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[PATCH] Revert "ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type"
by Takashi Iwai 04 Nov '21

04 Nov '21

This reverts commit 2d9ea39917a4e4293bc2caea902c7059a330b611. We've got a regression report showing that the audio got broken the device over AMD IOMMU. The conversion assumed the wrong pointer / page mapping for the indirect mapping case, and we need to correct this urgently, so let's revert it for now. Fixes: 2d9ea39917a4 ("ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type") Reported-and-tested-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 14 +-- sound/core/Makefile | 1 + sound/core/memalloc.c | 51 +--------- sound/core/sgbuf.c | 201 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 213 insertions(+), 54 deletions(-) create mode 100644 sound/core/sgbuf.c diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 653dfffb3ac8..1051b84e8579 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,6 +36,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -44,13 +51,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif /* * info for buffer allocation diff --git a/sound/core/Makefile b/sound/core/Makefile index 350d704ced98..79e1407cd0de 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -19,6 +19,7 @@ snd-$(CONFIG_SND_JACK) += ctljack.o jack.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o +snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 99cd0f67daa1..ea778f868cf3 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -560,50 +560,6 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_vmalloc_get_chunk_size, }; -/* x86-specific SG-buffer with WC pages */ -#ifdef CONFIG_SND_DMA_SGBUF -#define vmalloc_to_virt(v) (unsigned long)page_to_virt(vmalloc_to_page(v)) - -static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - void *p = snd_dma_noncontig_alloc(dmab, size); - size_t ofs; - - if (!p) - return NULL; - for (ofs = 0; ofs < size; ofs += PAGE_SIZE) - set_memory_uc(vmalloc_to_virt(p + ofs), 1); - return p; -} - -static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) -{ - size_t ofs; - - for (ofs = 0; ofs < dmab->bytes; ofs += PAGE_SIZE) - set_memory_wb(vmalloc_to_virt(dmab->area + ofs), 1); - snd_dma_noncontig_free(dmab); -} - -static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - /* FIXME: dma_mmap_noncontiguous() works? */ - return -ENOENT; /* continue with the default mmap handler */ -} - -const struct snd_malloc_ops snd_dma_sg_wc_ops = { - .alloc = snd_dma_sg_wc_alloc, - .free = snd_dma_sg_wc_free, - .mmap = snd_dma_sg_wc_mmap, - .sync = snd_dma_noncontig_sync, - .get_addr = snd_dma_vmalloc_get_addr, - .get_page = snd_dma_vmalloc_get_page, - .get_chunk_size = snd_dma_vmalloc_get_chunk_size, -}; -#endif /* CONFIG_SND_DMA_SGBUF */ - /* * Non-coherent pages allocator */ @@ -663,13 +619,14 @@ static const struct snd_malloc_ops *dma_ops[] = { [SNDRV_DMA_TYPE_DEV_WC] = &snd_dma_wc_ops, [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, -#endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ +#ifdef CONFIG_SND_DMA_SGBUF + [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, +#endif }; static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c new file mode 100644 index 000000000000..8352a5cdb19f --- /dev/null +++ b/sound/core/sgbuf.c @@ -0,0 +1,201 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Scatter-Gather buffer + * + * Copyright (c) by Takashi Iwai <tiwai(a)suse.de> + */ + +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/vmalloc.h> +#include <linux/export.h> +#include <sound/memalloc.h> +#include "memalloc_local.h" + +struct snd_sg_page { + void *buf; + dma_addr_t addr; +}; + +struct snd_sg_buf { + int size; /* allocated byte size */ + int pages; /* allocated pages */ + int tblsize; /* allocated table size */ + struct snd_sg_page *table; /* address table */ + struct page **page_table; /* page table (for vmap/vunmap) */ + struct device *dev; +}; + +/* table entries are align to 32 */ +#define SGBUF_TBL_ALIGN 32 +#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN) + +static void snd_dma_sg_free(struct snd_dma_buffer *dmab) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + struct snd_dma_buffer tmpb; + int i; + + if (!sgbuf) + return; + + vunmap(dmab->area); + dmab->area = NULL; + + tmpb.dev.type = SNDRV_DMA_TYPE_DEV; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + tmpb.dev.type = SNDRV_DMA_TYPE_DEV_WC; + tmpb.dev.dev = sgbuf->dev; + for (i = 0; i < sgbuf->pages; i++) { + if (!