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

[PATCH] sound: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 20 Feb '20

20 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertently introduced[3] to the codebase from now on. Also, notice that, dynamic memory allocations won't be affected by this change: "Flexible array members have incomplete type, and so the sizeof operator may not be applied. As a quirk of the original implementation of zero-length arrays, sizeof evaluates to zero."[1] This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- include/sound/control.h | 2 +- include/sound/intel-nhlt.h | 6 +++--- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/asound.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 +- sound/core/oss/pcm_plugin.h | 2 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/soc/intel/atom/sst-atom-controls.h | 2 +- sound/soc/intel/skylake/skl-i2s.h | 2 +- sound/soc/intel/skylake/skl-topology.h | 4 ++-- sound/soc/intel/skylake/skl.h | 2 +- sound/usb/usx2y/usbusx2y.h | 2 +- 13 files changed, 16 insertions(+), 16 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 11feeee31e35..aeaed2a05bae 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -75,7 +75,7 @@ struct snd_kcontrol { unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); - struct snd_kcontrol_volatile vd[0]; /* volatile data */ + struct snd_kcontrol_volatile vd[]; /* volatile data */ }; #define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index f657fd8fc0ad..743c2f442280 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -50,7 +50,7 @@ enum nhlt_device_type { struct nhlt_specific_cfg { u32 size; - u8 caps[0]; + u8 caps[]; } __packed; struct nhlt_fmt_cfg { @@ -60,7 +60,7 @@ struct nhlt_fmt_cfg { struct nhlt_fmt { u8 fmt_count; - struct nhlt_fmt_cfg fmt_config[0]; + struct nhlt_fmt_cfg fmt_config[]; } __packed; struct nhlt_endpoint { @@ -80,7 +80,7 @@ struct nhlt_endpoint { struct nhlt_acpi_table { struct acpi_table_header header; u8 endpoint_count; - struct nhlt_endpoint desc[0]; + struct nhlt_endpoint desc[]; } __packed; struct nhlt_resource_desc { diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1a9c5dd40228..f870f927b70c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -688,7 +688,7 @@ struct snd_soc_dapm_context { /* A list of widgets associated with an object, typically a snd_kcontrol */ struct snd_soc_dapm_widget_list { int num_widgets; - struct snd_soc_dapm_widget *widgets[0]; + struct snd_soc_dapm_widget *widgets[]; }; #define for_each_dapm_widgets(list, i, widget) \ diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..9944b978ae70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1146,7 +1146,7 @@ struct snd_soc_pcm_runtime { unsigned int fe_compr:1; /* for Dynamic PCM */ int num_components; - struct snd_soc_component *components[0]; /* CPU/Codec/Platform */ + struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; #define for_each_rtd_components(rtd, i, component) \ for ((i) = 0; \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..3809993f6cb4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -1071,7 +1071,7 @@ struct snd_ctl_elem_value { struct snd_ctl_tlv { unsigned int numid; /* control element numeric identification */ unsigned int length; /* in bytes aligned to 4 */ - unsigned int tlv[0]; /* first TLV */ + unsigned int tlv[]; /* first TLV */ }; #define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 9eee32f5e407..62f76f154d19 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -149,7 +149,7 @@ struct skl_dfw_algo_data { __u32 rsvd:30; __u32 param_id; __u32 max; - char params[0]; + char params[]; } __packed; enum skl_tkn_dir { diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 8d2f7a4e3ab6..46e273bd4a78 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -64,7 +64,7 @@ struct snd_pcm_plugin { char *buf; snd_pcm_uframes_t buf_frames; struct snd_pcm_plugin_channel *buf_channels; - char extra_data[0]; + char extra_data[]; }; int snd_pcm_plugin_build(struct snd_pcm_substream *handle, diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index dda797209a27..654e28a6669f 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -177,7 +177,7 @@ struct snd_efw_phys_meters { u32 in_meters; u32 reserved4; u32 reserved5; - u32 values[0]; + u32 values[]; } __packed; enum snd_efw_clock_source { SND_EFW_CLOCK_SOURCE_INTERNAL = 0, diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index 5356e954a732..620b48d2a064 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -410,7 +410,7 @@ struct sst_cmd_set_gain_dual { struct sst_cmd_set_params { struct sst_destination_id dst; u16 command_id; - char params[0]; + char params[]; } __packed; diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h index d7c15873c0d4..dfce91e11be1 100644 --- a/sound/soc/intel/skylake/skl-i2s.h +++ b/sound/soc/intel/skylake/skl-i2s.h @@ -46,7 +46,7 @@ struct skl_i2s_config_mclk { struct skl_i2s_config_mclk_ext { u32 mdivctrl; u32 mdivr_count; - u32 mdivr[0]; + u32 mdivr[]; } __packed; struct skl_i2s_config_blob_signature { diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..d2cd8ef8e97f 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -119,7 +119,7 @@ struct skl_cpr_gtw_cfg { struct skl_dma_control { u32 node_id; u32 config_length; - u32 config_data[0]; + u32 config_data[]; } __packed; struct skl_cpr_cfg { @@ -152,7 +152,7 @@ struct skl_up_down_mixer_cfg { struct skl_algo_cfg { struct skl_base_cfg base_cfg; - char params[0]; + char params[]; } __packed; struct skl_base_outfmt_cfg { diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 2bfbf59277c4..26057f38a014 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -49,7 +49,7 @@ struct skl_astate_param { struct skl_astate_config { u32 count; - struct skl_astate_param astate_table[0]; + struct skl_astate_param astate_table[]; }; struct skl_fw_config { diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index e0f77172ce8f..144b85f57bd2 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -18,7 +18,7 @@ struct snd_usX2Y_AsyncSeq { struct snd_usX2Y_urbSeq { int submitted; int len; - struct urb *urb[0]; + struct urb *urb[]; }; #include "usx2yhwdeppcm.h" -- 2.25.0

