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

[PATCH] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 10 Jul '23

10 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/sof_rt5682.c | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 2 files changed, 90 insertions(+), 3 deletions(-) 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

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[PATCH v2] soundwire: amd: Fix a check for errors in probe()
by Dan Carpenter 10 Jul '23

10 Jul '23

This code has two problems: 1) The devm_ioremap() function returns NULL, not error pointers. 2) It's checking the wrong variable. ->mmio instead of ->acp_mmio. Fixes: a673a8dfc214 ("soundwire: amd: Add support for AMD Manager driver") Suggested-by: "Mukunda,Vijendar" <vijendar.mukunda(a)amd.com> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- v2: Change the variable. My original fix would have prevented the driver from loading so many thanks to Vijendar for pointing that out. drivers/soundwire/amd_manager.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 08aeb7ed00e1..3a99f6dcdfaf 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -910,9 +910,9 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -ENOMEM; amd_manager->acp_mmio = devm_ioremap(dev, res->start, resource_size(res)); - if (IS_ERR(amd_manager->mmio)) { + if (!amd_manager->acp_mmio) { dev_err(dev, "mmio not found\n"); - return PTR_ERR(amd_manager->mmio); + return -ENOMEM; } amd_manager->instance = pdata->instance; amd_manager->mmio = amd_manager->acp_mmio + -- 2.39.2

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[PATCH v2] sound: seq: fix typo in system_2p_ev_to_ump_midi1()
by Minjie Du 10 Jul '23

10 Jul '23

Fix data->system.parm2 typo. Signed-off-by: Minjie Du <duminjie(a)vivo.com> --- sound/core/seq/seq_ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index eb1d86ff6..7cc84e137 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -714,7 +714,7 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, { data->system.status = status; data->system.parm1 = (event->data.control.value >> 7) & 0x7f; - data->system.parm1 = event->data.control.value & 0x7f; + data->system.parm2 = event->data.control.value & 0x7f; return 1; } -- 2.39.0

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[PATCH] ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

dB range for HPHL and HPHR gains are from +6dB to -30dB in steps of 1.5dB with register values range from 0 to 24. Current code maps these dB ranges incorrectly, fix them to allow proper volume setting. Fixes: e8ba1e05bdc0("ASoC: codecs: wcd938x: add basic controls") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index faa15a5ed2c8..3a3360711f8f 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -210,7 +210,7 @@ struct wcd938x_priv { }; static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(ear_pa_gain, 600, -1800); -static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(line_gain, 600, -3000); +static const DECLARE_TLV_DB_SCALE(line_gain, -3000, 150, -3000); static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(analog_gain, 0, 3000); struct wcd938x_mbhc_zdet_param { @@ -2655,8 +2655,8 @@ static const struct snd_kcontrol_new wcd938x_snd_controls[] = { wcd938x_get_swr_port, wcd938x_set_swr_port), SOC_SINGLE_EXT("DSD_R Switch", WCD938X_DSD_R, 0, 1, 0, wcd938x_get_swr_port, wcd938x_set_swr_port), - SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 0, line_gain), - SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 0, line_gain), + SOC_SINGLE_TLV("HPHL Volume", WCD938X_HPH_L_EN, 0, 0x18, 1, line_gain), + SOC_SINGLE_TLV("HPHR Volume", WCD938X_HPH_R_EN, 0, 0x18, 1, line_gain), WCD938X_EAR_PA_GAIN_TLV("EAR_PA Volume", WCD938X_ANA_EAR_COMPANDER_CTL, 2, 0x10, 0, ear_pa_gain), SOC_SINGLE_EXT("ADC1 Switch", WCD938X_ADC1, 1, 1, 0, -- 2.25.1

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[PATCH v2 0/4] soundwire: qcom: stablity fixes
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

