May 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] ASoC: remove two unnecessary gpiolib dependencies
by Pierre-Louis Bossart 20 May '22

20 May '22

Remove two dependencies - issues reported by Intel kernel test bot. Pierre-Louis Bossart (2): ASoC: max98357a: remove dependency on GPIOLIB ASoC: rt1015p: remove dependency on GPIOLIB sound/soc/codecs/Kconfig | 2 -- 1 file changed, 2 deletions(-) -- 2.30.2

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[PATCH v2 1/2] Revert "ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver"
by V sujith kumar Reddy 20 May '22

20 May '22

ASoC: amd : acp : Set Speaker enable/disable pin through rt1019 codec driver. RT1019 codec has two ways of controlling the en_spkr. one way is controlling through gpio pin method the another way is through codec register update through driver. Now Speaker enable/disable is controlled through codec register updated by codec driver. This patch reverts gpio logic. This reverts commit 7fa5c33d043160eba3be9fb8e21588dff2a467c7. Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index c855f50d6b34..fd6299844ebe 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,6 +21,7 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F +#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index bf61a1726f0e..8243765d490f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NONE, + .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

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[PATCH -next] ASoC: codecs: lpass: Fix passing zero to 'PTR_ERR'
by YueHaibing 20 May '22

20 May '22

sound/soc/codecs/lpass-macro-common.c:28 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:38 lpass_macro_pds_init() warn: passing zero to 'PTR_ERR' sound/soc/codecs/lpass-macro-common.c:54 lpass_macro_pds_init() warn: passing zero to 'ERR_PTR' dev_pm_domain_attach_by_name() may return NULL, set 'ret' as -ENODATA to fix this warning. Fixes: 1a8ee4cf8418 ("ASoC: codecs: Fix error handling in power domain init and exit handlers") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/lpass-macro-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 3c661fd61173..1b9082d237c1 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -25,7 +25,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); if (IS_ERR_OR_NULL(l_pds->macro_pd)) { - ret = PTR_ERR(l_pds->macro_pd); + ret = l_pds->macro_pd ? PTR_ERR(l_pds->macro_pd) : -ENODATA; goto macro_err; } @@ -35,7 +35,7 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev) l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { - ret = PTR_ERR(l_pds->dcodec_pd); + ret = l_pds->dcodec_pd ? PTR_ERR(l_pds->dcodec_pd) : -ENODATA; goto dcodec_err; } -- 2.17.1

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topology: plugins: nhlt: fix intel dmic mode calculation
by GitHub pull_request - opened 20 May '22

20 May '22

alsa-project/alsa-utils pull request #148 was opened from juimonen: Dmic find_modes function was used with same sampling rate for both fifo a and b parameter calculations. This bug was found in testing two simultaneous dmic dais in topology. Fix this by introducing a sampling rate argument for find_modes. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/148 Patch URL : https://github.com/alsa-project/alsa-utils/pull/148.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

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[PATCH v1 0/2] ASOC : amd: acp :Add Machine driver changes for
by V sujith kumar Reddy 19 May '22

19 May '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ Signed-off-by: V sujith kumar Reddy <Vsujithkumar.Reddy(a)amd.com> V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 18 +++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ 5 files changed, 263 insertions(+), 17 deletions(-) -- 2.25.1

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[v7] FROMLIST: ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload audio
by Terry Chen 18 May '22

