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

[alsa-devel] [PATCH 00/10] soundwire: bus: fix race conditions, add suspend-resume
by Pierre-Louis Bossart 25 Feb '20

25 Feb '20

The existing mainline code for SoundWire does not handle critical race conditions, and does not have any support for pm_runtime suspend or clock-stop modes needed for e.g. jack detection or external VAD. As suggested by Vinod, these patches for the bus are shared first - with the risk that they are separated from their actual use in Intel drivers, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 90+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 These patches are not Intel-specific and are likely required for e.g. Qualcomm-based implementations. All the patches in this series were generated during the joint Intel-Realtek validation effort on Intel reference designs and form-factor devices. The support for the initialization_complete signaling is already available in the Realtek codecs drivers merged in the ASoC tree (rt700, rt711, rt1308, rt715) Pierre-Louis Bossart (8): soundwire: bus: fix race condition with probe_complete signaling soundwire: bus: fix race condition with enumeration_complete signaling soundwire: bus: fix race condition with initialization_complete signaling soundwire: bus: add PM/no-PM versions of read/write functions soundwire: bus: write Slave Device Number without runtime_pm soundwire: bus: add helper to clear Slave status to UNATTACHED soundwire: bus: disable pm_runtime in sdw_slave_delete soundwire: bus: don't treat CMD_IGNORED as error on ClockStop Rander Wang (2): soundwire: bus: fix io error when processing alert event soundwire: bus: add clock stop helpers drivers/soundwire/bus.c | 509 ++++++++++++++++++++++++++++++++-- drivers/soundwire/bus.h | 9 + drivers/soundwire/bus_type.c | 5 + drivers/soundwire/slave.c | 4 + include/linux/soundwire/sdw.h | 24 ++ 5 files changed, 526 insertions(+), 25 deletions(-) -- 2.20.1

2 13

[alsa-devel] [PATCH] ASoC: fsl: Add generic CPU DAI driver
by Daniel Baluta (OSS) 25 Feb '20

