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

Re: [alsa-devel] [PATCH] ALSA: hda/realtek: Add quirks for several Clevo notebook barebones
by Takashi Iwai 19 Jun '19

19 Jun '19

On Sat, 15 Jun 2019 05:42:07 +0200, Richard Sailer wrote: > > This adds 4 SND_PCI_QUIRK(...) lines for several barebone models of the ODM > Clevo. The model names are written in regex syntax to describe/match all clevo > models that are similar enough and use the same PCI SSID that this fixup works > for them. > > Additionally the lines regarding SSID 0x96e1 and 0x97e1 didn't fix audio for the > all our Clevo notebooks using these SSIDs (models Clevo P960* and P970*) since > ALC1220_FIXP_CLEVO_PB51ED_PINS swapped pins that are not necesarry to be > swapped. This patch initiates ALC1220_FIXUP_CLEVO_P950 instead for these model > and fixes the audio. Your sign-off is missing. Could you resubmit with it? > This is strange since the System76 Oryx Pro also seems to be a P960 and > P970 barebone, so the original fixup should also not have worked for > them. @Jeremy: What's your take on this? ... and it's better if this mystery is clarified, too. Regarding the code change: > --- > sound/pci/hda/patch_realtek.c | 7 ++++--- > 1 file changed, 4 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 1afb268f3da0..61dcd293afb1 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -2461,9 +2461,10 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { > SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), > SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), > SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), > - SND_PCI_QUIRK(0x1558, 0x96e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), > - SND_PCI_QUIRK(0x1558, 0x97e1, "System76 Oryx Pro (oryp5)", ALC1220_FIXUP_CLEVO_PB51ED_PINS), > - SND_PCI_QUIRK(0x1558, 0x65d1, "Tuxedo Book XC1509", ALC1220_FIXUP_CLEVO_PB51ED_PINS), > + SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), > + SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), > + SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), > + SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), We don't have to list up necessarily the detailed model names here. Though, it's OK in this form as it's done concisely. thanks, Takashi

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[alsa-devel] [bug report] ASoC: meson: axg-card: fix null pointer dereference in clean up
by Dan Carpenter 19 Jun '19

19 Jun '19

Hello Jerome Brunet, The patch 960f428ca0a0: "ASoC: meson: axg-card: fix null pointer dereference in clean up" from Jun 10, 2019, leads to the following static checker warning: sound/soc/meson/axg-card.c:121 axg_card_clean_references() warn: address of 'link->codecs[j]' is non-NULL sound/soc/meson/axg-card.c 109 static void axg_card_clean_references(struct axg_card *priv) 110 { 111 struct snd_soc_card *card = &priv->card; 112 struct snd_soc_dai_link *link; 113 struct snd_soc_dai_link_component *codec; 114 int i, j; 115 116 if (card->dai_link) { 117 for_each_card_prelinks(card, i, link) { 118 if (link->cpus) 119 of_node_put(link->cpus->of_node); 120 for_each_link_codecs(link, j, codec) 121 if (codec) ^^^^^ "codec" is the list iterator so it can't possibly be NULL. 122 of_node_put(codec->of_node); 123 } 124 } 125 126 if (card->aux_dev) { 127 for (i = 0; i < card->num_aux_devs; i++) 128 of_node_put(card->aux_dev[i].codec_of_node); 129 } 130 131 kfree(card->dai_link); 132 kfree(priv->link_data); 133 } regards, dan carpenter

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[alsa-devel] [PATCH v2] ASoC: max98357a: avoid speaker pop when playback startup
by mac.chiang＠intel.com 19 Jun '19

