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

Support for TC-Helicon GoXLR Mini
by GitHub issues - edited 31 Dec '21

31 Dec '21

alsa-project/alsa-ucm-conf issue #133 was edited from lm41: Hi, many in the community had noticed that the Configuration for the GoXLR does not work for the GoXLR mini. https://www.tc-helicon.com/product.html?modelCode=P0DI7 Any ideas to get this to work as well? ref: #121 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/133 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Support for TC-Helicon GoXLR Mini
by GitHub issues - opened 31 Dec '21

31 Dec '21

alsa-project/alsa-ucm-conf issue #133 was opened from lm41: Hi, many in the community had noticed that the Configuration for the GoXLR does not work for the GoXLR mini. https://www.tc-helicon.com/product.html?modelCode=P0DI7 Any ideas to get this to work as well? ref: 121 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/133 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] ASoC: mediatek: mt8195: update control for RT5682 series
by Trevor Wu 29 Dec '21

29 Dec '21

Playback pop is observed and the root cause is the reference clock provided by MT8195 is diabled before RT5682 finishes the control flow. To ensure the reference clock supplied to RT5682 is disabled after RT5682 finishes all register controls. We replace BCLK with MCLK for RT5682 reference clock, and makes use of set_bias_level_post to handle MCLK which guarantees MCLK is off after all RT5682 register access. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- Changes since v1: 1. Adopt some suggestions from Tzung-bi. 2. mclk phandle is retrieved from afe component instead of dts. --- .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 56 ++++++++++++++- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 68 ++++++++++++++++--- 2 files changed, 110 insertions(+), 14 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index ce8dace6527c..8adbd46c4bc4 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -18,6 +18,7 @@ #include "../../codecs/rt1011.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" #define RT1011_CODEC_DAI "rt1011-aif" @@ -34,6 +35,7 @@ struct mt8195_mt6359_rt1011_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -84,8 +86,8 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_BCLK1, - rate * 64, rate * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; @@ -98,7 +100,7 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 128, + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, SND_SOC_CLOCK_OUT); } @@ -327,8 +329,14 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) struct mt8195_mt6359_rt1011_rt5682_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &priv->headset_jack; + struct snd_soc_component *cmpnt_afe = + snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); + struct mt8195_afe_private *afe_priv = afe->platform_priv; int ret; + priv->i2so1_mclk = afe_priv->clk[MT8195_CLK_TOP_APLL12_DIV2]; + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | @@ -562,6 +570,47 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1011_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + clk_disable_unprepare(priv->i2so1_mclk); + dev_dbg(card->dev, "Disable i2so1 mclk\n"); + break; + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable i2so1 mclk: %d\n", ret); + return ret; + } + dev_dbg(card->dev, "Enable i2so1 mclk\n"); + break; + default: + break; + } + + return 0; +} + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1037,6 +1086,7 @@ static struct snd_soc_card mt8195_mt6359_rt1011_rt5682_soc_card = { .