March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codec: Omit superfluous error message in jz4760_codec_probe()
by Tang Bin 02 Mar '21

02 Mar '21

The function devm_platform_ioremap_resource has already contained error message, so remove the redundant dev_err here. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/jz4760.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index e8f28ccc145a..c2d8a107f159 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -843,7 +843,6 @@ static int jz4760_codec_probe(struct platform_device *pdev) codec->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(codec->base)) { ret = PTR_ERR(codec->base); - dev_err(dev, "Failed to ioremap mmio memory: %d\n", ret); return ret; } -- 2.18.2

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[PATCH] ASoC: wm8960: Remove bitclk relax condition
by Shengjiu Wang 02 Mar '21

02 Mar '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> Using a higher bitclk then expected doesn't always work. Here is an example: aplay -Dhw:0,0 -d 5 -r 48000 -f S24_LE -c 2 audio48k24b2c.wav In this case, the required bitclk is 48000 * 24 * 2 = 2304000 but the closest bitclk that can be derived is 3072000. Since the clock is faster than expected, it will start to send bytes from the next channel so the sound will be corrupted. Fixes: 82bab88910ee ("ASoC: codec: wm8960: Relax bit clock computation when using PLL") Fixes: 3c01b9ee2ab9 ("ASoC: codec: wm8960: Relax bit clock computation") Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index df351519a3a6..368bec1dfbdb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -608,10 +608,6 @@ static const int bclk_divs[] = { * - lrclk = sysclk / dac_divs * - 10 * bclk = sysclk / bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @wm8960: codec private data * @mclk: MCLK used to derive sysclk * @sysclk_idx: sysclk_divs index for found sysclk @@ -629,7 +625,7 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, { int sysclk, bclk, lrclk; int i, j, k; - int diff, closest = mclk; + int diff; /* marker for no match */ *bclk_idx = -1; @@ -653,12 +649,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, *bclk_idx = k; break; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - } } if (k != ARRAY_SIZE(bclk_divs)) break; @@ -676,10 +666,6 @@ int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk, * - freq_out = sysclk * sysclk_divs * - 10 * sysclk = bclk * bclk_divs * - * If we cannot find an exact match for (sysclk, lrclk, bclk) - * triplet, we relax the bclk such that bclk is chosen as the - * closest available frequency greater than expected bclk. - * * @component: component structure * @freq_in: input frequency used to derive freq out via PLL * @sysclk_idx: sysclk_divs index for found sysclk @@ -697,14 +683,12 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, { struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); int sysclk, bclk, lrclk, freq_out; - int diff, closest, best_freq_out; + int diff; int i, j, k; bclk = wm8960->bclk; lrclk = wm8960->lrclk; - closest = freq_in; - best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { @@ -725,18 +709,11 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, *bclk_idx = k; return freq_out; } - if (diff > 0 && closest > diff) { - *sysclk_idx = i; - *dac_idx = j; - *bclk_idx = k; - closest = diff; - best_freq_out = freq_out; - } } } } - return best_freq_out; + return -EINVAL; } static int wm8960_configure_clocking(struct snd_soc_component *component) { -- 2.27.0

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[PATCH] ALSA: hda/realtek: Add quirk for Intel NUC 10
by Werner Sembach 02 Mar '21

02 Mar '21

From: Werner Sembach <wse(a)tuxedocomputers.com> ALSA: hda/realtek: Add quirk for Intel NUC 10 This adds a new SND_PCI_QUIRK(...) and applies it to the Intel NUC 10 devices. This fixes the issue of the devices not having audio input and output on the headset jack because the kernel does not recognize when something is plugged in. The new quirk was inspired by the quirk for the Intel NUC 8 devices, but it turned out that the NUC 10 uses another pin. This information was acquired by black box testing likely pins. Co-developed-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- Hi, this is my second ever submitted kernel patch with the first one send just some minutes ago, feel free to criticise me if I made an error or missed a best practice bullet point. Also: I'm unsure if this would fit the requirements for stable(a)vger.kernel.org and/or trivial(a)kernel.org, but I think not (correct me if I'm wrong). Kind regards Werner Sembach >From d281364b8ca6c054a0e5ce20caa599bf7d08160d Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Fri, 26 Feb 2021 13:54:30 +0100 Subject: [PATCH] Fix Intel NUC10 no output and input on headset jack --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..c14d624dbaf1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6362,6 +6362,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, @@ -7744,6 +7745,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8183,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. -- 2.25.1