19 Jun '19

From: Mac Chiang <mac.chiang(a)intel.com> Loud speaker pop happens during playback even when in slience playback. Specify Max98357a amp delay times to make sure clocks are always earlier than sdmode on. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v2: - added device_property_read_u32() func --- --- .../devicetree/bindings/sound/max98357a.txt | 4 ++ sound/soc/codecs/max98357a.c | 68 +++++++++++++++++++--- 2 files changed, 63 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 28645a2..4bce14c 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -9,6 +9,10 @@ Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. If this option is not specified then driver does not manage the pin state (e.g. chip is always on). +- sdmode-delay : specify delay time for SD_MODE pin. + If this option is specified, which means it's required i2s clocks + ready before SD_MODE is unmuted in order to avoid the speaker pop noise. + It's observed that 5ms is sufficient. Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e..91ca0aa 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,24 +27,42 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + unsigned int sdmode_delay; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *max98357a = + container_of(work, struct max98357a_priv, + enable_sdmode_work.work); + + gpiod_set_value(max98357a->sdmode, 1); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct max98357a_priv *max98357a = snd_soc_dai_get_drvdata(dai); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + queue_delayed_work(system_power_efficient_wq, + &max98357a->enable_sdmode_work, + msecs_to_jiffies(max98357a->sdmode_delay)); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -61,19 +79,32 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + max98357a->sdmode = devm_gpiod_get_optional(component->dev, + "sdmode", GPIOD_OUT_LOW); - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); - snd_soc_component_set_drvdata(component, sdmode); + snd_soc_component_set_drvdata(component, max98357a); + + INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, + max98357a_enable_sdmode_work); return 0; } +static void max98357a_component_remove(struct snd_soc_component *component) +{ + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); +} + static const struct snd_soc_component_driver max98357a_component_driver = { .probe = max98357a_component_probe, + .remove = max98357a_component_remove, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, @@ -112,6 +143,25 @@ static struct snd_soc_dai_driver max98357a_dai_driver = { static int max98357a_platform_probe(struct platform_device *pdev) { + struct max98357a_priv *max98357a; + int ret; + + max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL); + + if (!max98357a) + return -ENOMEM; + + ret = device_property_read_u32(&pdev->dev, "sdmode-delay", + &max98357a->sdmode_delay); + + if (ret) { + max98357a->sdmode_delay = 0; + dev_dbg(&pdev->dev, + "no optional property 'sdmode-delay' found, default: no delay\n"); + } + + dev_set_drvdata(&pdev->dev, max98357a); + return devm_snd_soc_register_component(&pdev->dev, &max98357a_component_driver, &max98357a_dai_driver, 1); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: max98357a: avoid speaker pop when playback startup
by mac.chiang＠intel.com 19 Jun '19

19 Jun '19

From: Mac Chiang <mac.chiang(a)intel.com> Loud speaker pop happens during playback even when in slience playback. Specify Max98357a amp delay times to make sure clocks are always earlier than sdmode on. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- .../devicetree/bindings/sound/max98357a.txt | 4 ++ sound/soc/codecs/max98357a.c | 50 ++++++++++++++++++---- 2 files changed, 45 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98357a.txt b/Documentation/devicetree/bindings/sound/max98357a.txt index 28645a2..4bce14c 100644 --- a/Documentation/devicetree/bindings/sound/max98357a.txt +++ b/Documentation/devicetree/bindings/sound/max98357a.txt @@ -9,6 +9,10 @@ Optional properties: - sdmode-gpios : GPIO specifier for the chip's SD_MODE pin. If this option is not specified then driver does not manage the pin state (e.g. chip is always on). +- sdmode-delay : specify delay time for SD_MODE pin. + If this option is specified, which means it's required i2s clocks + ready before SD_MODE is unmuted in order to avoid the speaker pop noise. + It's observed that 5ms is sufficient. Example: diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d037a3e..2141fb4 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,24 +27,43 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + unsigned int sdmode_delay; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *max98357a = + container_of(work, struct max98357a_priv, + enable_sdmode_work.work); + + gpiod_set_value(max98357a->sdmode, 1); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct snd_soc_component *component = dai->component; + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); - if (!sdmode) + if (!max98357a->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + queue_delayed_work(system_power_efficient_wq, + &max98357a->enable_sdmode_work, + msecs_to_jiffies(max98357a->sdmode_delay)); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); + gpiod_set_value(max98357a->sdmode, 0); break; } @@ -61,19 +80,32 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + max98357a->sdmode = devm_gpiod_get_optional(component->dev, + "sdmode", GPIOD_OUT_LOW); - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + if (IS_ERR(max98357a->sdmode)) + return PTR_ERR(max98357a->sdmode); - snd_soc_component_set_drvdata(component, sdmode); + snd_soc_component_set_drvdata(component, max98357a); + + INIT_DELAYED_WORK(&max98357a->enable_sdmode_work, + max98357a_enable_sdmode_work); return 0; } +static void max98357a_component_remove(struct snd_soc_component *component) +{ + struct max98357a_priv *max98357a = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&max98357a->enable_sdmode_work); +} + static const struct snd_soc_component_driver max98357a_component_driver = { .probe = max98357a_component_probe, + .remove = max98357a_component_remove, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, -- 2.7.4