18 May '22

To be able to do driver data for adl_mx98360a_cs4242 which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on ADL platform. Also add the capability to machine driver of creating DAI Link for BT offload. Although BT offload always uses SSP2 port but we reserve the flexibility to assign the port number in macro. Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_cs42l42.c | 86 ++++++++++++++++++- .../intel/common/soc-acpi-intel-adl-match.c | 7 ++ 2 files changed, 89 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index ce78c18798876..56bae2185904b 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -41,8 +41,13 @@ #define SOF_CS42L42_DAILINK_MASK (GENMASK(24, 10)) #define SOF_CS42L42_DAILINK(link1, link2, link3, link4, link5) \ ((((link1) | ((link2) << 3) | ((link3) << 6) | ((link4) << 9) | ((link5) << 12)) << SOF_CS42L42_DAILINK_SHIFT) & SOF_CS42L42_DAILINK_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(25) -#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_SSP_BT_SHIFT 26 +#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) +#define SOF_CS42L42_SSP_BT(quirk) \ + (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(29) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(30) enum { LINK_NONE = 0, @@ -50,6 +55,7 @@ enum { LINK_SPK = 2, LINK_DMIC = 3, LINK_HDMI = 4, + LINK_BT = 5, }; /* Default: SSP2 */ @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + static int create_spk_amp_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -467,9 +480,50 @@ static int create_hdmi_dai_links(struct device *dev, return -ENOMEM; } +static int create_bt_offload_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_bt) +{ + /* bt offload */ + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) + return 0; + + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ssp_bt); + if (!links[*id].name) + goto devm_err; + + links[*id].id = *id; + links[*id].codecs = dummy_component; + links[*id].num_codecs = ARRAY_SIZE(dummy_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_bt); + if (!links[*id].cpus->dai_name) + goto devm_err; + + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, int ssp_codec, int ssp_amp, + int ssp_bt, int dmic_be_num, int hdmi_num) { @@ -522,6 +576,14 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, goto devm_err; } break; + case LINK_BT: + ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); + if (ret < 0) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + break; case LINK_NONE: /* caught here if it's not used as terminator in macro */ default: @@ -543,7 +605,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach; struct sof_card_private *ctx; int dmic_be_num, hdmi_num; - int ret, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp, ssp_codec; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -568,6 +630,9 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + SOF_CS42L42_SSP_BT_SHIFT; + ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; @@ -578,9 +643,11 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) sof_audio_card_cs42l42.num_links++; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, - dmic_be_num, hdmi_num); + ssp_bt, dmic_be_num, hdmi_num); if (!dai_links) return -ENOMEM; @@ -621,6 +688,17 @@ static const struct platform_device_id board_ids[] = { SOF_CS42L42_SSP_AMP(1)) | SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_NONE), }, + { + .name = "adl_mx98360a_cs4242", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1) | + SOF_CS42L42_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_SSP_BT(2)) | + SOF_CS42L42_DAILINK(LINK_HP, LINK_DMIC, LINK_HDMI, LINK_SPK, LINK_BT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7c8cd00457f81..311adeb3afbc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -384,6 +384,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-cs35l41.tplg", }, + { + .id = "10134242", + .drv_name = "adl_mx98360a_cs4242", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

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[PATCH] ALSA: usb-audio: Fix wrong kfree issue in snd_usb_endpoint_free_all
by Wan Jiabing 18 May '22

18 May '22

Fix following coccicheck error: ./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671. Here should be 'cp' rather than 'ip'. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index df5a70013a85..f9c921683948 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1668,7 +1668,7 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) kfree(ip); list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(ip); + kfree(cp); } /* -- 2.36.1

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[PATCH v2 00/15] ASoC: Intel: avs: Driver core and PCM operations
by Cezary Rojewski 17 May '22