2 2

[PATCH v2] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 20 Feb '20

20 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Fix init values to lower volume CFG4 is the LSB and CFG1 is MSB sound/soc/codecs/tas2562.c | 94 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 98 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..92e4bf34b35c 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -334,6 +353,22 @@ static int tas2562_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; + /* Set the Digital volume to -110dB */ + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, 0x43); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, 0x0d); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, 0x00); + if (ret) + return ret; + + tas2562->volume_lvl = 0; + return 0; } @@ -414,6 +449,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +506,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +607,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

1 0

Re: [alsa-devel] No sound since 5.4 on skl_n88l25_s4567
by Kuninori Morimoto 20 Feb '20

20 Feb '20

Hi ojab > > Please take a look, I want my sound back ._. > > Forgot to mention, it's reproducible on linux master git. Hmm... This patch 1) removes unneeded if check 2) adds missing error check. 1) shouldn't have effect to your PC. Thus, maybe your PC failed by 2). It had been worked by accident before ? I guess your kernel is telling some message like this ? ASoC: no dapm match for %s --> %s --> %s Thank you for your help !! Best regards --- Kuninori Morimoto

4 13

[PATCH] ASoC: samsung: Update dependencies for Arizona machine drivers
by Charles Keepax 20 Feb '20

20 Feb '20

Currently it is possible to get the following bad config: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] commit ea00d95200d0 ("ASoC: Use imply for SND_SOC_ALL_CODECS") commit d8dd3f92a6ba ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout") After these two patches the machine drivers still selects the SND_SOC_WM5110 symbol which doesn't take account of the dependency added on the MFD_WM5110 symbol, fix this by also adding a dependency on MFD_WM5110 itself. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1a0b163ca47b5..112911dc271ba 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -204,7 +204,7 @@ config SND_SOC_ARNDALE config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 -- 2.11.0

2 1

[PATCH] ASoC: soc-core: fix for_rtd_codec_dai_rollback() macro
by Pierre-Louis Bossart 20 Feb '20

20 Feb '20

The use of parentheses to protect the argument is fine for (i)++ but not for (--i). Fixes: 83f94a2e293d61 ("ASoC: soc-core: add snd_soc_close_delayed_work()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..8a2266676b2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1157,7 +1157,7 @@ struct snd_soc_pcm_runtime { ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); -- 2.20.1