(sgbuf->table[i].addr & ~PAGE_MASK)) + continue; /* continuous pages */ + tmpb.area = sgbuf->table[i].buf; + tmpb.addr = sgbuf->table[i].addr & PAGE_MASK; + tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; + snd_dma_free_pages(&tmpb); + } + + kfree(sgbuf->table); + kfree(sgbuf->page_table); + kfree(sgbuf); + dmab->private_data = NULL; +} + +#define MAX_ALLOC_PAGES 32 + +static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + struct snd_sg_buf *sgbuf; + unsigned int i, pages, chunk, maxpages; + struct snd_dma_buffer tmpb; + struct snd_sg_page *table; + struct page **pgtable; + int type = SNDRV_DMA_TYPE_DEV; + pgprot_t prot = PAGE_KERNEL; + void *area; + + dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); + if (!sgbuf) + return NULL; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { + type = SNDRV_DMA_TYPE_DEV_WC; +#ifdef pgprot_noncached + prot = pgprot_noncached(PAGE_KERNEL); +#endif + } + sgbuf->dev = dmab->dev.dev; + pages = snd_sgbuf_aligned_pages(size); + sgbuf->tblsize = sgbuf_align_table(pages); + table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); + if (!table) + goto _failed; + sgbuf->table = table; + pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); + if (!pgtable) + goto _failed; + sgbuf->page_table = pgtable; + + /* allocate pages */ + maxpages = MAX_ALLOC_PAGES; + while (pages > 0) { + chunk = pages; + /* don't be too eager to take a huge chunk */ + if (chunk > maxpages) + chunk = maxpages; + chunk <<= PAGE_SHIFT; + if (snd_dma_alloc_pages_fallback(type, dmab->dev.dev, + chunk, &tmpb) < 0) { + if (!sgbuf->pages) + goto _failed; + size = sgbuf->pages * PAGE_SIZE; + break; + } + chunk = tmpb.bytes >> PAGE_SHIFT; + for (i = 0; i < chunk; i++) { + table->buf = tmpb.area; + table->addr = tmpb.addr; + if (!i) + table->addr |= chunk; /* mark head */ + table++; + *pgtable++ = virt_to_page(tmpb.area); + tmpb.area += PAGE_SIZE; + tmpb.addr += PAGE_SIZE; + } + sgbuf->pages += chunk; + pages -= chunk; + if (chunk < maxpages) + maxpages = chunk; + } + + sgbuf->size = size; + area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); + if (!area) + goto _failed; + return area; + + _failed: + snd_dma_sg_free(dmab); /* free the table */ + return NULL; +} + +static dma_addr_t snd_dma_sg_get_addr(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + dma_addr_t addr; + + addr = sgbuf->table[offset >> PAGE_SHIFT].addr; + addr &= ~((dma_addr_t)PAGE_SIZE - 1); + return addr + offset % PAGE_SIZE; +} + +static struct page *snd_dma_sg_get_page(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + unsigned int idx = offset >> PAGE_SHIFT; + + if (idx >= (unsigned int)sgbuf->pages) + return NULL; + return sgbuf->page_table[idx]; +} + +static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, + unsigned int ofs, + unsigned int size) +{ + struct snd_sg_buf *sg = dmab->private_data; + unsigned int start, end, pg; + + start = ofs >> PAGE_SHIFT; + end = (ofs + size - 1) >> PAGE_SHIFT; + /* check page continuity */ + pg = sg->table[start].addr >> PAGE_SHIFT; + for (;;) { + start++; + if (start > end) + break; + pg++; + if ((sg->table[start].addr >> PAGE_SHIFT) != pg) + return (start << PAGE_SHIFT) - ofs; + } + /* ok, all on continuous pages */ + return size; +} + +static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return -ENOENT; /* continue with the default mmap handler */ +} + +const struct snd_malloc_ops snd_dma_sg_ops = { + .alloc = snd_dma_sg_alloc, + .free = snd_dma_sg_free, + .get_addr = snd_dma_sg_get_addr, + .get_page = snd_dma_sg_get_page, + .get_chunk_size = snd_dma_sg_get_chunk_size, + .mmap = snd_dma_sg_mmap, +}; -- 2.26.2