1 0

[PATCH] ALSA: hda: intel-nhlt: verify config type
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

Multiple bug reports report issues with the SOF and SST drivers when dealing with single microphone cases. We currently read the DMIC array information unconditionally but we don't check that the configuration type is actually a mic array. When the DMIC link does not rely on a mic array configuration, the recommendation is to check the format information to infer the maximum number of channels, and map this to the number of microphones. This leaves a potential for a mismatch between actual microphones available in hardware and what the ACPI table contains, but we have no other source of information. Note that single microphone configurations can alternatively be handled with a 'mic array' configuration along with a 'vendor-defined' geometry. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=201251 BugLink: https://github.com/thesofproject/linux/issues/2725 Fixes: 7a33ea70e1868 ('ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/intel-nhlt.h | 5 ++++ sound/hda/intel-nhlt.c | 54 +++++++++++++++++++++++++++++++------- 2 files changed, 50 insertions(+), 9 deletions(-) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index 743c2f442280..d0574805865f 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -112,6 +112,11 @@ struct nhlt_vendor_dmic_array_config { /* TODO add vendor mic config */ } __packed; +enum { + NHLT_CONFIG_TYPE_GENERIC = 0, + NHLT_CONFIG_TYPE_MIC_ARRAY = 1 +}; + enum { NHLT_MIC_ARRAY_2CH_SMALL = 0xa, NHLT_MIC_ARRAY_2CH_BIG = 0xb, diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 059aaf04f536..d053beccfaec 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -31,18 +31,44 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) struct nhlt_endpoint *epnt; struct nhlt_dmic_array_config *cfg; struct nhlt_vendor_dmic_array_config *cfg_vendor; + struct nhlt_fmt *fmt_configs; unsigned int dmic_geo = 0; - u8 j; + u16 max_ch = 0; + u8 i, j; if (!nhlt) return 0; - epnt = (struct nhlt_endpoint *)nhlt->desc; + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { - for (j = 0; j < nhlt->endpoint_count; j++) { - if (epnt->linktype == NHLT_LINK_DMIC) { - cfg = (struct nhlt_dmic_array_config *) - (epnt->config.caps); + if (epnt->linktype != NHLT_LINK_DMIC) + continue; + + cfg = (struct nhlt_dmic_array_config *)(epnt->config.caps); + fmt_configs = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); + + /* find max number of channels based on format_configuration */ + if (fmt_configs->fmt_count) { + dev_dbg(dev, "%s: found %d format definitions\n", + __func__, fmt_configs->fmt_count); + + for (i = 0; i < fmt_configs->fmt_count; i++) { + struct wav_fmt_ext *fmt_ext; + + fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; + + if (fmt_ext->fmt.channels > max_ch) + max_ch = fmt_ext->fmt.channels; + } + dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + } else { + dev_dbg(dev, "%s: No format information found\n", __func__); + } + + if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { + dmic_geo = max_ch; + } else { switch (cfg->array_type) { case NHLT_MIC_ARRAY_2CH_SMALL: case NHLT_MIC_ARRAY_2CH_BIG: @@ -59,13 +85,23 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) dmic_geo = cfg_vendor->nb_mics; break; default: - dev_warn(dev, "undefined DMIC array_type 0x%0x\n", - cfg->array_type); + dev_warn(dev, "%s: undefined DMIC array_type 0x%0x\n", + __func__, cfg->array_type); + } + + if (dmic_geo > 0) { + dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + } + if (max_ch > dmic_geo) { + dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", + __func__, max_ch, dmic_geo); } } - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); } + dev_dbg(dev, "%s: dmic number %d max_ch %d\n", + __func__, dmic_geo, max_ch); + return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); -- 2.25.1

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[PATCH v2] ALSA: hda: fix kernel-doc warnings
by Pierre-Louis Bossart 02 Mar '21