25 Feb '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On i.MX8 platforms that have a DSP the DAI handling is taken care of by two entities: * Application Processor (AP), which runs Linux * DSP, which runs a firmware (typically Sound Open Firmware) The DSP has access to DAI IP registers, but it cannot easily handle resources like: * clocks * power domain management * pinctrl For this reason we introduce a generic FSL DAI driver which will take care of the resources above. --- sound/soc/fsl/Kconfig | 8 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dai.c | 276 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 286 insertions(+) create mode 100644 sound/soc/fsl/fsl_dai.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 65e8cd4be930..bea8ab2c24f9 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -75,6 +75,14 @@ config SND_SOC_FSL_ESAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_DAI + tristate "Generic FSL DAI support for Sound Open Firmware" + help + Say Y if you want to enable generic FSL DAI support to be used + with Sound Open Firmware. This module takes care of enabling + clocks, power domain, pinctrl for FSL DAIs. The rest of DAI + control is taken care of by SOF firmware. + config SND_SOC_FSL_MICFIL tristate "Pulse Density Modulation Microphone Interface (MICFIL) module support" select REGMAP_MMIO diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8cde88c72d93..e4ed253b6657 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -20,6 +20,7 @@ snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o +snd-soc-fsl-dai-objs := fsl_dai.o snd-soc-fsl-micfil-objs := fsl_micfil.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o @@ -32,6 +33,7 @@ obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o +obj-$(CONFIG_SND_SOC_FSL_DAI) += snd-soc-fsl-dai.o obj-$(CONFIG_SND_SOC_FSL_MICFIL) += snd-soc-fsl-micfil.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_FSL_MQS) += snd-soc-fsl-mqs.o diff --git a/sound/soc/fsl/fsl_dai.c b/sound/soc/fsl/fsl_dai.c new file mode 100644 index 000000000000..f70889939801 --- /dev/null +++ b/sound/soc/fsl/fsl_dai.c @@ -0,0 +1,276 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic DAI driver for DSP +// +// Copyright 2019 NXP +// Author: Daniel Baluta <daniel.baluta(a)nxp.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/pm_domain.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +enum fsl_dai_type { + FSL_DAI_TYPE_NONE, + FSL_DAI_TYPE_SAI, + FSL_DAI_TYPE_ESAI, +}; + +#define FSL_DAI_ESAI_CLK_NUM 4 +static const char *esai_clks[FSL_DAI_ESAI_CLK_NUM] = { + "core", + "extal", + "fsys", + "spba", +}; + +#define FSL_DAI_SAI_CLK_NUM 5 +static const char *sai_clks[FSL_DAI_SAI_CLK_NUM] = { + "bus", + "mclk0", + "mclk1", + "mclk2", + "mclk3", +}; + +struct fsl_dai { + struct platform_device *pdev; + + /* DAI clocks */ + struct clk **clks; + const char **clk_names; + int num_clks; + + /* Power Domain handling */ + int num_domains; + struct device **pd_dev; + struct device_link **link; + + /* DAIS */ + struct snd_soc_dai_driver *dai_drv; + int num_drv; +}; + +static struct snd_soc_dai_driver fsl_esai_dai = { + .name = "esai0", +}; + +static struct snd_soc_dai_driver fsl_sai_dai = { + .name = "sai1", +}; + +static const struct snd_soc_component_driver fsl_dai_component = { + .name = "fsl-dai", +}; + +static int fsl_dai_init_clocks(struct fsl_dai *dai_priv) +{ + struct device *dev = &dai_priv->pdev->dev; + int i; + + dai_priv->clks = devm_kcalloc(dev, dai_priv->num_clks, + sizeof(*dai_priv->clks), GFP_KERNEL); + if (!dai_priv->clks) + return -ENOMEM; + + for (i = 0; i < dai_priv->num_clks; i++) { + dai_priv->clks[i] = devm_clk_get(dev, dai_priv->clk_names[i]); + if (IS_ERR(dai_priv->clks[i])) { + dev_dbg(dev, "Failed to get clk %s\n", + dai_priv->clk_names[i]); + dai_priv->clks[i] = NULL; + } + } + + return 0; +} + +int fsl_get_dai_type(struct fsl_dai *dai_priv) +{ + struct device_node *np = dai_priv->pdev->dev.of_node; + + if (of_device_is_compatible(np, "fsl,esai-dai")) + return