June 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [PATCH 2/3] input: Add SW_UNSUPPORT_INSERT define
by bgoswami＠codeaurora.org 22 Aug '19

22 Aug '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Some devices may not support specific type of input devices. For example, when a headset or extension cable with GND/MIC swap is plugged into a headset jack that does not support the headset/cable, it needs to be reported with a corresponding input event. Also, increase the max values for INPUT_DEVICE_ID_SW_MAX and SW_MAX, to accommodate future extension of the number of event. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- include/linux/mod_devicetable.h | 2 +- include/uapi/linux/input-event-codes.h | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 448621c..7586099 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -299,7 +299,7 @@ struct pcmcia_device_id { #define INPUT_DEVICE_ID_LED_MAX 0x0f #define INPUT_DEVICE_ID_SND_MAX 0x07 #define INPUT_DEVICE_ID_FF_MAX 0x7f -#define INPUT_DEVICE_ID_SW_MAX 0x0f +#define INPUT_DEVICE_ID_SW_MAX 0x1f #define INPUT_DEVICE_ID_PROP_MAX 0x1f #define INPUT_DEVICE_ID_MATCH_BUS 1 diff --git a/include/uapi/linux/input-event-codes.h b/include/uapi/linux/input-event-codes.h index 85387c7..960fa86 100644 --- a/include/uapi/linux/input-event-codes.h +++ b/include/uapi/linux/input-event-codes.h @@ -808,7 +808,8 @@ #define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MUTE_DEVICE 0x0e /* set = device disabled */ #define SW_PEN_INSERTED 0x0f /* set = pen inserted */ -#define SW_MAX 0x0f +#define SW_UNSUPPORT_INSERT 0x10 /* set = unsupported device inserted */ +#define SW_MAX 0x1f #define SW_CNT (SW_MAX+1) /* -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

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[alsa-devel] [PATCH v1] ASoC: Intel: Skylake: Print constant literals from format specifier
by Andy Shevchenko 15 Aug '19

15 Aug '19

Instead of using two additional "%s" specifiers, put the constant string literals directly to the format specifier. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/skylake/skl-sst.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 13c636dece56..f3e1399450db 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -421,8 +421,7 @@ static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", - "intel/dsp_fw_", guid, ".bin"); + snprintf(mod_name, sizeof(mod_name), "intel/dsp_fw_%pUL.bin", guid); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { -- 2.20.1

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[alsa-devel] [PATCH v4 0/7] Allwinner H6 SPDIF support
by Clément Péron 12 Aug '19

12 Aug '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Actually all these features are unused and only a bit for flushing the TX fifo is required. Also this series requires the DMA working on H6, a first version has been submitted by Jernej Škrabec[1] but has not been accepted yet. [1] https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 Changes since v3: - rename reg_fctl_ftx to val_fctl_ftx - rebase this series on sound-next - fix dt-bindings due to change in sound-next - change node name sound_spdif to sound-spdif Changes since v2: - Split quirks and H6 support patch - Add specific section for quirks comment Changes since v1: - Remove H3 compatible - Add TX fifo bit flush quirks - Add H6 bindings in SPDIF driver Clément Péron (7): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../sound/allwinner,sun4i-a10-spdif.yaml | 1 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 ++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 49 ++++++++++++++++--- 5 files changed, 87 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v4 0/9]ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 12 Aug '19

12 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. As the LibreELEC project is progressing extremely well then there has been some activity getting surround sound working and this is included. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and some bug fixes that have been discovered along the way. I can see more usage of the tdm property since I last attempted to push these patches and the examples currently in mainline sort of the opposite to what I'm trying to achieve. When we first started looking at the i2s driver, the codecs that we were using allowed for the frame width to be determined based on the sampling resolution but in most use cases it seems that a fixed width is required(my highest priority should be to get HDMI audio support in). We're using the tdm property to override the old way to calculate the frame width. What I've seen in what has already been mainlined is that the i2s driver has a frame width that is fixed to 32 bits and this can be overridden using the tdm property. I still need to investigate the FIFO syncing issues which i've not had a chance to change or address the concerns that broonie and wens brought up. This change has been moved to the top of the patch stack. BR, CK --- v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. Marcus Cooper (9): ASoC: sun4i-i2s: Fix sun8i tx channel offset mask ASoC: sun4i-i2s: Add offset to RX channel select ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Reduce quirks for sun8i-h3 ASoC: sun4i-i2s: Add set_tdm_slot functionality ASoC: sun4i-i2s: Add multi-lane functionality ASoC: sun4i-i2s: Add multichannel functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Adjust regmap settings sound/soc/sunxi/sun4i-i2s.c | 242 ++++++++++++++++++++++++------------ 1 file changed, 164 insertions(+), 78 deletions(-) -- 2.21.0

