July 2020 - Alsa-devel - mailman.alsa-project.org

Move ownership of snd-firewire-ctl-services repository
by Takashi Sakamoto 05 Jul '20

05 Jul '20

Hi Jaroslav, I'm currently working for the idea of 'ALSA control service in userspace'[1] for audio and music units on IEEE 1394 bus under my personal repository[2], and the work mostly done for Digidesign Digi 002/003 family and Tascam FireWire series[3]. As well as libhinawa and alsa-gobject repositories[4], I'd like to move ownership of the repository to 'GObject Introspection' team of alsa-project account, then continue the work. I'd like you to arrange access rights and so on, again. Besides, as I tols you in previous message[5], the nested team is not good in reasons. I strongly request you to leave 'GObject Introspection' team and drop 'Hinawa' team, again. Would I expect your arrangement for it? [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [2] https://github.com/takaswie/snd-firewire-ctl-services/ [3] https://github.com/takaswie/snd-firewire-ctl-services/tree/topic/services-f… [4] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [5] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169191.html Thanks Takashi Sakamoto

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Some questions about asoc codec mixer elements naming
by Станислав Александров 05 Jul '20

05 Jul '20

Hello. I am writing an asoc codec driver and stuck with mixer and output naming. Could someone clarify a few things? This codec has two playback PGAs and 4 outputs: PGA1 -> [Speaker,Headphone,Line Out] PGA2 -> [Mono Output] [PGA1] controls volume and mute for all three outputs. [Speaker,Headphone,Line Out] have only mute. [PGA2] controls volume and mute. [Mono Output] also has mute. Codec is bound to platform driver via simple-audio-card: simple-audio-card,widgets = "Headphone", "Headphones", "Speaker", "Speakers", "Line", "Line Out Jack"; simple-audio-card,routing = "Headphones", "HPOUT", "Speakers", "SPKOUT", "Line Out Jack", "LINEOUT"; Is it possible to name mixer controls so that no further configuration is required when the driver is loaded? i.e. pulseaudio automatically detects all these outputs and controls, correctly maps them and allows switching between them? Currently i have this configuration(simplified): SOC_DOUBLE_S8_TLV("Master Playback Volume", PGA1, ...) SOC_DOUBLE("Master Playback Switch", PGA1, ...) SOC_SINGLE("Speaker Playback Switch", ...) SOC_SINGLE("Headphone Playback Switch", ...) SOC_SINGLE("Line Out Playback Switch", ...) SOC_SINGLE_S8_TLV("Mono Output Playback Volume", PGA2...) SOC_SINGLE("Mono Output Playback Switch", PGA2....) With these settings pulseaudio only detects Speaker and Headphone (but not Line Out because it wants to find its Jack) But as I understand "Master Playback" assumes control for all sound streams, so this is probably not correct =\ If it is impossible to name those elements that way so everything is autodetected, what should i use then? Write a machine audio driver or use alsaucm?

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Asus Xonar SE
by Jefferson Smith Pizarro Gutierrez 05 Jul '20

05 Jul '20

hi, i have issues with my Asus Xonar SE souncard, i need to get a solution, please helpme to get sound through the spdif of the soundcard. i dont have sound

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Looking for guidance for porting home made audio driver to ASoC/FSL
by Christophe Leroy 04 Jul '20

04 Jul '20

Hello I'm looking for guidance for porting a home made audio driver to ASoC/FSL on a custom board. The board has an MPC8321 freescale/NXP cpu. The board has a TDM bus connected to one of the cpu TDM port. On that TDM bus, there are three IDT 821034 quad codecs (PCM G.711) and an E1 chip that carries IP over E1. Today, there is a home made audio driver providing raw read/write devices plus an home made WAN HDLC driver for the E1. The CPU Quicc Engine (QE) is configured to connect the TDM port to one UCC. QUICC Multi-Channel Controller (QMC) is used to distribute E1 timeslots to the HDLC driver and each of the 12 codec timeslots to the audio driver. (The QMC is able to manage two types of channels: transparent and HDLC). We would like to make all this more standard and port it to use ALSA ASoC for the audio part, and the Freescale Quicc Engine HDLC. As of today, the Freescale QE HDLC wan driver uses the UCC directly, it doesn't interface through the QMC. And the FSL ASoC drivers seem to neither handle QE TDM via QE QMC nor QE UCC. We are thinking about adding a QE QMC layer in driver/soc/fsl/ which would more or less present to the consumer the same kind of API as the FSL UCC layer, and enhance the WAN FSL HDLC driver to be able to use that new layer. For the ALSA ASoC part, could some parts of existing FSL ASoC drivers be re-used of would we have to start implementing an ASoC from scratch ? Any recommendation and/or guidance would be welcomed and appreciated. Thanks in advance Christophe

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[PATCH] ASoC: amd: fix Kconfig warning for Renoir
by Randy Dunlap 04 Jul '20