3 2

[PATCH] ASoC: dpcm: remove confusing trace in dpcm_get_be()
by Jerome Brunet 20 Feb '20

20 Feb '20

Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error if this function does not find a BE for the provided widget. Remove the related dev_err() trace which is confusing since things might be working as expected. When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to find a BE for the provided widget. The necessary error trace is already done in this case. Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..aff27c8599ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1270,9 +1270,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } -- 2.24.1

4 3

[PATCH] ASoC: tas2562: Add support for ISENSE and VSENSE
by Dan Murphy 20 Feb '20

20 Feb '20

Add additional support for ISENSE and VSENSE feature for the TAS2562. This feature monitors the output to the loud speaker attempts to eliminate IR drop errors due to packaging. This feature is defined in Section 8.4.5 IV Sense of the data sheet. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 32 +++++++++++++++++++++++++++----- sound/soc/codecs/tas2562.h | 6 +++--- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5b4803a76719..b2682c2360b6 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -383,18 +383,34 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_POST_PMU: - dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_MUTE); + if (ret) + goto end; break; case SND_SOC_DAPM_PRE_PMD: - dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_SHUTDOWN); + if (ret) + goto end; break; default: - break; + dev_err(tas2562->dev, "Not supported evevt\n"); + return -EINVAL; } +end: + if (ret < 0) + return ret; + return 0; } @@ -416,7 +432,6 @@ static const struct snd_kcontrol_new tas2562_snd_controls[] = { static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), - SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), @@ -431,7 +446,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = { {"ASI1 Sel", "Left", "ASI1"}, {"ASI1 Sel", "Right", "ASI1"}, {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, - { "DAC", NULL, "DAC IN" }, + { "DAC", NULL, "ASI1 Sel" }, { "OUT", NULL, "DAC" }, {"ISENSE", "Switch", "IMON"}, {"VSENSE", "Switch", "VMON"}, @@ -472,6 +487,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2562_FORMATS, }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, .ops = &tas2562_speaker_dai_ops, }, }; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 62e659ab786d..6f55ebcf19ea 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -40,7 +40,7 @@ #define TAS2562_RESET BIT(0) -#define TAS2562_MODE_MASK 0x3 +#define TAS2562_MODE_MASK GENMASK(1,0) #define TAS2562_ACTIVE 0x0 #define TAS2562_MUTE 0x1 #define TAS2562_SHUTDOWN 0x2 @@ -73,8 +73,8 @@ #define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) #define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) -#define TAS2562_VSENSE_POWER_EN BIT(2) -#define TAS2562_ISENSE_POWER_EN BIT(3) +#define TAS2562_VSENSE_POWER_EN 2 +#define TAS2562_ISENSE_POWER_EN 3 #define TAS2562_TDM_CFG5_VSNS_EN BIT(6) #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) -- 2.25.0

2 3

[alsa-devel] Implicit feedback solution for Boss GT-1 and maybe other devices
by Mike Oliphant 20 Feb '20

20 Feb '20

I received a very helpful email response to my previous thread about trying to get the Boss GT-1 pedalboard working reliably, and as a result I now have a working solution. The issue seems to be that the GT-1 does not send any capture data until it gets some playback data first. This block in the "snd_usb_endpoint_start()" function in endpoint.c: "if (snd_usb_endpoint_implicit_feedback_sink(ep)) { ... }" causes the system to wait to send output until some capture urbs are received. As a result, both sides end up waiting on the other, and nothing happens. Simply commenting out that "if" block makes the GT-1 work perfectly with implicit feedback enabled. I plan to submit a patch for this after I do a bit of legwork to see if the fix works for other recent Boss devices (Katana, other GT devices, etc.). Perhaps making the relevant "if" block conditional on a flag, with a USB_ID switch statement to set the flag? Any feedback would be welcome. Thanks, Mike

2 1

Enable Headset Mic for Dell desktop
by Kailang 20 Feb '20

20 Feb '20

Hi Takashi, Attach patch was supported headset Mic. BR, Kailang

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read using a snd_rawmidi_read. (port is opened for input only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0