May 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 0/6] Add driver for SAMA7G5's PDMC
by Codrin Ciubotariu 05 May '22

05 May '22

This patch series adds support for Pulse Density Microphone Controller (PDMC), present on Microchip's SAMA7G5. The PDMC interfaces up to 4 digital microphones having Pulse Density Modulated (PDM) outputs. It generates a single clock line and samples 1 or 2 data lines. The signal path includes an audio grade programmable decimation filter and outputs 24-bit audio words. The source of each channel can be independently defined as PDMC_DS0 or PDMC_DS1, sampled at the rising or falling edge of PDMC_CLK. The patch series starts with a fix on the ASoC DMA engine support. Then continues with the bindings and the driver of PDMC. It is followed by the DT nodes for SAMA7G5 and SAMA7G5-EK. In the end, the drivers for PDMC and PDM microphones are enabled in sama7_defconfig. Changes in v3: - addressed new DT bindings comments from Krzysztof Kozlowski Changes in v2: - addressed DT bindings comments from Krzysztof Kozlowski Codrin Ciubotariu (6): ASoC: dmaengine: do not use a NULL prepare_slave_config() callback ASoC: dt-bindings: Document Microchip's PDMC ASoC: atmel: mchp-pdmc: add PDMC driver ARM: dts: at91: sama7g5: add nodes for PDMC ARM: dts: at91: sama7g5ek: add node for PDMC0 ARM: configs: at91: sama7_defconfig: add MCHP PDMC and DMIC drivers .../bindings/sound/microchip,pdmc.yaml | 100 ++ arch/arm/boot/dts/at91-sama7g5ek.dts | 21 +- arch/arm/boot/dts/sama7g5.dtsi | 24 + arch/arm/configs/sama7_defconfig | 2 + include/dt-bindings/sound/microchip,pdmc.h | 13 + sound/soc/atmel/Kconfig | 16 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-pdmc.c | 1084 +++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +- 9 files changed, 1264 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/microchip,pdmc.yaml create mode 100644 include/dt-bindings/sound/microchip,pdmc.h create mode 100644 sound/soc/atmel/mchp-pdmc.c -- 2.32.0

6 16

[PATCH v3 20/21] xen/sndfront: use xenbus_setup_ring() and xenbus_teardown_ring()
by Juergen Gross 05 May '22

