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

[PATCH] ASoC: Intel: sof_sdw: Always register the HDMI dai links
by Peter Ujfalusi 24 Nov '23

24 Nov '23

The topology files for SDW devices require HDMI dai links to be present and this is granted under normal conditions but in case of special use cases the display (i915) driver might not be enabled due to deny-listing, booting with nomodeset or just not compiled at all. This should not block the non HDMI audio to be usable so register the dai links unconditionally. The code has been prepared for this and in case of no HDMI audio the link is created with dummy codec. Closes: https://github.com/thesofproject/linux/issues/4594 Closes: https://github.com/thesofproject/linux/issues/4648 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3312ad8a563b..4e4284729773 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1546,7 +1546,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); - int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, hdmi_num = 0, bt_num = 0; + int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0; struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; @@ -1564,6 +1564,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) char *codec_name, *codec_dai_name; int i, j, be_id = 0; int codec_index; + int hdmi_num; int ret; ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); @@ -1584,14 +1585,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ssp_num = hweight_long(ssp_mask); } - if (mach_params->codec_mask & IDISP_CODEC_MASK) { + if (mach_params->codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) - hdmi_num = SOF_TGL_HDMI_COUNT; - else - hdmi_num = SOF_PRE_TGL_HDMI_COUNT; - } + if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) + hdmi_num = SOF_TGL_HDMI_COUNT; + else + hdmi_num = SOF_PRE_TGL_HDMI_COUNT; /* enable dmic01 & dmic16k */ if (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) @@ -1601,7 +1601,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bt_num = 1; dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d, bt: %d\n", - sdw_be_num, ssp_num, dmic_num, hdmi_num, bt_num); + sdw_be_num, ssp_num, dmic_num, + ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); /* allocate BE dailinks */ num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num; -- 2.42.1

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[PATCH] ASoC: Intel: skl_hda_dsp_generic: Drop HDMI routes when HDMI is not available
by Peter Ujfalusi 24 Nov '23

24 Nov '23

When the HDMI is not present due to disabled display support we will use dummy codec and the HDMI routes will refer to non existent DAPM widgets. Trim the route list from the HDMI routes to be able to probe the card even if the HDMI dais are not registered. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6c6ef63cd5d9..6e172719c979 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -154,6 +154,8 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); if (!ctx->idisp_codec) { + card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; + num_route -= IDISP_ROUTE_COUNT; for (i = 0; i < IDISP_DAI_COUNT; i++) { skl_hda_be_dai_links[i].codecs = &snd_soc_dummy_dlc; skl_hda_be_dai_links[i].num_codecs = 1; -- 2.42.1

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[PATCH v3 0/2] Introduce playback-only/capture-only DAI link flags
by Daniel Baluta 24 Nov '23

24 Nov '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries allows users to specify a link only direction with audio-graph-card2. Changes since v1 - https://lore.kernel.org/lkml/20230801082433.548206-1-daniel.baluta@oss.nxp.… - used audio-graph-card2 instead of simple-card as we are deprecting simple-card. Changes since v2: - Fix device tree binding error reported by Rob Daniel Baluta (2): ASoC: audio-graph-card2: Introduce playback-only/capture only DAI link flags ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only .../bindings/sound/audio-graph-port.yaml | 6 ++++++ include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 4 files changed, 34 insertions(+) -- 2.25.1

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pcm: clarify documentation of poll descriptor usage
by GitHub pull_request - opened 24 Nov '23

24 Nov '23

alsa-project/alsa-lib pull request #370 was opened from z-s-e: This is based on my understanding of the intended behavior and the test/pcm.c example code. There needs to be more clarifiaction regarding the exact semantics of the value of the revents output parameter of snd_pcm_poll_descriptors_revents, since there are events that do not necessarily correspond to POLLIN or POLLOUT (such as period events), but I believe this is a lot less obvious and needs confirmation first. Request URL : https://github.com/alsa-project/alsa-lib/pull/370 Patch URL : https://github.com/alsa-project/alsa-lib/pull/370.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[RESEND] ASoC: cs43130: Allow configuration of bit clock and frame inversion
by Maciej Strozek 24 Nov '23