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[alsa-devel] [PATCH] ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY
by Kuninori Morimoto 19 Jun '19

19 Jun '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dobj is used only when SND_SOC_TOPOLOGY was selected. Let's enable it under SND_SOC_TOPOLOGY. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1094cd8..315a6e1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -853,7 +853,9 @@ struct snd_soc_dai_link { unsigned int ignore:1; struct list_head list; /* DAI link list of the soc card */ +#ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ +#endif }; #define for_each_link_codecs(link, i, codec) \ for ((i) = 0; \ @@ -1167,7 +1169,9 @@ struct soc_mixer_control { unsigned int sign_bit; unsigned int invert:1; unsigned int autodisable:1; +#ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; +#endif }; struct soc_bytes { @@ -1178,8 +1182,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; +#ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; - +#endif /* used for TLV byte control */ int (*get)(struct snd_kcontrol *kcontrol, unsigned int __user *bytes, unsigned int size); @@ -1203,7 +1208,9 @@ struct soc_enum { const char * const *texts; const unsigned int *values; unsigned int autodisable:1; +#ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; +#endif }; /* device driver data */ -- 2.7.4

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[alsa-devel] Serious bug calling ALSA lib functions from .so vs. .o file
by scott andrew franco 18 Jun '19

18 Jun '19

Hi, I am having a strange issue calling snd_pcm_* alsa-lib functions from a .so file. The example is a pretty run of the mill .wav file output example in C from the internet, which I have placed in playwav.c. I put the routine that does the pcm play, alsaplaywave(), into a routine and have a duplicate in another external file, test2.c. I named the one in playwav.c alsaplaywav1(), and the one in test2.c alsaplaywav(). The compile is done as follows: gcc -g3 -Iinclude -fPIC -c test2.c -o test2.o gcc -shared test2.o -o test2.so # # Compiling from .so causes bug, compiling from .o does not # gcc -g3 -Iinclude linux/playwav.c ./test2.so -lasound -o playwav #gcc -g3 -Iinclude linux/playwav.c ./test2.o -lasound -o playwav If the test2.so file is used, the playback has its parameters such as rate messed up, and the result is it plays as fast as possible (series of chirps). If the above line is just changed to test2.o, the playback is fine. Further, if the copy of alsaplaywav() local to the playwav.c file is executed, the playback is correct. It is only when playing from the routine in the .so file that it messes up. To find out what the key point of failure is, I put all of the preamble for the routine up until it does a snd_pcm_hw_params_set_rate_near() at the calling point (in playwav.c) and commented that out in the routine. The idea was to see exactly which alsa call being done inside the .so caused the malfunction. The rate set appears to be the key. Do that outside the .so, it works, do it inside, it does not. I note that snd_pcm_hw_params_set_rate_near() goes through a few macros in alsa-lib. I think I could debug this better if I can get alsa symbols into gdb. As in the other message, I cannot figure out where the libasound.so gets constructed in the build. I see libasound.a, but no libasound.so. Apologies for not cutting the example down further, but the failure mode is quite complex and that is difficult. Find playwav.c and test2.c attached, and the compile instructions I used above or in cplaywav. The files are fairly small. Thanks for any help. Scott Franco

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[alsa-devel] [PATCH v7 0/4] Fix jack detection for Chromebook Pixel
by Fletcher Woodruff 18 Jun '19

18 Jun '19

Headphone/mic jack detection doesn't work on the Chromebook Pixel 2015. This patch changes the irq implementation to support polarity flipping and fixes the configuration code so that correct GPIO pins are read from ACPI. With this series, plugging and unplugging the headphone jack switches between headphones and speakers automatically, and headset microphones are also detected. v7: - Rebase onto for-next branch of broonie/sound.git v6: - Move refactoring into its own patch - Reorder patches so that DT property names patch is first - Clarify commit message for patch which implements irq handler - Remove unused work struct - Make IRQ function return IRQ_HANDLED only if IRQs actually fire v5: - Fix void* parameter to devm_request_threaded_irq v4: - Fix incorrect void* cast in rt5677_irq() v3: - Update commit message for patch 1/3 to clarify why we implement our own irq_chip. v2: - Split IRQ change into two patches: adding and fixing potential race - Change config reading code to try both DT and ACPI style names Ben Zhang (2): ASoC: rt5677: clear interrupts by polarity flip ASoC: rt5677: handle concurrent interrupts Fletcher Woodruff (2): ASoC: rt5677: fall back to DT prop names on error ASoC: rt5677: move jack-detect init to i2c probe sound/soc/codecs/rt5677.c | 319 ++++++++++++++++++++++++++------------ sound/soc/codecs/rt5677.h | 13 +- 2 files changed, 236 insertions(+), 96 deletions(-) -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH] ASoC: Intel: sof_rt5682: use GFP_KERNEL instead of GFP_ATOMIC
by Tzung-Bi Shih 18 Jun '19