05 May '22

Simplify sndfront's ring creation and removal via xenbus_setup_ring() and xenbus_teardown_ring(). Signed-off-by: Juergen Gross <jgross(a)suse.com> --- sound/xen/xen_snd_front_evtchnl.c | 44 +++++++------------------------ 1 file changed, 10 insertions(+), 34 deletions(-) diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c index 3e21369c8216..26d1b3987887 100644 --- a/sound/xen/xen_snd_front_evtchnl.c +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -143,12 +143,12 @@ void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *channel) static void evtchnl_free(struct xen_snd_front_info *front_info, struct xen_snd_front_evtchnl *channel) { - unsigned long page = 0; + void *page = NULL; if (channel->type == EVTCHNL_TYPE_REQ) - page = (unsigned long)channel->u.req.ring.sring; + page = channel->u.req.ring.sring; else if (channel->type == EVTCHNL_TYPE_EVT) - page = (unsigned long)channel->u.evt.page; + page = channel->u.evt.page; if (!page) return; @@ -167,10 +167,7 @@ static void evtchnl_free(struct xen_snd_front_info *front_info, xenbus_free_evtchn(front_info->xb_dev, channel->port); /* End access and free the page. */ - if (channel->gref != INVALID_GRANT_REF) - gnttab_end_foreign_access(channel->gref, page); - else - free_page(page); + xenbus_teardown_ring(&page, 1, &channel->gref); memset(channel, 0, sizeof(*channel)); } @@ -196,8 +193,7 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, enum xen_snd_front_evtchnl_type type) { struct xenbus_device *xb_dev = front_info->xb_dev; - unsigned long page; - grant_ref_t gref; + void *page; irq_handler_t handler; char *handler_name = NULL; int ret; @@ -207,12 +203,9 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, channel->index = index; channel->front_info = front_info; channel->state = EVTCHNL_STATE_DISCONNECTED; - channel->gref = INVALID_GRANT_REF; - page = get_zeroed_page(GFP_KERNEL); - if (!page) { - ret = -ENOMEM; + ret = xenbus_setup_ring(xb_dev, GFP_KERNEL, &page, 1, &channel->gref); + if (ret) goto fail; - } handler_name = kasprintf(GFP_KERNEL, "%s-%s", XENSND_DRIVER_NAME, type == EVTCHNL_TYPE_REQ ? @@ -226,33 +219,18 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, mutex_init(&channel->ring_io_lock); if (type == EVTCHNL_TYPE_REQ) { - struct xen_sndif_sring *sring = (struct xen_sndif_sring *)page; + struct xen_sndif_sring *sring = page; init_completion(&channel->u.req.completion); mutex_init(&channel->u.req.req_io_lock); - SHARED_RING_INIT(sring); - FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); - - ret = xenbus_grant_ring(xb_dev, sring, 1, &gref); - if (ret < 0) { - channel->u.req.ring.sring = NULL; - goto fail; - } + XEN_FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); handler = evtchnl_interrupt_req; } else { - ret = gnttab_grant_foreign_access(xb_dev->otherend_id, - virt_to_gfn((void *)page), 0); - if (ret < 0) - goto fail; - - channel->u.evt.page = (struct xensnd_event_page *)page; - gref = ret; + channel->u.evt.page = page; handler = evtchnl_interrupt_evt; } - channel->gref = gref; - ret = xenbus_alloc_evtchn(xb_dev, &channel->port); if (ret < 0) goto fail; @@ -279,8 +257,6 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, return 0; fail: - if (page) - free_page(page); kfree(handler_name); dev_err(&xb_dev->dev, "Failed to allocate ring: %d\n", ret); return ret; -- 2.35.3

1 0

[PATCH v3 08/21] xen/sound: switch xen_snd_front to use INVALID_GRANT_REF
by Juergen Gross 05 May '22

05 May '22

Instead of using a private macro for an invalid grant reference use the common one. Signed-off-by: Juergen Gross <jgross(a)suse.com> --- sound/xen/xen_snd_front_evtchnl.c | 4 ++-- sound/xen/xen_snd_front_evtchnl.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c index ecbc294fc59a..3e21369c8216 100644 --- a/sound/xen/xen_snd_front_evtchnl.c +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -167,7 +167,7 @@ static void evtchnl_free(struct xen_snd_front_info *front_info, xenbus_free_evtchn(front_info->xb_dev, channel->port); /* End access and free the page. */ - if (channel->gref != GRANT_INVALID_REF) + if (channel->gref != INVALID_GRANT_REF) gnttab_end_foreign_access(channel->gref, page); else free_page(page); @@ -207,7 +207,7 @@ static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, channel->index = index; channel->front_info = front_info; channel->state = EVTCHNL_STATE_DISCONNECTED; - channel->gref = GRANT_INVALID_REF; + channel->gref = INVALID_GRANT_REF; page = get_zeroed_page(GFP_KERNEL); if (!page) { ret = -ENOMEM; diff --git a/sound/xen/xen_snd_front_evtchnl.h b/sound/xen/xen_snd_front_evtchnl.h index cbe51fd1ec15..3675fba70564 100644 --- a/sound/xen/xen_snd_front_evtchnl.h +++ b/sound/xen/xen_snd_front_evtchnl.h @@ -15,15 +15,6 @@ struct xen_snd_front_info; -#ifndef GRANT_INVALID_REF -/* - * FIXME: usage of grant reference 0 as invalid grant reference: - * grant reference 0 is valid, but never exposed to a PV driver, - * because of the fact it is already in use/reserved by the PV console. - */ -#define GRANT_INVALID_REF 0 -#endif - /* Timeout in ms to wait for backend to respond. */ #define VSND_WAIT_BACK_MS 3000 -- 2.35.3