04 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> SND_SOC_DMIC depends on GPIOLIB so this driver should also depend on GPIOLIB to make kconfig happy. WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] Fixes: 1d3776669323 ("ASoC: amd: enable build for RN machine driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> --- Found in mmotm. There are no object file build errors. I suppose that gpiolib stubs take care of that. Maybe some other patch would be more appropriate. sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) --- mmotm-2020-0703-1503.orig/sound/soc/amd/Kconfig +++ mmotm-2020-0703-1503/sound/soc/amd/Kconfig @@ -47,5 +47,6 @@ config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" select SND_SOC_DMIC depends on SND_SOC_AMD_RENOIR + depends on GPIOLIB help This option enables machine driver for DMIC

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[PATCH v2] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 04 Jul '20

04 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 54 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..4105ef2c4f99 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,55 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +649,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +662,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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[PATCH v2 0/2] ASoC: rt5670: 2 small cleanups
by Hans de Goede 03 Jul '20

03 Jul '20

Hi All, Here is in essence a resend of my 2 cleanup patches for the rt5670 ASoC codec code, rebased on top of broonie/sound/for-5.9 with broonie/sound/for-5.8 merged in. Regards, Hans p.s. I'll also send out the patch improving the comment in cht_bsw_rt5672.c cht_codec_fixup() which Pierre-Louis requested soon.

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[PATCH v2 0/6] ASoC: fsl: fix kernel-doc
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

This cleanup patchset is only about kernel-doc, mostly trivial edits and format correction. v2: added Nicolin Chen's Acked-by tags and included the three suggested edits. Pierre-Louis Bossart (6): ASoC: fsl: fsl_ssi_dbg: remove spurious kernel-doc comment start ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl-asoc-card: fix kernel-doc ASoC: fsl: fsl_spdif: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: fsl: fsl_esai: fix kernel-doc sound/soc/fsl/fsl-asoc-card.c | 21 +++++------ sound/soc/fsl/fsl_asrc.c | 57 +++++++++++++++++++--------- sound/soc/fsl/fsl_esai.c | 32 +++++++++------- sound/soc/fsl/fsl_spdif.c | 5 ++- sound/soc/fsl/fsl_ssi.c | 70 ++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 +- 6 files changed, 115 insertions(+), 74 deletions(-) -- 2.25.1

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[RESEND PATCH V3] ASoC: amd: add logic to check dmic hardware runtime
by Vijendar Mukunda 03 Jul '20

03 Jul '20

Add logic to check DMIC hardware exists or not on the platform at runtime. Add module param for overriding DMIC hardware check at runtime. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v3: changed dmic_acpi_check module param as bint v2: changed module param usage -1 as default, 0 - skip, 1 - force sound/soc/amd/renoir/rn-pci-acp3x.c | 29 +++++++++++++++++++++++++++++ sound/soc/amd/renoir/rn_acp3x.h | 2 ++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 859ed67..d1faea5 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -5,6 +5,7 @@ //Copyright 2020 Advanced Micro Devices, Inc. #include <linux/pci.h> +#include <linux/acpi.h> #include <linux/module.h> #include <linux/io.h> #include <linux/delay.h> @@ -18,6 +19,16 @@ static int acp_power_gating; module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); +/** + * dmic_acpi_check = -1 - Checks ACPI method to know DMIC hardware status runtime + * = 0 - Skips the DMIC device creation and returns probe failure + * = 1 - Assumes that platform has DMIC support and skips ACPI + * method check + */ +static int dmic_acpi_check = ACP_DMIC_AUTO; +module_param(dmic_acpi_check, bint, 0644); +MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime"); + struct acp_dev_data { void __iomem *acp_base; struct resource *res; @@ -157,6 +168,8 @@ static int snd_rn_acp_probe(struct pci_dev *pci, { struct acp_dev_data *adata; struct platform_device_info pdevinfo[ACP_DEVS]; + acpi_handle handle; + acpi_integer dmic_status; unsigned int irqflags; int ret, index; u32 addr; @@ -201,6 +214,22 @@ static int snd_rn_acp_probe(struct pci_dev *pci, if (ret) goto disable_msi; + if (!dmic_acpi_check) { + ret = -ENODEV; + goto de_init; + } else if (dmic_acpi_check == ACP_DMIC_AUTO) { + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (ACPI_FAILURE(ret)) { + ret = -EINVAL; + goto de_init; + } + if (!dmic_status) { + ret = -ENODEV; + goto de_init; + } + } + adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource) * 2, GFP_KERNEL); diff --git a/sound/soc/amd/renoir/rn_acp3x.h b/sound/soc/amd/renoir/rn_acp3x.h index 75228e3..1462039 100644 --- a/sound/soc/amd/renoir/rn_acp3x.h +++ b/sound/soc/amd/renoir/rn_acp3x.h @@ -55,6 +55,8 @@ #define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER +#define ACP_DMIC_AUTO -1 + struct pdm_dev_data { u32 pdm_irq; void __iomem *acp_base; -- 2.7.4

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[PATCH -next] ASoC: ti: j721e-evm: Fix missing unlock on error in j721e_audio_hw_params()
by Wei Yongjun 03 Jul '20

03 Jul '20

Add the missing unlock before return from function j721e_audio_hw_params() in the error handling case. Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 3a2a8b1f3aa3..174306cf53ad 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -330,7 +330,7 @@ static int j721e_audio_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, slot_width); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } ret = j721e_configure_refclk(priv, domain_id, params_rate(params));

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