02 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for snd_hda_input_mux_info_info(). Prototype was for snd_hda_input_mux_info() instead sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for snd_hda_input_mux_info_put(). Prototype was for snd_hda_input_mux_put() instead sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for _snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead sound/pci/hda/hda_jack.c:223: warning: expecting prototype for snd_hda_set_dirty_all(). Prototype was for snd_hda_jack_set_dirty_all() instead sound/pci/hda/hda_jack.c:309: warning: expecting prototype for snd_hda_jack_detect_enable_mst(). Prototype was for snd_hda_jack_detect_enable_callback_mst() instead sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for snd_dha_gen_add_mute_led_cdev(). Prototype was for snd_hda_gen_add_mute_led_cdev() instead sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for snd_dha_gen_add_micmute_led_cdev(). Prototype was for snd_hda_gen_add_micmute_led_cdev() instead sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for Prepare and send the SCP message to DSP(). Prototype was for dspio_scp() instead sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for Allocate router ports(). Prototype was for dsp_allocate_router_ports() instead sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for Write a block of data into DSP code or data RAM using pre(). Prototype was for dspxfr_one_seg() instead sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for data overlay to DSP memories(). Prototype was for dspxfr_image() instead sound/hda/hdac_regmap.c:393: warning: expecting prototype for snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit() instead sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype for snd_hdac_ext_bus_get_link_index(). Prototype was for snd_hdac_ext_bus_get_link() instead sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for snd_hdac_ext_linkstream_start(). Prototype was for snd_hdac_ext_link_stream_start() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 2 +- sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_regmap.c | 2 +- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 9 ++++----- 7 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a9bd39b93697..b2df7b4f9227 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -133,7 +133,7 @@ void snd_hdac_link_free_all(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** - * snd_hdac_ext_bus_get_link_index - get link based on codec name + * snd_hdac_ext_bus_get_link - get link based on codec name * @bus: the pointer to HDAC bus object * @codec_name: codec name */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c4d54a838773..0c005d67fa89 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -133,7 +133,7 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** - * snd_hdac_ext_linkstream_start - start a stream + * snd_hdac_ext_link_stream_start - start a stream * @stream: HD-audio ext core stream to start */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78..fe3587547cfe 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -386,7 +386,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); /** - * snd_hdac_regmap_init - Release the regmap from HDA codec + * snd_hdac_regmap_exit - Release the regmap from HDA codec * @codec: the codec object */ void snd_hdac_regmap_exit(struct hdac_device *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..2026f1ccaf5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3483,7 +3483,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); */ /** - * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * snd_hda_input_mux_info - Info callback helper for the input-mux enum * @imux: imux helper object * @uinfo: pointer to get/store the data */ @@ -3506,7 +3506,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** - * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * snd_hda_input_mux_put - Put callback helper for the input-mux enum * @codec: the HDA codec * @imux: imux helper object * @ucontrol: pointer to get/store the data @@ -3941,7 +3941,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); /** - * _snd_hda_pin_ctl - Helper to set pin ctl value + * _snd_hda_set_pin_ctl - Helper to set pin ctl value * @codec: the HDA codec * @pin: referred pin NID * @val: pin control value to set diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5e40944e7342..8b7c5508f368 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3923,7 +3923,7 @@ static void vmaster_update_mute_led(void *private_data, int enabled) } /** - * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED + * snd_hda_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking */ @@ -4074,7 +4074,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) } /** - * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED + * snd_hda_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking * diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..ac00866d8032 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -213,7 +213,7 @@ static void jack_detect_update(struct hda_codec *codec, } /** - * snd_hda_set_dirty_all - Mark all the cached as dirty + * snd_hda_jack_set_dirty_all - Mark all the cached as dirty * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. @@ -293,7 +293,7 @@ find_callback_from_list(struct hda_jack_tbl *jack, } /** - * snd_hda_jack_detect_enable_mst - enable the jack-detection + * snd_hda_jack_detect_enable_callback_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 7e62aed172a9..c966f49fa942 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2338,7 +2338,7 @@ static int dspio_send_scp_message(struct hda_codec *codec, } /** - * Prepare and send the SCP message to DSP + * dspio_scp - Prepare and send the SCP message to DSP * @codec: the HDA codec * @mod_id: ID of the DSP module to send the command * @src_id: ID of the source @@ -2865,7 +2865,7 @@ static int dsp_dma_stop(struct hda_codec *codec, } /** - * Allocate router ports + * dsp_allocate_router_ports - Allocate router ports * * @codec: the HDA codec * @num_chans: number of channels in the stream @@ -3178,8 +3178,7 @@ static int dspxfr_hci_write(struct hda_codec *codec, } /** - * Write a block of data into DSP code or data RAM using pre-allocated - * DMA engine. + * dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -3376,7 +3375,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, } /** - * Write the entire DSP image of a DSP code/data overlay to DSP memories + * dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories * * @codec: the HDA codec * @fls_data: pointer to a fast load image -- 2.25.1