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1011_rt5682_routes), .codec_conf = rt1011_amp_conf, .num_configs = ARRAY_SIZE(rt1011_amp_conf), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_mt6359_rt1011_rt5682_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index c15c58170e9d..20b351faeaff 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -19,6 +19,7 @@ #include "../../codecs/mt6359.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" #define RT1019_CODEC_DAI "HiFi" @@ -46,6 +47,7 @@ struct mt8195_mt6359_rt1019_rt5682_priv { struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; + struct clk *i2so1_mclk; }; static const struct snd_soc_dapm_widget @@ -92,8 +94,6 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); unsigned int rate = params_rate(params); - unsigned int mclk_fs_ratio = 128; - unsigned int mclk_fs = rate * mclk_fs_ratio; int bitwidth; int ret; @@ -109,25 +109,22 @@ static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, - RT5682_PLL1_S_BCLK1, - params_rate(params) * 64, - params_rate(params) * 512); + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL1, RT5682_PLL1_S_MCLK, + rate * 256, rate * 512); if (ret) { dev_err(card->dev, "failed to set pll\n"); return ret; } - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5682_SCLK_S_PLL1, - params_rate(params) * 512, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + rate * 512, SND_SOC_CLOCK_IN); if (ret) { dev_err(card->dev, "failed to set sysclk\n"); return ret; } - return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT); + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, + SND_SOC_CLOCK_OUT); } static const struct snd_soc_ops mt8195_rt5682_etdm_ops = { @@ -322,8 +319,14 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) struct mt8195_mt6359_rt1019_rt5682_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &priv->headset_jack; + struct snd_soc_component *cmpnt_afe = + snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); + struct mt8195_afe_private *afe_priv = afe->platform_priv; int ret; + priv->i2so1_mclk = afe_priv->clk[MT8195_CLK_TOP_APLL12_DIV2]; + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | @@ -560,6 +563,48 @@ static const struct snd_soc_ops mt8195_capture_ops = { .startup = mt8195_capture_startup, }; +static int mt8195_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + struct snd_soc_component *component = dapm->component; + struct mt8195_mt6359_rt1019_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + int ret; + + /* + * It's required to control mclk directly in the set_bias_level_post + * function for rt5682 and rt5682s codec, or the unexpected pop happens + * at the end of playback. + */ + if (!component || + (strcmp(component->name, RT5682_DEV0_NAME) && + strcmp(component->name, RT5682S_DEV0_NAME))) + return 0; + + + switch (level) { + case SND_SOC_BIAS_OFF: + if (!__clk_is_enabled(priv->i2so1_mclk)) + return 0; + + clk_disable_unprepare(priv->i2so1_mclk); + dev_dbg(card->dev, "Disable i2so1 mclk\n"); + break; + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(priv->i2so1_mclk); + if (ret) { + dev_err(card->dev, "Can't enable i2so1 mclk: %d\n", ret); + return ret; + } + dev_dbg(card->dev, "Enable i2so1 mclk\n"); + break; + default: + break; + } + + return 0; +} + enum { DAI_LINK_DL2_FE, DAI_LINK_DL3_FE, @@ -1199,6 +1244,7 @@ static struct snd_soc_card mt8195_mt6359_rt1019_rt5682_soc_card = { .num_dapm_widgets = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_widgets), .dapm_routes = mt8195_mt6359_rt1019_rt5682_routes, .num_dapm_routes = ARRAY_SIZE(mt8195_mt6359_rt1019_rt5682_routes), + .set_bias_level_post = mt8195_set_bias_level_post, }; static int mt8195_dailink_parse_of(struct snd_soc_card *card, struct device_node *np, -- 2.18.0