17 May '22

Part three of main AVS driver series. This series was originally part of the initial series which was later divided [1] into smaller, easier-to-review chunks. Thus, many patches found here were already present on the list. This series consists of code typical to many drivers - PCI driver operations, trace ability, PM operations - as well as PCM handlers for all standard audio interfaces, that is, HDA, I2S and DMIC are found here. Series starts with updating firmware boot flow - libraries are no longer ignored. This change is dependent on already merged topology code [2] and because of that could not be part of the initial series [1]. PCM operations are split into four changes. First component operations alone i.e. operations which are usually agnostic towards path position (FE/BE). Then it continues with "generic" FE operations - there is no interface split here as from Intel ADSP point of view, FE, or HOST side as it's called in the specs, involves HD-Audio operations only. BE (also known as LINK) side on the other hand is divided into "non-HD-Audio" and HD-Audio part. The former represents transfer over DMIC and I2S interfaces both. While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: While patches implementing standard PCI driver operations along (again standard) HD-Audio initialization routines followed up by power management handlers are two major ones, series covers also other important subjects such as: - event tracing - preparation for firmware tracing (debugability) - coredump (debugability) - recovery flow (attempt recovery after IPC timeout or exception) - D0ix (D0 device substate, complements standard power management) Series is finalized by actual addition of supported platforms: SKL and APL-based. Platform-specific files are limited to firmware-specific bits, that is, bits that are specific to given firmware generation. Everything else is shared and is part of already upstream messaging code found in ipc.c, messages.c and messages.h files. Changes in v2: - usage of avs_releast_last_firmware() dropped in error path for library loading procedure as suggested by Pierre - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre. Existing code addressed with new patch: "ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask" - fixed possible race during recovery flow (->recovering flag is now atomic and tested in single location only). - dropped platform prefixes for basically all i2s board descriptors - 'ssp_test' renamed to 'i2s_test' to match naming convention of other boards - simplified PM implementation for current series, 'low_power' bits moved to future series - replaced SND_INTEL_DSP_DRIVER_SST with _AVS as suggested by Mark. Required changes to intel-dspcfg will be added in future series - number of typos across commit messages addressed [1]: https://lore.kernel.org/all/20220311153544.136854-1-cezary.rojewski@intel.c… [2]: https://lore.kernel.org/all/20220331135246.993089-1-cezary.rojewski@intel.c… Cezary Rojewski (14): ASoC: Intel: avs: Account for libraries when booting basefw ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support Piotr Maziarz (1): ASoC: Intel: avs: Replace link_mask usage with i2s_link_mask include/sound/intel-dsp-config.h | 3 +- include/sound/soc-acpi.h | 2 + sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 7 +- sound/soc/intel/avs/apl.c | 250 ++++++ sound/soc/intel/avs/avs.h | 79 ++ sound/soc/intel/avs/board_selection.c | 501 +++++++++++ sound/soc/intel/avs/core.c | 631 +++++++++++++ sound/soc/intel/avs/dsp.c | 27 +- sound/soc/intel/avs/ipc.c | 253 +++++- sound/soc/intel/avs/loader.c | 84 ++ sound/soc/intel/avs/messages.c | 35 +- sound/soc/intel/avs/messages.h | 51 ++ sound/soc/intel/avs/pcm.c | 1182 +++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 8 + sound/soc/intel/avs/skl.c | 125 +++ sound/soc/intel/avs/topology.c | 14 +- sound/soc/intel/avs/trace.c | 33 + sound/soc/intel/avs/trace.h | 158 ++++ sound/soc/intel/avs/utils.c | 23 + 20 files changed, 3447 insertions(+), 21 deletions(-) create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h -- 2.25.1

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Re: Re: [PATCH] conf: vc4-hdmi: use a proper hdmi pcm, fix broken default pcm
by popcorn mix 17 May '22

17 May '22

Matthias Reichl wrote: > Userspace expects to see a HDMI pcm, so remove the somewhat broken > front and iec958 pcms and add a proper hdmi pcm instead. > > The hdmi pcm supports HD/HBR audio passthrough and relays the IEC958 > status bits on to the kernel driver so it can switch to HBR audio > packets if needed. The control hook is marked optional because kernels > before 5.14 didn't expose the IEC958 controls. > > The default pcm never worked as dmix doesn't support the iec958 format. > So drop dmix, only use plug and softvol, and use the hdmi pcm for > iec958 formatting. I can confirm the upstream cards/vc4-hdmi.conf is not usable, and RPiOS ships a patched version (which was an earlier version of this patch). I've tested this version with RPiOS bullseye and all seems good. I could play stereo and multi-channel PCM from VLC. I could also play DTS, DTS-HD and TrueHD formats as passthrough using kodi. It would be good if we can switch from a custom version of vc4-hdmi.conf to an upstream version in RPiOS when this is merged. Tested-by: Dom Cobley <popcornmix(a)gmail.com>

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