1 0

[PATCH] ALSA: firewire-motu: add support for MOTU Traveler mk3
by Takashi Sakamoto 04 Nov '21

04 Nov '21

Mark of the Unicorn (MOTU) shipped Traveler mk3 as one of models in third generation of its FireWire series, and discontinued it already. The model consists of below ICs: * Texas Instruments TSB41AB2 * Phillips Semiconductors PDI1394L40 * Altera cyclone EP1C3 * Texas Instruments TMS320VC5402 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and whether to enable ADAT channels in rx/tx packets. The model supports transmission of PCM frames as well as MIDI messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. $ cd linux-firewire-tools/src $ python crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0410af0a bus_info_length 4, crc_length 16, crc 44810 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 00090911 device_id 0000090911 | EUI-64 0001f20000090911 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 00031733 directory_length 3, crc 5939 42c 120001f2 specifier id 430 1300001b version 434 17102800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 00028484 leaf_length 2, crc 33924 43c 0001f200 company_id 0001f2 | 440 00090911 device_id 0000090911 | EUI-64 0001f20000090911 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 22 +++++++++++++++++++--- sound/firewire/motu/motu.c | 1 + sound/firewire/motu/motu.h | 1 + 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 05608e8ca0bc..1371f38505d0 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -16,6 +16,7 @@ #define V3_CLOCK_SRC_INTERNAL 0x00 #define V3_CLOCK_SRC_WORD_ON_BNC 0x01 #define V3_CLOCK_SRC_SPH 0x02 +#define V3_CLOCK_SRC_AESEBU_ON_XLR 0x08 #define V3_CLOCK_SRC_SPDIF_ON_COAX 0x10 #define V3_CLOCK_SRC_OPT_IFACE_A 0x18 #define V3_CLOCK_SRC_OPT_IFACE_B 0x19 @@ -126,6 +127,9 @@ int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu, case V3_CLOCK_SRC_SPH: *src = SND_MOTU_CLOCK_SOURCE_SPH; break; + case V3_CLOCK_SRC_AESEBU_ON_XLR: + *src = SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR; + break; case V3_CLOCK_SRC_SPDIF_ON_COAX: *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; break; @@ -185,7 +189,7 @@ int snd_motu_protocol_v3_switch_fetching_mode(struct snd_motu *motu, sizeof(reg)); } -static int detect_packet_formats_828mk3(struct snd_motu *motu, u32 data) +static int detect_packet_formats_dual_opt_iface(struct snd_motu *motu, u32 data) { if (data & V3_ENABLE_OPT_IN_IFACE_A) { if (data & V3_NO_ADAT_OPT_IN_IFACE_A) { @@ -255,8 +259,10 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) motu->spec->rx_fixed_pcm_chunks, sizeof(motu->rx_packet_formats.pcm_chunks)); - if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) - return detect_packet_formats_828mk3(motu, data); + if (motu->spec == &snd_motu_spec_828mk3_fw || + motu->spec == &snd_motu_spec_828mk3_hybrid || + motu->spec == &snd_motu_spec_traveler_mk3) + return detect_packet_formats_dual_opt_iface(motu, data); else return 0; } @@ -281,6 +287,16 @@ const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. }; +const struct snd_motu_spec snd_motu_spec_traveler_mk3 = { + .name = "TravelerMk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q | + SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {18, 14, 10}, + .rx_fixed_pcm_chunks = {14, 14, 10}, +}; + const struct snd_motu_spec snd_motu_spec_ultralite_mk3 = { .name = "UltraLiteMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 5fc7ae475537..080728dd278f 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -169,6 +169,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x00001b, &snd_motu_spec_traveler_mk3), SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 79704ae6a73e..6ad1feab478b 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -138,6 +138,7 @@ extern const struct snd_motu_spec snd_motu_spec_8pre; extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; +extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; extern const struct snd_motu_spec snd_motu_spec_4pre; -- 2.32.0

2 1

[PATCH] ASoC: SOF:control: Fix variable type in snd_sof_refresh_control()
by Peter Ujfalusi 04 Nov '21