18 Jun '19

Change the memory allocation flag from GFP_ATOMIC to GFP_KERNEL because probe of platform device is unlikely a place where cannot sleep. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- It might be a typo introduced from copied-and-pasted. sound/soc/intel/boards/sof_rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index e2e5f97d9920..02138aa8a850 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -498,7 +498,7 @@ static int sof_audio_probe(struct platform_device *pdev) int dmic_num, hdmi_num; int ret, ssp_amp, ssp_codec; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH v2 1/1] ASoC: rsnd: fixup mod ID calculation in rsnd_ctu_probe_
by Jiada Wang 18 Jun '19

18 Jun '19

From: Nilkanth Ahirrao <anilkanth(a)jp.adit-jv.com> commit c16015f36cc1 ("ASoC: rsnd: add .get_id/.get_id_sub") introduces rsnd_ctu_id which calcualates and gives the main Device id of the CTU by dividing the id by 4. rsnd_mod_id uses this interface to get the CTU main Device id. But this commit forgets to revert the main Device id calcution previously done in rsnd_ctu_probe_ which also divides the id by 4. This path corrects the same to get the correct main Device id. The issue is observered when rsnd_ctu_probe_ is done for CTU1 Fixes: c16015f36cc1 ("ASoC: rsnd: add .get_id/.get_id_sub") Signed-off-by: Nilkanth Ahirrao <anilkanth(a)jp.adit-jv.com> Signed-off-by: Suresh Udipi <sudipi(a)jp.adit-jv.com> Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- v2: fixed commit id v1: initial version --- sound/soc/sh/rcar/ctu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index 8cb06dab234e..7647b3d4c0ba 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -108,7 +108,7 @@ static int rsnd_ctu_probe_(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - return rsnd_cmd_attach(io, rsnd_mod_id(mod) / 4); + return rsnd_cmd_attach(io, rsnd_mod_id(mod)); } static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, -- 2.19.2

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[alsa-devel] Issues with tip alsa-lib (stable, 1.1.9)
by scott andrew franco 18 Jun '19

18 Jun '19

A roundup of my current issues with the alsa-lib repository: 1. No apparent .so (dynamic link) file output. The configure and make instructions were followed, and no apparent .so resulted: samiam@samiam-home-pc:~/projects/alsa/alsa-lib-1.1.9$ find . -name *.so samiam@samiam-home-pc:~/projects/alsa/alsa-lib-1.1.9$ find . -name *.a ./src/timer/.libs/libtimer.a ./src/rawmidi/.libs/librawmidi.a ./src/ucm/.libs/libucm.a ./src/topology/.libs/libtopology.a ./src/.libs/libasound.a ./src/hwdep/.libs/libhwdep.a ./src/seq/.libs/libseq.a ./src/mixer/.libs/libmixer.a ./src/control/.libs/libcontrol.a ./src/pcm/.libs/libpcm.a samiam@samiam-home-pc:~/projects/alsa/alsa-lib-1.1.9$ This has made it difficult to debug issues with the library. Using the configure flags does not seem to have an effect on this, IE: ./configure --enable-shared=yes --enable-static=no Linux distributions have libasound.so files in them, so clearly it was possible at some time in the past. 2. Build instructions are conflicting. ./configure make Does work on this version. This is the standard GNU build method, and I have seen these build instructions around the internet for building alsa. The INSTALL file recommends: ./gitcompile Which also (appears) to work, but no reason appears for why this special file is needed or what it does that is different. I can't get a .so (dynamic library from the build, so I can't check the function of the resulting build. Is there a reason the instructions from ./gitcompile cannot be merged back into the make? 3. make install brings the system down. Executing: make install Brought my ubuntu 18.04 sound system down. I had to reinstall Ubuntu to get it to work again. In summary, ALSA is not a development project anymore. It is the "official sound system of Linux" and is part of the kernel. There is a lot of activity on the ALSA list. Cannot some effort be put into professionalizing the repository? Thank you for your attention, Scott A. Franco San Jose, CA

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