3 2

[PATCH] ASoC: samsung: idma: Check of ioremap return value
by Jiasheng Jiang 29 Dec '21

29 Dec '21

Because of the potential failure of the ioremap(), the buf->area could be NULL. Therefore, we need to check it and return -ENOMEM in order to transfer the error. Fixes: f09aecd50f39 ("ASoC: SAMSUNG: Add I2S0 internal dma driver") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/samsung/idma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 66bcc2f97544..c3f1b054e238 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -360,6 +360,8 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) buf->addr = idma.lp_tx_addr; buf->bytes = idma_hardware.buffer_bytes_max; buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes); + if (!buf->area) + return -ENOMEM; return 0; } -- 2.25.1

3 2

[PATCH] ASoC: mediatek: use of_device_get_match_data()
by Tzung-Bi Shih 29 Dec '21

29 Dec '21

Uses of_device_get_match_data() helper to clean some boilerplate code. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 7 ++----- sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 7 ++----- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 7 ++----- 3 files changed, 6 insertions(+), 15 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index ca893feab7eb..718505c75418 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -685,7 +685,6 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct mt8183_da7219_max98357_priv *priv; struct pinctrl *pinctrl; - const struct of_device_id *match; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -695,11 +694,9 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - match = of_match_device(pdev->dev.driver->of_match_table, &pdev->dev); - if (!match || !match->data) + card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); + if (!card) return -EINVAL; - - card = (struct snd_soc_card *)match->data; card->dev = &pdev->dev; hdmi_codec = of_parse_phandle(pdev->dev.of_node, diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 19f8aead775d..b0ec5ebd4f2d 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -637,7 +637,6 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) struct device_node *platform_node, *ec_codec, *hdmi_codec; struct snd_soc_dai_link *dai_link; struct mt8183_mt6358_ts3a227_max98357_priv *priv; - const struct of_device_id *match; int ret, i; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -647,11 +646,9 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - match = of_match_device(pdev->dev.driver->of_match_table, &pdev->dev); - if (!match || !match->data) + card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); + if (!card) return -EINVAL; - - card = (struct snd_soc_card *)match->data; card->dev = &pdev->dev; ec_codec = of_parse_phandle(pdev->dev.of_node, "mediatek,ec-codec", 0); diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 2552f30d8fe4..f7daad1bfe1e 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1106,7 +1106,6 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) struct device_node *platform_node, *hdmi_codec; int ret, i; struct snd_soc_dai_link *dai_link; - const struct of_device_id *match; struct mt8192_mt6359_priv *priv; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -1116,11 +1115,9 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) return -EINVAL; } - match = of_match_device(pdev->dev.driver->of_match_table, &pdev->dev); - if (!match || !match->data) + card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); + if (!card) return -EINVAL; - - card = (struct snd_soc_card *)match->data; card->dev = &pdev->dev; hdmi_codec = of_parse_phandle(pdev->dev.of_node, -- 2.34.1.448.ga2b2bfdf31-goog

2 1

[PATCH] ASoC: rt5682: Register wclk with its parent_hws instead of parent_data
by derek.fang＠realtek.com 29 Dec '21

29 Dec '21

From: Derek Fang <derek.fang(a)realtek.com> The mclk might not be registered as a fixed clk name "mclk" on some platforms. In those platforms, if the mclk needed to be controlled by codec driver and acquired by a fixed name, it would be a problem. This patch to fix the issue that wclk becomes an orphan due to the fixed mclk's name. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 5224123d0d3b..4709c0628804 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2858,7 +2858,6 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { struct clk_init_data init = { }; - struct clk_parent_data parent_data; const struct clk_hw *parent; dai_clk_hw = &rt5682->dai_clks_hw[i]; @@ -2867,10 +2866,8 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) case RT5682_DAI_WCLK_IDX: /* Make MCLK the parent of WCLK */ if (rt5682->mclk) { - parent_data = (struct clk_parent_data){ - .fw_name = "mclk", - }; - init.parent_data = &parent_data; + parent = __clk_get_hw(rt5682->mclk); + init.parent_hws = &parent; init.num_parents = 1; } break; -- 2.17.1

2 1

[PATCH RESEND 1/2] ASoC: cs4265: Fix part number ID error message
by Fabio Estevam 29 Dec '21

29 Dec '21

From: Fabio Estevam <festevam(a)denx.de> The Chip ID - Register 01h contains the following description as per the CS4265 datasheet: "Bits 7 through 4 are the part number ID, which is 1101b (0Dh)" The current error message is incorrect as it prints CS4265_CHIP_ID, which is the register number, instead of printing the expected part number ID value. To make it clearer, also do a shift by 4, so that the error message would become: [ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- Sorry, I am resending because I was not subscribed in the alsa-list from the other e-mail account. sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index cffd6111afac..b89002189a2b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -611,8 +611,8 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, if (devid != CS4265_CHIP_ID_VAL) { ret = -ENODEV; dev_err(&i2c_client->dev, - "CS4265 Device ID (%X). Expected %X\n", - devid, CS4265_CHIP_ID); + "CS4265 Part Number ID: 0x%x Expected: 0x%x\n", + devid >> 4, CS4265_CHIP_ID_VAL >> 4); return ret; } dev_info(&i2c_client->dev, -- 2.25.1

3 4

[patch] speaker-test/samples/sample_map.csv is not accurate, which does not affect operation of speaker-test
by GitHub issues - edited 28 Dec '21