FSL_DAI_TYPE_ESAI; + + if (of_device_is_compatible(np, "fsl,sai-dai")) + return FSL_DAI_TYPE_SAI; + + return FSL_DAI_TYPE_NONE; +} + +static int fsl_dai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_dai *priv; + int dai_type; + int ret; + int i; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->pdev = pdev; + + dev_set_drvdata(&pdev->dev, priv); + + dai_type = fsl_get_dai_type(priv); + switch (dai_type) { + case FSL_DAI_TYPE_ESAI: + priv->clk_names = esai_clks; + priv->num_clks = FSL_DAI_ESAI_CLK_NUM; + priv->dai_drv = &fsl_esai_dai; + priv->num_drv = 1; + break; + case FSL_DAI_TYPE_SAI: + priv->clk_names = sai_clks; + priv->num_clks = FSL_DAI_SAI_CLK_NUM; + priv->dai_drv = &fsl_sai_dai; + priv->num_drv = 1; + break; + default: + dev_err(&pdev->dev, "Invalid DAI type %d\n", dai_type); + return -EINVAL; + } + + ret = fsl_dai_init_clocks(priv); + if (ret < 0) { + dev_err(&pdev->dev, "Error at init clocks\n"); + return ret; + } + + priv->num_domains = of_count_phandle_with_args(np, "power-domains", + "#power-domain-cells"); + if (priv->num_domains < 0) { + dev_err(&pdev->dev, "no power-domains property in %pOF\n", np); + return priv->num_domains; + } + + priv->pd_dev = devm_kmalloc_array(&pdev->dev, priv->num_domains, + sizeof(*priv->pd_dev), GFP_KERNEL); + if (!priv->pd_dev) + return -ENOMEM; + + priv->link = devm_kmalloc_array(&pdev->dev, priv->num_domains, + sizeof(*priv->link), GFP_KERNEL); + if (!priv->link) + return -ENOMEM; + + for (i = 0; i < priv->num_domains; i++) { + priv->pd_dev[i] = dev_pm_domain_attach_by_id(&pdev->dev, i); + if (IS_ERR(priv->pd_dev[i])) { + ret = PTR_ERR(priv->pd_dev[i]); + goto unroll_pm; + } + + priv->link[i] = device_link_add(&pdev->dev, priv->pd_dev[i], + DL_FLAG_STATELESS | + DL_FLAG_PM_RUNTIME | + DL_FLAG_RPM_ACTIVE); + if (!priv->link[i]) { + ret = -EINVAL; + dev_pm_domain_detach(priv->pd_dev[i], false); + goto unroll_pm; + } + } + + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_dai_component, + priv->dai_drv, priv->num_drv); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register DAI ret = %d\n", ret); + return ret; + } + return 0; + +unroll_pm: + while (--i >= 0) { + device_link_del(priv->link[i]); + dev_pm_domain_detach(priv->pd_dev[i], false); + } + return ret; +} + +static int fsl_dai_remove(struct platform_device *pdev) +{ + struct fsl_dai *priv = platform_get_drvdata(pdev); + int i; + + pm_runtime_disable(&priv->pdev->dev); + + for (i = 0; i < priv->num_domains; i++) { + device_link_del(priv->link[i]); + dev_pm_domain_detach(priv->pd_dev[i], false); + } + + return 0; +} + +static const struct of_device_id fsl_dai_dt_ids[] = { + { .compatible = "fsl,esai-dai", }, + { .compatible = "fsl,sai-dai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_dai_dt_ids); + +#ifdef CONFIG_PM +static int fsl_dai_runtime_resume(struct device *dev) +{ + struct fsl_dai *priv = dev_get_drvdata(dev); + int i, ret; + + for (i = 0; i < priv->num_clks; i++) { + ret = clk_prepare_enable(priv->clks[i]); + if (ret < 0) { + dev_err(dev, "Failed to enable clk %s\n", + priv->clk_names[i]); + goto out; + } + } + return 0; +out: + while (--i >= 0) + clk_disable_unprepare(priv->clks[i]); + + return ret; +} + +static int fsl_dai_runtime_suspend(struct device *dev) +{ + struct fsl_dai *priv = dev_get_drvdata(dev); + int i; + + for (i = 0; i < priv->num_clks; i++) + clk_disable_unprepare(priv->clks[i]); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_dai_pm_ops = { + SET_RUNTIME_PM_OPS(fsl_dai_runtime_suspend, + fsl_dai_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static struct platform_driver fsl_dai_driver = { + .probe = fsl_dai_probe, + .remove = fsl_dai_remove, + .driver = { + .name = "fsl-dai", + .pm = &fsl_dai_pm_ops, + .of_match_table = fsl_dai_dt_ids, + }, +}; + +module_platform_driver(fsl_dai_driver); + +MODULE_ALIAS("platform:fsl-dai"); + +MODULE_AUTHOR("Daniel Baluta <daniel.baluta(a)nxp.com>"); +MODULE_DESCRIPTION("FSL Generic DAI driver for DSP"); +MODULE_LICENSE("GPL v2"); -- 2.17.1