1 0

[PATCH] ASoC: cs35l45: Enable BOOST
by Richard Fitzgerald 04 May '22

04 May '22

Do not disable the boost converter during probe. The silicon contains functional default tunings so the boost converter can be left at the chip default enabled state. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 86daa2574388..2367c1a4c10e 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -587,10 +587,6 @@ static int cs35l45_initialize(struct cs35l45_private *cs35l45) return ret; } - /* Default to boost bypass */ - regmap_update_bits(cs35l45->regmap, CS35L45_BLOCK_ENABLES, CS35L45_BST_EN_MASK, - CS35L45_BST_DISABLE_FET_ON << CS35L45_BST_EN_SHIFT); - ret = cs35l45_apply_property_config(cs35l45); if (ret < 0) return ret; -- 2.30.2

2 1

[PATCH] ASoC: SOF: ipc3: Remove the ipc3-ops.h header file
by Peter Ujfalusi 04 May '22

04 May '22

Only the main IPC ops struct should be visible outside of IPC3 code to make sure that the code is correctly abstracted. Instead of keeping the ipc3-ops.h with only the high level ops struct declaration, put the ipc3_ops to sof-priv.h and move all other ops struct declaration into ipc3-priv.h New IPC implementation should follow this route: the main IPC ops should be declared in sof-priv.h and no other IPC version related header be used for generic code. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc.c | 1 - sound/soc/sof/ipc3-control.c | 2 +- sound/soc/sof/ipc3-loader.c | 1 - sound/soc/sof/ipc3-ops.h | 21 --------------------- sound/soc/sof/ipc3-pcm.c | 2 +- sound/soc/sof/ipc3-priv.h | 3 +++ sound/soc/sof/ipc3-topology.c | 2 +- sound/soc/sof/ipc3.c | 1 - sound/soc/sof/sof-priv.h | 3 +++ 9 files changed, 9 insertions(+), 27 deletions(-) delete mode 100644 sound/soc/sof/ipc3-ops.h diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index a6ec4b5a4923..41f3a217be5d 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -17,7 +17,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" -#include "ipc3-ops.h" /** * sof_ipc_send_msg - generic function to prepare and send one IPC message diff --git a/sound/soc/sof/ipc3-control.c b/sound/soc/sof/ipc3-control.c index 4347adcc6543..3fdc0d854e65 100644 --- a/sound/soc/sof/ipc3-control.c +++ b/sound/soc/sof/ipc3-control.c @@ -9,7 +9,7 @@ #include "sof-priv.h" #include "sof-audio.h" -#include "ipc3-ops.h" +#include "ipc3-priv.h" /* IPC set()/get() for kcontrols. */ static int sof_ipc3_set_get_kcontrol_data(struct snd_sof_control *scontrol, bool set) diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index 14158c52a2b7..f3c741b49519 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -9,7 +9,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ipc3-priv.h" -#include "ipc3-ops.h" #include "ops.h" static int ipc3_fw_ext_man_get_version(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/ipc3-ops.h b/sound/soc/sof/ipc3-ops.h deleted file mode 100644 index a4784626a3d7..000000000000 --- a/sound/soc/sof/ipc3-ops.h +++ /dev/null @@ -1,21 +0,0 @@ -/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ -/* - * This file is provided under a dual BSD/GPLv2 license. When using or - * redistributing this file, you may do so under either license. - * - * Copyright(c) 2021 Intel Corporation. All rights reserved. - * - * Author: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> - */ - -#ifndef __SOUND_SOC_SOF_IPC3_OPS_H -#define __SOUND_SOC_SOF_IPC3_OPS_H - -#include "sof-priv.h" - -extern const struct sof_ipc_tplg_ops ipc3_tplg_ops; -extern const struct sof_ipc_ops ipc3_ops; -extern const struct sof_ipc_tplg_control_ops tplg_ipc3_control_ops; -extern const struct sof_ipc_pcm_ops ipc3_pcm_ops; - -#endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index d7b6c67b2180..c8774a891d6f 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -8,7 +8,7 @@ // #include <sound/pcm_params.h> -#include "ipc3-ops.h" +#include "ipc3-priv.h" #include "ops.h" #include "sof-priv.h" #include "sof-audio.h" diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h index a9b9201508a5..82f9d0cbfb93 100644 --- a/sound/soc/sof/ipc3-priv.h +++ b/sound/soc/sof/ipc3-priv.h @@ -12,6 +12,9 @@ #include "sof-priv.h" /* IPC3 specific ops */ +extern const struct sof_ipc_pcm_ops ipc3_pcm_ops; +extern const struct sof_ipc_tplg_ops ipc3_tplg_ops; +extern const struct sof_ipc_tplg_control_ops tplg_ipc3_control_ops; extern const struct sof_ipc_fw_loader_ops ipc3_loader_ops; /* helpers for fw_ready and ext_manifest parsing */ diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 220ab6c6803f..043554d7cb4a 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -11,7 +11,7 @@ #include <sound/pcm_params.h> #include "sof-priv.h" #include "sof-audio.h" -#include "ipc3-ops.h" +#include "ipc3-priv.h" #include "ops.h" /* Full volume for default values */ diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index efcd201597fa..a8ffc4f99565 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -12,7 +12,6 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ipc3-priv.h" -#include "ipc3-ops.h" #include "ops.h" typedef void (*ipc3_rx_callback)(struct snd_sof_dev *sdev, void *msg_buf); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index a1141ea8d907..4801849cb2ab 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -763,4 +763,7 @@ static inline int sof_resume_clients(struct snd_sof_dev *sdev) } #endif /* CONFIG_SND_SOC_SOF_CLIENT */ +/* Main ops for IPC implementations */ +extern const struct sof_ipc_ops ipc3_ops; + #endif -- 2.36.0