28 Dec '21

alsa-project/alsa-utils issue #133 was edited from ZjYwMj: With alsa-utils 1.2.6: The [commit](https://github.com/alsa-project/alsa-utils/commit/fb2f99f1f99c430d1… in question. ``` --- a/usr/share/alsa/speaker-test/sample_map.csv 2021-12-06 18:44:25.000000000 +0000 +++ b/usr/share/alsa/speaker-test/sample_map.csv 2021-12-28 19:48:02.390305957 +0000 @@ -1,2 +1,2 @@ -0, "Front Left", "/usr/share/alsa/samples/Front_Left.wav" -1, "Front Right", "/usr/share/alsa/samples/Front_Right.wav" +0, "Front Left", "/usr/share/sounds/alsa/Front_Left.wav" +1, "Front Right", "/usr/share/sounds/alsa/Front_Right.wav" ``` The directory ``/usr/share/alsa/samples/`` does not exist. It sounds as if ``speaker-test -t wav`` plays the files mentioned in the diff. This inaccuracy does not affect the operation of ``speaker-test``. Do the paths to the wav files hard coded within ``speaker-test``? Does only their ``basename`` counts? Still, I find it a minor bug. Why is ``/usr/share/alsa/speaker-test/sample_map.csv`` distributed anyway? Am I misunderstanding [Install sound samples to correct places](https://github.com/alsa-project/alsa-utils/commit/fb2f99f1f99c430d1… Is it for informational purposes? Was it used in the past, and no longer serves its purpose? Is it planed to be used in the future? Is it something else? [speaker-test.c:745](https://github.com/alsa-project/alsa-utils/blob/00be486… and [746](https://github.com/alsa-project/alsa-utils/blob/00be486131129a6d209de6… has the ``basename``s hardcoded. And there is also the default value for ``speaker-test --wavdir`` of ``/usr/share/sounds/alsa``. Issue URL : https://github.com/alsa-project/alsa-utils/issues/133 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[patch] speaker-test/samples/sample_map.csv is not accurate, which does not affect operation of speaker-test
by GitHub issues - edited 28 Dec '21

28 Dec '21

alsa-project/alsa-utils issue #133 was edited from ZjYwMj: With alsa-utils 1.2.6: ``` --- a/usr/share/alsa/speaker-test/sample_map.csv 2021-12-06 18:44:25.000000000 +0000 +++ b/usr/share/alsa/speaker-test/sample_map.csv 2021-12-28 19:48:02.390305957 +0000 @@ -1,2 +1,2 @@ -0, "Front Left", "/usr/share/alsa/samples/Front_Left.wav" -1, "Front Right", "/usr/share/alsa/samples/Front_Right.wav" +0, "Front Left", "/usr/share/sounds/alsa/Front_Left.wav" +1, "Front Right", "/usr/share/sounds/alsa/Front_Right.wav" ``` The directory ``/usr/share/alsa/samples/`` does not exist. It sounds as if ``speaker-test -t wav`` plays the files mentioned in the diff. This inaccuracy does not affect the operation of ``speaker-test``. Do the paths to the wav files hard coded within ``speaker-test``? Does only their ``basename`` counts? Still, I find it a minor bug. Why is ``/usr/share/alsa/speaker-test/sample_map.csv`` distributed anyway? Am I misunderstanding [Install sound samples to correct places](https://github.com/alsa-project/alsa-utils/commit/fb2f99f1f99c430d1… Is it for informational purposes? Was it used in the past, and no longer serves its purpose? Is it planed to be used in the future? Is it something else? [speaker-test.c:745](https://github.com/alsa-project/alsa-utils/blob/00be486… and [746](https://github.com/alsa-project/alsa-utils/blob/00be486131129a6d209de6… has the ``basename``s hardcoded. And there is also the default value for ``speaker-test --wavdir`` of ``/usr/share/sounds/alsa``. Issue URL : https://github.com/alsa-project/alsa-utils/issues/133 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[patch] alsa-utils/alsactl/init/00main:1, which is a comment and contains a typo
by GitHub issues - edited 28 Dec '21

28 Dec '21

alsa-project/alsa-utils issue #132 was edited from ZjYwMj: With alsa-utils 1.2.6: This should probably get applied to [00main:1](https://github.com/alsa-project/alsa-utils/blob/90f59671784a7e47b…. ``` --- a/usr/share/alsa/init/00main 2021-12-06 18:44:25.000000000 +0000 +++ b/usr/share/alsa/init/00main 2021-12-28 17:04:36.835174838 +0000 @@ -1,4 +1,4 @@ -# This is toplevel configuration for for 'alsactl init'. +# This is the toplevel configuration file for 'alsactl init'. # See 'man alsactl_init' for syntax. # set root device directory in sysfs for soundcard for ATTR{} command ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/132 Repository URL: https://github.com/alsa-project/alsa-utils

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