24 Nov '23

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- Resending patch which previously broke x86 allmodconfig [1], now rebased on for-next branch [1]: https://lore.kernel.org/all/ZVfLMIbCpShVAYBf@finisterre.sirena.org.uk/ sound/soc/codecs/cs43130.c | 42 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 4f16baf4eafb..0f3ead84665f 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -620,6 +620,27 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, return -EINVAL; } + switch (cs43130->dais[dai_id].dai_invert) { + case SND_SOC_DAIFMT_NB_NF: + sclk_edge = 1; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + sclk_edge = 0; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_NB_IF: + sclk_edge = 1; + lrck_edge = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + sclk_edge = 0; + lrck_edge = 1; + break; + default: + return -EINVAL; + } + switch (cs43130->dais[dai_id].dai_mode) { case SND_SOC_DAIFMT_CBS_CFS: dai_mode_val = 0; @@ -632,8 +653,6 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, } frm_size = bitwidth_sclk * params_channels(params); - sclk_edge = 1; - lrck_edge = 0; loc_ch1 = 0; loc_ch2 = bitwidth_sclk * (params_channels(params) - 1); @@ -1516,6 +1535,25 @@ static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_NF; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_NF; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_IF; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_IF; + break; + default: + dev_err(cs43130->dev, "Unsupported invert mode 0x%x\n", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index 694286b78d03..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -497,6 +497,7 @@ struct cs43130_dai { unsigned int sclk; unsigned int dai_format; unsigned int dai_mode; + unsigned int dai_invert; }; struct cs43130_private { -- 2.34.1

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[PATCH] soundwire: qcom: set controller id to hw master id
by srinivas.kandagatla＠linaro.org 24 Nov '23

24 Nov '23

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Qualcomm Soundwire Controllers IP version after 1.3 have a dedicated master id register which will provide a unique id value for each controller instance. Use this value instead of artificially generated value from idr. Versions 1.3 and below only have one instance of soundwire controller which does no have this register, so let them use value from idr. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- This patch depends on "[RFC PATCH 0/2] soundwire: introduce controller ID" https://lore.kernel.org/lkml/4815c8d6-635d-415c-9e02-4e39e92a3d6d@sirena.or… patchset. drivers/soundwire/qcom.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 8e027eee8b73..48291fbaf674 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1624,9 +1624,13 @@ static int qcom_swrm_probe(struct platform_device *pdev) } } - /* FIXME: is there a DT-defined value to use ? */ ctrl->bus.controller_id = -1; + if (ctrl->version > SWRM_VERSION_1_3_0) { + ctrl->reg_read(ctrl, SWRM_COMP_MASTER_ID, &val); + ctrl->bus.controller_id = val; + } + ret = sdw_bus_master_add(&ctrl->bus, dev, dev->fwnode); if (ret) { dev_err(dev, "Failed to register Soundwire controller (%d)\n", -- 2.42.0

3 2

[RFC PATCH 0/2] soundwire: introduce controller ID
by Pierre-Louis Bossart 24 Nov '23

24 Nov '23

This patchset is an alternate proposal to the solution suggested in [1], which breaks Intel machine drivers. The only difference is to use a known controller ID instead of an IDA, which wouldn't work with the hard-coded device name. This patchset was tested on Intel and AMD platforms, testing on Qualcomm platforms is required - hence the RFC status. [1] https://lore.kernel.org/alsa-devel/20231004130243.493617-1-krzysztof.kozlow… Krzysztof Kozlowski (1): soundwire: fix initializing sysfs for same devices on different buses Pierre-Louis Bossart (1): soundwire: bus: introduce controller_id drivers/soundwire/amd_manager.c | 8 ++++++++ drivers/soundwire/bus.c | 4 ++++ drivers/soundwire/debugfs.c | 2 +- drivers/soundwire/intel_auxdevice.c | 3 +++ drivers/soundwire/master.c | 2 +- drivers/soundwire/qcom.c | 3 +++ drivers/soundwire/slave.c | 12 ++++++------ include/linux/soundwire/sdw.h | 4 +++- sound/soc/intel/boards/sof_sdw.c | 4 ++-- 9 files changed, 31 insertions(+), 11 deletions(-) -- 2.39.2

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[PATCH] ASoC: fsl_xcvr: refine the requested phy clock frequency
by Shengjiu Wang 23 Nov '23