2 1

[PATCH] ASoC: soc-generic-dmaengine-pcm: Use default config when none is given
by Sascha Hauer 04 May '22

04 May '22

snd_dmaengine_pcm_register() can be passed a NULL pointer for the config which means that the we have to test for pcm->config being non NULL before accessing it. Make the code more straight forward by providing a default config when none is passed. With this pcm->config is never NULL and we can skip all the if (pcm->config) tests. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 57 ++++++++++++--------------- 1 file changed, 25 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 285441d6aeed4..87858462bba99 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -79,29 +79,19 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - int (*prepare_slave_config)(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct dma_slave_config *slave_config); struct dma_slave_config slave_config; + int ret; - memset(&slave_config, 0, sizeof(slave_config)); - - if (!pcm->config) - prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; - else - prepare_slave_config = pcm->config->prepare_slave_config; + if (!pcm->config->prepare_slave_config) + return 0; - if (prepare_slave_config) { - int ret = prepare_slave_config(substream, params, &slave_config); - if (ret) - return ret; + memset(&slave_config, 0, sizeof(slave_config)); - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - } + ret = pcm->config->prepare_slave_config(substream, params, &slave_config); + if (ret) + return ret; - return 0; + return dmaengine_slave_config(chan, &slave_config); } static int @@ -121,7 +111,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, return -EINVAL; } - if (pcm->config && pcm->config->pcm_hardware) + if (pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -188,7 +178,6 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct snd_dmaengine_dai_dma_data *dma_data; - dma_filter_fn fn = NULL; if (rtd->num_cpus > 1) { dev_err(rtd->dev, @@ -201,13 +190,11 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config && pcm->config->compat_request_channel) + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - if (pcm->config) - fn = pcm->config->compat_filter_fn; - - return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); + return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, + dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct device *dev, @@ -239,12 +226,12 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, size_t max_buffer_size; unsigned int i; - if (config && config->prealloc_buffer_size) + if (config->prealloc_buffer_size) prealloc_buffer_size = config->prealloc_buffer_size; else prealloc_buffer_size = prealloc_buffer_size_kbytes * 1024; - if (config && config->pcm_hardware && config->pcm_hardware->buffer_bytes_max) + if (config->pcm_hardware && config->pcm_hardware->buffer_bytes_max) max_buffer_size = config->pcm_hardware->buffer_bytes_max; else max_buffer_size = SIZE_MAX; @@ -254,7 +241,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, if (!substream) continue; - if (!pcm->chan[i] && config && config->chan_names[i]) + if (!pcm->chan[i] && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, config->chan_names[i]); @@ -367,10 +354,10 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct dma_chan *chan; if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && - !(config && config->dma_dev && config->dma_dev->of_node))) + !(config->dma_dev && config->dma_dev->of_node))) return 0; - if (config && config->dma_dev) { + if (config->dma_dev) { /* * If this warning is seen, it probably means that your Linux * device structure does not match your HW device structure. @@ -387,7 +374,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; - if (config && config->chan_names[i]) + if (config->chan_names[i]) name = config->chan_names[i]; chan = dma_request_chan(dev, name); if (IS_ERR(chan)) { @@ -425,6 +412,10 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) } } +static const struct snd_dmaengine_pcm_config snd_dmaengine_pcm_default_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -445,6 +436,8 @@ int snd_dmaengine_pcm_register(struct device *dev, #ifdef CONFIG_DEBUG_FS pcm->component.debugfs_prefix = "dma"; #endif + if (!config) + config = &snd_dmaengine_pcm_default_config; pcm->config = config; pcm->flags = flags; @@ -452,7 +445,7 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_dma; - if (config && config->process) + if (config->process) driver = &dmaengine_pcm_component_process; else driver = &dmaengine_pcm_component; -- 2.30.2