4 3

[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix incorrect check on p->sink
by Colin King 25 Feb '20

25 Feb '20

From: Colin Ian King <colin.king(a)canonical.com> The check on p->sink looks bogus, I believe it should be p->source since the following code blocks are related to p->source. Fix this by replacing p->sink with p->source. Addresses-Coverity: ("Copy-paste error") Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- [ Note: this has not been tested ] --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f1..f883c9340eee 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && -- 2.24.0

3 2

[alsa-devel] [PATCH 0/7] ASoC: soc-pcm cleanup step2
by Kuninori Morimoto 25 Feb '20

25 Feb '20

Hi Mark These are soc-pcm cleanup step2. Very random cleanup... Kuninori Morimoto (7): ASoC: soc-pcm: add snd_soc_runtime_action() ASoC: soc-pcm: adjustment for DAI member 0 reset ASoC: soc-pcm: add for_each_dapm_widgets() macro ASoC: soc-pcm: goto error after trying for_each_rtd_codec_dai ASoC: soc-pcm: goto error after trying all component open ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() ASoC: soc-pcm: tidyup soc_pcm_open() order include/sound/soc-dapm.h | 5 + sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 266 ++++++++++++++++++++--------------------------- 3 files changed, 118 insertions(+), 161 deletions(-) -- 2.7.4

5 17

[alsa-devel] [PATCH RFC v4 0/6] ASoC: Add Multi CPU DAI support
by Bard Liao 25 Feb '20

25 Feb '20

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v4: - fix "rtd->num_codecs > 1" error in ASoC: Add Multi CPU DAI support - split "ASoC: pcm: check if cpu DAI support the given stream" from "ASoC: Add multiple CPU DAI support for PCM ops" - add helper "ASoC: Add dapm_add_valid_dai_widget helper" for "ASoC: Add multiple CPU DAI support in DAPM" Bard Liao (2): ASoC: return error if the function is not support multi cpu yet. ASoC: pcm: check if cpu DAI support the given stream Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 + sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 205 +++++----- sound/soc/soc-dapm.c | 133 ++++--- sound/soc/soc-generic-dmaengine-pcm.c | 18 + sound/soc/soc-pcm.c | 554 ++++++++++++++++++-------- 6 files changed, 615 insertions(+), 315 deletions(-) -- 2.17.1

3 9

[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 24 Feb '20

24 Feb '20

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..80e4757176d8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

2 1

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

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[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

[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 0/9] add ASoC components for AHUB
by Sameer Pujar 19 Feb '20

19 Feb '20

Overview ======== The Audio Hub (AHUB) is part of the Audio Processing Engine (APE) which comprises a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. 1. Add YAML DT binding documentation for above mentioned modules. 2. ACIF programming is same for Tegra generations and hence it is moved to a common file. 3. Add ASoC driver components for each of the above modules. 4. Add DT entries for above components for Tegra210, Tegra186 and Tegra194. 5. Enable these components for Jetson-Tx1, Jetson-Tx2 and Jetson-Xavier. Machine driver series will be sent separately. Sameer Pujar (9): dt-bindings: sound: tegra: add DT binding for AHUB ASoC: tegra: add support for CIF programming ASoC: tegra: add Tegra210 based DMIC driver ASoC: tegra: add Tegra210 based I2S driver ASoC: tegra: add Tegra210 based AHUB driver ASoC: tegra: add Tegra186 based DSPK driver ASoC: tegra: add Tegra210 based ADMAIF driver arm64: tegra: add AHUB components for few Tegra chips arm64: tegra: enable AHUB modules for few Tegra chips .../bindings/sound/nvidia,tegra186-dspk.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 165 ++++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 130 +++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 112 +++ arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 ++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 40 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 145 ++++ sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 12 + sound/soc/tegra/tegra186_dspk.c | 516 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 73 ++ sound/soc/tegra/tegra210_admaif.c | 896 ++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 164 ++++ sound/soc/tegra/tegra210_ahub.c | 667 +++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 125 +++ sound/soc/tegra/tegra210_dmic.c | 522 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 85 ++ sound/soc/tegra/tegra210_i2s.c | 941 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 132 +++ sound/soc/tegra/tegra30_ahub.c | 94 +- sound/soc/tegra/tegra30_ahub.h | 129 --- sound/soc/tegra/tegra30_i2s.c | 35 +- sound/soc/tegra/tegra30_i2s.h | 7 - sound/soc/tegra/tegra_cif.c | 34 + sound/soc/tegra/tegra_cif.h | 50 ++ sound/soc/tegra/tegra_pcm.c | 224 ++++- sound/soc/tegra/tegra_pcm.h | 23 +- 31 files changed, 5897 insertions(+), 244 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.c create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

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