23 Nov '23

As the input phy clock frequency will divided by 2 by default on i.MX8MP with the implementation of clk-imx8mp-audiomix driver, So the requested frequency need to be updated. The relation of phy clock is: sai_pll_ref_sel sai_pll sai_pll_bypass sai_pll_out sai_pll_out_div2 earc_phy_cg Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 77f8e2394bf9..f0fb33d719c2 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -358,7 +358,7 @@ static int fsl_xcvr_en_aud_pll(struct fsl_xcvr *xcvr, u32 freq) struct device *dev = &xcvr->pdev->dev; int ret; - freq = xcvr->soc_data->spdif_only ? freq / 10 : freq; + freq = xcvr->soc_data->spdif_only ? freq / 5 : freq; clk_disable_unprepare(xcvr->phy_clk); ret = clk_set_rate(xcvr->phy_clk, freq); if (ret < 0) { @@ -409,7 +409,7 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 m_ctl = 0, v_ctl = 0; u32 r = substream->runtime->rate, ch = substream->runtime->channels; - u32 fout = 32 * r * ch * 10 * 2; + u32 fout = 32 * r * ch * 10; int ret = 0; switch (xcvr->mode) { -- 2.34.1

3 2

[PATCH v5] ASoC: cs43130: Allow driver to work without IRQ connection
by Maciej Strozek 23 Nov '23

23 Nov '23

Add a polling mechanism that will keep the driver operational even in absence of physical IRQ connection. If IRQ line is detected, the driver will continue working as usual, in case of missing IRQ line it will fallback to the polling mechanism introduced in this change. This will support users which choose not to connect an IRQ line as it is not critical to part's operation. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- V4 -> V5: Do not use hpload interrupt with the new polling function V3 -> V4: - Removed unnecesary else and indentation - Changed cs43130_wait_for_completion return codes and updated callers accordingly - Changed approach for checking for presence of IRQ connection: see if client->irq equals 0 which seems to be a standard way of checking it used by other drivers (instead of inspecting devm_request_threaded_irq error code) V2 -> V3: Amended changelog message and subject line V1 -> V2: Add changelog message sound/soc/codecs/cs43130.c | 77 ++++++++++++++++++++++++++------------ sound/soc/codecs/cs43130.h | 1 + 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 02e7f45b9208..0f3ead84665f 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,34 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag, ret; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + if (ret == 0) + return -ETIMEDOUT; + else + return 0; // Discard number of jiffies left till timeout and return success + } + + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else { + return -EINVAL; + } + + return regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,14 +392,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -400,14 +427,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -416,14 +442,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for PLL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for PLL_READY interrupt: %d\n", ret); + return ret; } regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, @@ -2053,7 +2078,7 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); - return -1; + return -ETIMEDOUT; } dev_dbg(cs43130->dev, "HP load stat: %x, INT_MASK_4: %x\n", @@ -2548,13 +2573,19 @@ static int cs43130_i2c_probe(struct i2c_client *client) init_completion(&cs43130->pll_rdy); init_completion(&cs43130->hpload_evt); - ret = devm_request_threaded_irq(cs43130->dev, client->irq, - NULL, cs43130_irq_thread, - IRQF_ONESHOT | IRQF_TRIGGER_LOW, - "cs43130", cs43130); - if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + if (!client->irq) { + dev_dbg(cs43130->dev, "IRQ not found, will poll instead\n"); + cs43130->has_irq_line = 0; + } else { + ret = devm_request_threaded_irq(cs43130->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index d3f595bbd3ba..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -508,6 +508,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; -- 2.34.1

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[PATCH v2 0/2] Introduce playback-only/capture-only DAI link
by Daniel Baluta 23 Nov '23

23 Nov '23

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries allows users to specify a link only direction with audio-graph-card2. Changes since v1 - https://lore.kernel.org/lkml/20230801082433.548206-1-daniel.baluta@oss.nxp.… - used audio-graph-card2 instead of simple-card as we are deprecting simple-card. Daniel Baluta (2): ASoC: audio-graph-card2: Introduce playback-only/capture only DAI link flags ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only .../bindings/sound/audio-graph-port.yaml | 6 ++++++ include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 4 files changed, 34 insertions(+) -- 2.25.1

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