2 1

ASoC component/card relationship
by Pierre-Louis Bossart 04 May '22

04 May '22

Hi, In the existing ASoC code, there is a fixed mapping between ASoC card and component. A component relies on a ->card pointer that is set during the probe. A component cannot be used by or "bound to" more than one card [1] This has interesting impacts on how a codec or DSP driver need to be implemented. In the AVS series posted this week, multiple components are registered by the DSP driver, following an interface-based split. There's in addition a second-level split, where the logic is pushed further: the DSP driver partitions the SSP DAIs in different set of 'dai_driver's used by different components, which are in turn used by different cards. What is done in these patches is not wrong, and is probably the only solution to support a real-world platform with the existing ASoC code, but are the framework assumptions correct? In this example, the board-level information on which interface is used for what functionality trickles down to the lowest level of the DSP driver implementation. I believe this breaks to some extent the 'clean' split between platform and machine driver(s), and it's not quite aligned with the usual notion of register/probe used across frameworks, be it for drivers/clocks/you name it. A similar case could happen in a codec driver, if independent functionality such as headset and amplifier support was exposed by separate cards, that would in turn mandate that the codec driver exposed N components, each handling different functionality but the same type of DAI. An alternative approach would be that the DSP driver exposes all the possible DAIs that can be used, and the binding is refined to allow for more flexibility. I think it's really the individual DAI that cannot be used by more than one card. I figured I would ask on this mailing list if a) I am not mistaken on the component/card relationship and b) if this is by design, or if we want to clarify what a component is and what its restrictions might be. Thanks for your feedback/comments -Pierre [1] https://elixir.bootlin.com/linux/latest/source/sound/soc/soc-core.c#L1364

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[PATCH v5 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 04 May '22

04 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v5: - Reword and add to description in dt-binding - Change 'speed' dt property to 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on `filemode` to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../devicetree/bindings/sound/serialmidi.yaml | 46 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 377 ++++++++++++++++++ 4 files changed, 443 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serialmidi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: 3e71713c9e75c34fc03f55ea86b381856ca952ee -- 2.33.0

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Re: [PATCH] ASoC: Intel: sof_cs42l42: add support for adl_mx98360a_cs4242 and BT offload audio
by Cezary Rojewski 04 May '22