2 1

[PATCH 0/4]: ALSA: usb-audio: Add Pioneer DJM-850 support
by Nicolas MURE 02 Mar '21

02 Mar '21

Hello, This patches serie adds support for the Pioneer DJM-850 mixer. The mixer has a 24bits (little endian) sound card, which features 4 stereo channels for capture and 4 stereo channels for playback. The mixer also has controls to specify the capture source for each channels, and also a control to set the sample rate (44.1kHz, 48kHz, 96kHz). Thank you :) Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com>

2 11

[PATCH] ASoC: es8316: Simplify adc_pga_gain_tlv table
by Hans de Goede 02 Mar '21

02 Mar '21

Most steps in this table are steps of 3dB (300 centi-dB), so we can simplify the table. This not only reduces the amount of space it takes inside the kernel, this also makes alsa-lib's mixer code actually accept the table, where as before this change alsa-lib saw the "ADC PGA Gain" control as a control without a dB scale. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/es8316.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 1ce84f12a4cf..9ab973958fc7 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, -- 2.30.1

2 1

[PATCH] ASoC: Intel: boards: sof-wm8804: add check for PLL setting
by Colin King 02 Mar '21

02 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> Currently the return from snd_soc_dai_set_pll is not checking for failure, this is the only driver in the kernel that ignores this, so it probably should be added for sake of completeness. Fix this by adding an error return check. Addresses-Coverity: ("Unchecked return value") Fixes: f139546fb7d4 ("ASoC: Intel: boards: sof-wm8804: support for Hifiberry Digiplus boards") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/sof_wm8804.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index a46ba13e8eb0..6a181e45143d 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -124,7 +124,11 @@ static int sof_wm8804_hw_params(struct snd_pcm_substream *substream, } snd_soc_dai_set_clkdiv(codec_dai, WM8804_MCLK_DIV, mclk_div); - snd_soc_dai_set_pll(codec_dai, 0, 0, sysclk, mclk_freq); + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, sysclk, mclk_freq); + if (ret < 0) { + dev_err(rtd->card->dev, "Failed to set WM8804 PLL\n"); + return ret; + } ret = snd_soc_dai_set_sysclk(codec_dai, WM8804_TX_CLKSRC_PLL, sysclk, SND_SOC_CLOCK_OUT); -- 2.30.0

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[PATCH 0/5] AsoC: rt5640/rt5651: Volume control fixes
by Hans de Goede 02 Mar '21

02 Mar '21

Hi All, Here is a series of rt5640/rt5651 volume-control fixes which I wrote while working on a bytcr-rt5640 UCM profile patch-series adding hardware-volume control to devices using this UCM profile. The UCM series will also work on older kernels, but it works best on kernels with this series applied, giving e.g. finer grained volume control and support for hardware muting the outputs. Regards, Hans Hans de Goede (5): ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5640: Add emulated 'DAC1 Playback Switch' control ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' ASoC: Intel: bytcr_rt5640: Add used AIF to the components string sound/soc/codecs/rt5640.c | 106 +++++++++++++++++++++++--- sound/soc/codecs/rt5640.h | 4 + sound/soc/codecs/rt5651.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 11 ++- 4 files changed, 111 insertions(+), 14 deletions(-) -- 2.30.1

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[PATCH] sound: soc/uniphier: Simplify the return expression of uniphier_aio_startup
by dingsenjie＠163.com 02 Mar '21

02 Mar '21

From: dingsenjie <dingsenjie(a)yulong.com> Simplify the return expression in the aio-cpu.c. Signed-off-by: dingsenjie <dingsenjie(a)yulong.com> --- sound/soc/uniphier/aio-cpu.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 25c40c2..cf98141 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -256,17 +256,12 @@ static int uniphier_aio_startup(struct snd_pcm_substream *substream, { struct uniphier_aio *aio = uniphier_priv(dai); struct uniphier_aio_sub *sub = &aio->sub[substream->stream]; - int ret; sub->substream = substream; sub->pass_through = 0; sub->use_mmap = true; - ret = aio_init(sub); - if (ret) - return ret; - - return 0; + return aio_init(sub); } static void uniphier_aio_shutdown(struct snd_pcm_substream *substream, -- 1.9.1

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