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[alsa-devel] Patch Update for OpenFrame Sigmatel STAC9202
by Andy Davison 31 Jul '19

31 Jul '19

Hi all, Would anybody on this list have a little time to investigate this patch and bring it up to date for application against the latest 5.1 kernel, or indeed anything being actively developed? https://gist.github.com/andydvsn/c0c159f99bf19d5b30b5e5e156dcac3e This applies correctly against the 2.6.33 kernel, but I I lack the necessary ALSA knowledge and programming skills to make the appropriate corrections for anything more recent. Alternatively, any other advice on where I could seek help with this would be much appreciated. It is the only outstanding kernel issue for this device and would love to have it fixed. All the best, Andy.

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[alsa-devel] [PATCH] ALSA: usb-audio: fix PCM device order
by Alexander Tsoy 27 Jul '19

27 Jul '19

Some cards have alternate setting with non-PCM format as the first altsetting in the interface descriptors. This confuses userspace, since alsa-lib uses device 0 by default. So lets parse interfaces in two steps: 1. Parse altsettings with PCM formats. 2. Parse altsettings with non-PCM formats. This fixes at least following cards: - Audinst HUD-mx2 - Audinst HUD-mini Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/stream.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8e9548bc1f1a..b2ebeae91396 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -477,7 +477,8 @@ static struct uac2_output_terminal_descriptor * return NULL; } -int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no, + bool *has_non_pcm, bool non_pcm) { struct usb_device *dev; struct usb_interface *iface; @@ -634,6 +635,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) iface_no, altno); continue; } + if (fmt->bFormatType != UAC_FORMAT_TYPE_I) + *has_non_pcm = true; + if ((fmt->bFormatType == UAC_FORMAT_TYPE_I) == non_pcm) + continue; if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { dev_err(&dev->dev, @@ -740,3 +745,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return 0; } +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + int err; + bool has_non_pcm = false; + + /* parse PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false); + if (err < 0) + return err; + + if (has_non_pcm) { + /* parse non-PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true); + if (err < 0) + return err; + } + + return 0; +} + -- 2.13.0

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[alsa-devel] Plugged in headphones ignored
by Paul Menzel 27 Jul '19

27 Jul '19

Dear Linux folks, On a Dell OptiPlex 5040 with Linux 5.2-rc6 plugging in a head phone into the front case connector, it is detected just fine and Xfce shows a notification. Then logging out, turning off the monitor connected over DisplayPort at the end of the day, and turning the monitor back on the next morning, logging in, the state is forgotten. I need to unplug the head phone and plug it back in. Please find the output alsa-config.sh for the non-working (1) and replugged (2) situations attached. Is that a Linux kernel problem, or a user space issue? Kind regards, Paul

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[alsa-devel] [PATCH v2] ASoC: SOF: Intel: hda: Make hdac_device device-managed
by Ranjani Sridharan 22 Jul '19

22 Jul '19

snd_hdac_ext_bus_device_exit() has been recently modified to no longer free the hdac device. SOF allocates memory for hdac_device and hda_hda_priv with kzalloc. Make them device-managed instead so that they will be freed when the SOF driver is unloaded. Because of the above change, hda_codec is device-managed and it will be freed when the ASoC device is removed. Freeing the codec in snd_hda_codec_dev_release() leads to kernel panic while unloading and reloading the ASoC driver. So, avoid freeing the hda_codec for ASoC driver. This is done in the same patch to avoid bisect failure. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- sound/soc/sof/intel/hda-codec.c | 6 ++---- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6c51b8363f8b..a2e23d7e768f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev) snd_hda_sysfs_clear(codec); kfree(codec->modelname); kfree(codec->wcaps); - kfree(codec); + + /* + * In the case of ASoC HD-audio, hda_codec is device managed. + * It will be freed when the ASoC device is removed. + */ + if (codec->core.type == HDA_DEV_LEGACY) + kfree(codec); } #define DEV_NAME_LEN 31 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index b8b37f082309..0d8437b080bf 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) address, resp); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL); + hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL); if (!hda_priv) return -ENOMEM; @@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) return 0; #else - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; -- 2.17.1

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