During x13s audio testing we hit few corner cases due to issues in codec drivers and some obvious code bugs. Here are the fixes for those issues, mostly the issues are around devices loosing the sync in between runtime pm suspend resume path. With codec fixes along with these fixes, audio on x13s is pretty stable. Thanks, Srini Changes since v1: - dropped runtime pm changes patch as unable to reproduced it anymore - fixed clk stop flag as suggested by Pierre - rebased on top of linux-next Srinivas Kandagatla (4): soundwire: qcom: update status correctly with mask soundwire: qcom: wait for fifo to be empty before suspend soundwire: qcom: add software workaround for bus clash interrupt assertion soundwire: qcom: set clk stop need reset flag at runtime drivers/soundwire/qcom.c | 100 ++++++++++++++++++++++++++------------- 1 file changed, 68 insertions(+), 32 deletions(-) -- 2.21.0

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on (not) breaking user space
by Oswald Buddenhagen 09 Jul '23

09 Jul '23

hi linus, the alsa maintainers seem to be applying a reading of "we don't break user space" that i find counter-productive. so i would appreciate you clarifying your position on that matter (once more ...) to be clear, we're not arguing over changing any abi, but simply exercising one somewhat more than before. the claim is that this would likely have unacceptable consequences. i disagree. context: > Due to hardware constraints, changing the clock multiplier (CM) > changes the available audio ports and the number of available > channels. [...] Therefore, we do dynamic reconfiguration (DR) of the > mixer in response to changing the CM. arguments: > DR is somewhat controversial, as it has the potential to crash poorly > programmed applications. > But that in itself isn't a very convincing argument against it, as by > that logic we'd have to ban all hot-plugging. (1) > Such crashes would also not really qualify as regressions, as the > D.A.S. mode is a new opt-in feature, and therefore no previously > stable setups would be impacted. (2) the full text is at https://lore.kernel.org/all/20230630144542.664190-7-oswald.buddenhagen@gmx.… the preceding discussion is at https://lore.kernel.org/all/87v8fren1k.wl-tiwai@suse.de/T/#u regards, ossi

1 0

[Feature Request] New hdajackretask ncurses TUI
by GitHub issues - opened 09 Jul '23

09 Jul '23

alsa-project/alsa-tools issue #16 was opened from HalanoSiblee: Hi Why i need gtk GUI when I do such a simple retaskable jack input Please make hdajackretask with ncurses like alsamixer to work in tty and terminal's at this point it will be so useful less bloat ,and blazingly fast Thanks. Issue URL : https://github.com/alsa-project/alsa-tools/issues/16 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH 1/2] ALSA: pcmtest: Convert to platform remove callback returning void
by Uwe Kleine-König 07 Jul '23

07 Jul '23

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new() which already returns void. Eventually after all drivers are converted, .remove_new() is renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/drivers/pcmtest.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 2ae912a64d16..1fff85feaf82 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -576,10 +576,9 @@ static int pcmtst_probe(struct platform_device *pdev) return 0; } -static int pdev_remove(struct platform_device *dev) +static void pdev_remove(struct platform_device *dev) { snd_pcmtst_free(pcmtst); - return 0; } static struct platform_device pcmtst_pdev = { @@ -589,7 +588,7 @@ static struct platform_device pcmtst_pdev = { static struct platform_driver pcmtst_pdrv = { .probe = pcmtst_probe, - .remove = pdev_remove, + .remove_new = pdev_remove, .driver = { .name = "pcmtest", }, base-commit: 5133c9e51de41bfa902153888e11add3342ede18 -- 2.39.2

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[PATCH] dt-bindings: sound: Update maintainer email id
by Rohit kumar 07 Jul '23

07 Jul '23

Updated my mail id to latest quicinc id. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> --- .../devicetree/bindings/sound/google,sc7180-trogdor.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 666a95ac22c8..ba5b7728cf33 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Google SC7180-Trogdor ASoC sound card driver maintainers: - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> - Cheng-Yi Chiang <cychiang(a)chromium.org> description: diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index 6cc8f86c7531..3a559bd07a79 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -8,7 +8,7 @@ title: Qualcomm Technologies Inc. LPASS CPU dai driver maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> description: | Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist -- 2.25.1

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[PATCH] ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode
by Shengjiu Wang 07 Jul '23

07 Jul '23

On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..dae16a14f177 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,6 +437,12 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, 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; } -- 2.17.1

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