04 May '22

On 2022-05-04 2:24 PM, Terry Chen wrote: > This patch adds driver data for adl_mx98360a_cs4242 which supports > two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec > on SSP0 running on ADL platform. Also add the capability to machine > driver of creating DAI Link for BT offload. Although BT offload > always uses SSP2 port but we reserve the flexibility to assign > the port number in macro. Title's length seems off. Please shorten it a bit. Commit message formatting (line width) is not cohevise across the lines either. Another suggestion: skip redundant bits at the front e.g.: s/This patch adds/Add/. When adding new code (new functionality), it's good to plainly state: "To be able to do X we add capability Y". The last sentence in your message is a side note. It's good that you're being transparent here, my only advice is that you add a new line before such notes as these are not the main dish here, really. Title suggest you're doing two things here - adding support for some ADL configuration (1) and BT audio offload (2). Recommendation is to splitting non-atomic changes so each patch in the series answers single problem. > Signed-off-by: Terry Chen <terry_chen(a)wistron.corp-partner.google.com> > --- > sound/soc/intel/boards/sof_cs42l42.c | 88 ++++++++++++++++++- > .../intel/common/soc-acpi-intel-adl-match.c | 8 ++ > 2 files changed, 92 insertions(+), 4 deletions(-) > ... > @@ -278,6 +284,13 @@ static struct snd_soc_dai_link_component dmic_component[] = { > } > }; > > +static struct snd_soc_dai_link_component dummy_component[] = { > + { > + .name = "snd-soc-dummy", > + .dai_name = "snd-soc-dummy-dai", > + } > +}; Isn't there some SND_SOC_DAILINK_DEF() macro for that already? > static int create_spk_amp_dai_links(struct device *dev, > struct snd_soc_dai_link *links, > struct snd_soc_dai_link_component *cpus, > @@ -467,9 +480,52 @@ static int create_hdmi_dai_links(struct device *dev, > return -ENOMEM; > } > > +static int create_bt_offload_dai_links(struct device *dev, > + struct snd_soc_dai_link *links, > + struct snd_soc_dai_link_component *cpus, > + int *id, int ssp_bt) > +{ > + int ret = 0; Looks like there is no need for 'ret' variable here. > + > + /* bt offload */ > + if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) > + return 0; > + > + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", > + ssp_bt); Current limit is 100. > + if (!links[*id].name) > + goto devm_err; This 'goto' statement seems redundant. I might be missing something but returning -ENOMEM should suffice. Label 'devm_err' does not do anything special anyway. > + > + links[*id].id = *id; > + links[*id].codecs = dummy_component; > + links[*id].num_codecs = ARRAY_SIZE(dummy_component); > + links[*id].platforms = platform_component; > + links[*id].num_platforms = ARRAY_SIZE(platform_component); > + > + links[*id].dpcm_playback = 1; > + links[*id].dpcm_capture = 1; > + links[*id].no_pcm = 1; > + links[*id].cpus = &cpus[*id]; > + links[*id].num_cpus = 1; > + > + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, > + "SSP%d Pin", > + ssp_bt); Ditto - limit 100. Applies to every single occurence below too. > + if (!links[*id].cpus->dai_name) > + goto devm_err; > + > + (*id)++; Are the parenthesis needed? Should be save to drop them. > + > + return 0; > + > +devm_err: > + return ret; Once 'ret/devm_err' situation is clarified, this block can be dropped. > +}

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pcm: multi: return correct hwptr and avail from snd_pcm_multi_status()
by GitHub pull_request - opened 04 May '22

04 May '22

alsa-project/alsa-lib pull request #224 was opened from infmagic2047: Ensure the logic of snd_pcm_multi_status() is consistent with snd_pcm_multi_avail_update(). Signed-off-by: Yutao Yuan <infmagic2047reg(a)outlook.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/224 Patch URL : https://github.com/alsa-project/alsa-lib/pull/224.patch Repository URL: https://github.com/alsa-project/alsa-lib

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