gpio_interrupt) + emu->gpio_interrupt(emu); + else + snd_emu10k1_intr_disable(emu, INTE_A_GPIOENABLE); + status &= ~IPR_A_GPIO; + } if (status) { dev_err(emu->card->dev, -- 2.40.0.152.g15d061e6df

2 7

ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - edited 11 Jul '23

11 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! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock Z790 PG-ITX/TB4
by GitHub issues - opened 11 Jul '23

11 Jul '23

alsa-project/alsa-ucm-conf issue #334 was opened 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 required. Thanks for the great work! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/334 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0