04 Nov '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> The second parameter for snd_sof_ipc_set_get_comp_data() is ipc_cmd, not ipc_ctrl_type and the type is u32. Fixes: 756bbe4205bc6 ("ASoC: SOF: Handle control change notification from firmware") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- sound/soc/sof/control.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 58bb89af4de1..bb1dfe4f6d40 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -69,7 +69,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) { struct sof_ipc_ctrl_data *cdata = scontrol->control_data; struct snd_soc_component *scomp = scontrol->scomp; - enum sof_ipc_ctrl_type ctrl_type; + u32 ipc_cmd; int ret; if (!scontrol->comp_data_dirty) @@ -79,9 +79,9 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) return; if (scontrol->cmd == SOF_CTRL_CMD_BINARY) - ctrl_type = SOF_IPC_COMP_GET_DATA; + ipc_cmd = SOF_IPC_COMP_GET_DATA; else - ctrl_type = SOF_IPC_COMP_GET_VALUE; + ipc_cmd = SOF_IPC_COMP_GET_VALUE; /* set the ABI header values */ cdata->data->magic = SOF_ABI_MAGIC; @@ -89,7 +89,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) /* refresh the component data from DSP */ scontrol->comp_data_dirty = false; - ret = snd_sof_ipc_set_get_comp_data(scontrol, ctrl_type, + ret = snd_sof_ipc_set_get_comp_data(scontrol, ipc_cmd, SOF_CTRL_TYPE_VALUE_CHAN_GET, scontrol->cmd, false); if (ret < 0) { -- 2.33.1

3 2

[PATCH v3 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 04 Nov '21

04 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlöf (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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Creative Soundblaster AE-7 Headphones Issue
by Emre Fidan 04 Nov '21

04 Nov '21

Hi There, Headphones connected to Creative Soundblaster AE-7 are normally auto-detected on alsamixer, but sink shows the headphone port as not available. alsamixer can change speakers to headphones and vice-versa, however, the audio silences on the headphones after a couple of seconds unless you keep system sound settings/easyeffects/pavucontol opened or minimized. acpi_listen reports Line-In plugged/unplugged instead of headphones, trying to reassign the with hdajackretask does the sensing (acpi_listen shows headphones plugged/unplugged), however, no sound from headphones this time at all. Any suggestions? Thanks. -- Emre Fidan

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HDA-Intel: fixes for Headphone+LO variant (Realtek ALC233)
by GitHub pull_request - opened 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf pull request #116 was opened from perexg: Issue: https://github.com/alsa-project/alsa-ucm-conf/issues/114 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/116 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/116.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: Description: An issue with the headphone detection (Speaker/Headphone auto-mute). More info: https://github.com/thesofproject/linux/issues/3088 and https://bugzilla.suse.com/show_bug.cgi?id=1188685 Mixer: ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 # Switch only Simple mixer control 'Headphone+LO',0 # Volume only Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 # Switch only Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` Jacks: ``` name 'Mic Jack' name 'Front Mic Jack' name 'Line Out Jack' name 'Front Headphone Jack' name 'Speaker Phantom Jack' name 'HDMI/DP,pcm=3 Jack' name 'HDMI/DP,pcm=4 Jack' name 'HDMI/DP,pcm=5 Jack' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: Description: An issue with the headphone detection (Speaker/Headphone auto-mute). More info: https://github.com/thesofproject/linux/issues/3088 and https://bugzilla.suse.com/show_bug.cgi?id=1188685 ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 # Switch only Simple mixer control 'Headphone+LO',0 # Volume only Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 # Switch only Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` ``` name 'Mic Jack' name 'Front Mic Jack' name 'Line Out Jack' name 'Front Headphone Jack' name 'Speaker Phantom Jack' name 'HDMI/DP,pcm=3 Jack' name 'HDMI/DP,pcm=4 Jack' name 'HDMI/DP,pcm=5 Jack' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: fix flexible_array.cocci warning
by Guo Zhengkui 04 Nov '21

04 Nov '21

Fix following coccicheck warnings: ./sound/pci/hda/patch_realtek.c:128:4-24: WARNING use flexible-array member instead. There is a regular need in the kernel to provide a way to declare having a dynamically sized set of trailing elements in a structure. Kernel code should always use “flexible array members” for these cases. The older style of one-element or zero-length arrays should no longer be used. Signed-off-by: Guo Zhengkui <guozhengkui(a)vivo.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6322fac9e694..7935c363092e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -125,7 +125,7 @@ struct alc_spec { unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; struct input_dev *kb_dev; - u8 alc_mute_keycode_map[1]; + u8 alc_mute_keycode_map[]; }; /* -- 2.20.1

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Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: More info: https://github.com/thesofproject/linux/issues/3088 ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Headphone+LO',0 Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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