February 2024 - Alsa-devel - mailman.alsa-project.org

[GIT PULL] ASoC fixes for v6.8-rc5
by Mark Brown 29 Feb '24

29 Feb '24

The following changes since commit 0db0c1770834f39e11a2902e20e1f11a482f4465: ASoC: cs35l56: Workaround for ACPI with broken spk-id-gpios property (2024-02-13 19:37:25 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-rc5 for you to fetch changes up to ed00a6945dc32462c2d3744a3518d2316da66fcc: ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 (2024-02-28 12:24:59 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 A few small fixes, some driver specific and one slightly larger one from Richard which adds a new core helper and updates a small clutch of drivers to deal with the fact that they were using a helper which requires that the lock for the list of controls without holding that lock. We also have some quirks for new AMD based Lenovo systems. ---------------------------------------------------------------- Colin Ian King (1): ASoC: qcom: Fix uninitialized pointer dmactl Jiawei Wang (2): ASoC: amd: yc: add new YC platform variant (0x63) support ASoC: amd: yc: Fix non-functional mic on Lenovo 21J2 Johnny Hsieh (1): ASoC: amd: yc: Add Lenovo ThinkBook 21J0 into DMI quirk table Richard Fitzgerald (2): ASoC: cs35l56: Must clear HALO_STATE before issuing SYSTEM_RESET ASoC: soc-card: Fix missing locking in snd_soc_card_get_kcontrol() include/sound/soc-card.h | 2 ++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/amd/yc/pci-acp6x.c | 1 + sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 12 +++++++++++- sound/soc/qcom/lpass-cdc-dma.c | 2 +- sound/soc/soc-card.c | 24 ++++++++++++++++++++++-- 9 files changed, 54 insertions(+), 6 deletions(-)

2 1

alsa cant initialize on raspberry pi 5
by GitHub issues - opened 29 Feb '24

29 Feb '24

alsa-project/alsa-lib issue #386 was opened from sy-project: I can use alsa command ``` # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplusadcpro], device 0: HiFiBerry DAC+ADC Pro HiFi multicodec-0 [HiFiBerry DAC+ADC Pro HiFi multicodec-0] Subdevices: 1/1 Subdevice #0: subdevice #0 # arecord -l **** List of CAPTURE Hardware Devices **** card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplusadcpro], device 0: HiFiBerry DAC+ADC Pro HiFi multicodec-0 [HiFiBerry DAC+ADC Pro HiFi multicodec-0] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` and I can aplay to use wav file. ``` # aplay -D hw:0,0 /root/C_Code/audiotest/test1.wav Playing WAVE '/root/C_Code/audiotest/test1.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo ``` but I cant initialize on c code. my code is here. ```c printf("alsa start\n"); int err; snd_pcm_t *capture_handle, *playback_handle; snd_pcm_hw_params_t *hw_params; unsigned char *buffer; int sample_rate = SAMPLE_RATE; int dir; snd_pcm_uframes_t frames = 32; snd_pcm_uframes_t bufferSize, periodSize; double x_prev = 0.0; double y_prev = 0.0; double b0 = 1.0; double b1 = -0.8; double a1 = 0.0; printf("alsa open capture handle\n"); err = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0); if(err < 0) { fprintf(stderr, "Capture open error: %s\n", snd_strerror(err)); exit(1); } printf("alsa open playback handle\n"); err = snd_pcm_open(&playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if(err < 0) { fprintf(stderr, "Playback open error: %s\n", snd_strerror(err)); exit(1); } printf("alsa pcm hw param alloc\n"); snd_pcm_hw_params_alloca(&hw_params); printf("alsa capture_handle set start\n"); err = snd_pcm_hw_params_any(capture_handle, hw_params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_access cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_format cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, NUM_CHANNELS); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_channels cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &sample_rate, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_rate_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_buffer_size_near(capture_handle, hw_params, &bufferSize); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_buffer_size_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, &periodSize, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_set_period_size_near cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } err = snd_pcm_hw_params(capture_handle, hw_params); if (err < 0) { fprintf(stderr, "alsa cant set hw params. [%s]\n", snd_strerror(err)); exit(1); } snd_pcm_hw_params_get_buffer_size(hw_params, &bufferSize); snd_pcm_hw_params_get_period_size(hw_params, &periodSize, &dir); printf("Buffer size: %lu frames\n", bufferSize); printf("Period size: %lu frames\n", periodSize); printf("alsa playback_handle set start\n"); snd_pcm_hw_params_any(playback_handle, hw_params); snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(playback_handle, hw_params, NUM_CHANNELS); snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &sample_rate, 0); snd_pcm_hw_params(playback_handle, hw_params); printf("alsa playback_handle set success\n"); printf("alsa buf malloc\n"); buffer = (unsigned char*)malloc(BUFFER_SIZE*NUM_CHANNELS*2); while(1) { printf("main loop start\n"); err = snd_pcm_readi(capture_handle, buffer, BUFFER_SIZE); if(err<0) { fprintf(stderr, "Capture read error[%d]: %s\n", err, snd_strerror(err)); break; } for(int i = 0; i < BUFFER_SIZE * NUM_CHANNELS * 2; i += 2) { short sample = ((short)buffer[i + 1] << 8) | buffer[i]; double x = (double)sample / 32768.0; double y = b0 * x + b1 * x_prev + a1 * y_prev; x_prev = x; y_prev = y; short filter_sample = (short)(y * 32768.0); buffer[i] = (unsigned char)(filter_sample & 0xFF); buffer[i+1] = (unsigned char)((filter_sample >> 8) & 0xFF); } err = snd_pcm_writei(playback_handle, buffer, BUFFER_SIZE); if(err < 0) { fprintf(stderr, "Playback write error: %s\n", snd_strerror(err)); break; } } free(buffer); snd_pcm_close(capture_handle); snd_pcm_close(playback_handle); exit(0); ``` This code said ``` alsa start alsa open capture handle alsa open playback handle alsa pcm hw param alloc alsa capture_handle set start snd_pcm_hw_params_set_buffer_size_near cant set hw params. [Invalid argument] ``` what should I do? Issue URL : https://github.com/alsa-project/alsa-lib/issues/386 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: dice: add stream format parameters for Weiss devices
by Michele Perrone 29 Feb '24

29 Feb '24

Hard-coded stream format parameters are added for Weiss Engineering FireWire devices. When the device vendor and model match, the parameters are copied into the stream format cache. This allows for setting all supported sampling rates up to 192kHz, and consequently adjusting the number of available I/O channels. Signed-off-by: Rolf Anderegg <rolf.anderegg(a)weiss.ch> Signed-off-by: Michele Perrone <michele.perrone(a)weiss.ch> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-weiss.c | 120 +++++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 72 +++++++++++++++++++ 3 files changed, 193 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-weiss.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index a5f3fbf28b8c..bac8712f9014 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -2,5 +2,5 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ - dice-harman.o dice-focusrite.o + dice-harman.o dice-focusrite.o dice-weiss.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-weiss.c b/sound/firewire/dice/dice-weiss.c new file mode 100644 index 000000000000..ad22bfafd324 --- /dev/null +++ b/sound/firewire/dice/dice-weiss.c @@ -0,0 +1,120 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-weiss.c - a part of driver for DICE based devices + * + * Copyright (c) 2023 Rolf Anderegg and Michele Perrone + */ + +#include "dice.h" + +struct dice_weiss_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +/* Weiss DAC202: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss MAN301: 192kHz 2-channel music archive network player */ +static const struct dice_weiss_spec man301 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int202 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{0, 0, 0}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ +static const struct dice_weiss_spec int203 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ +static const struct dice_weiss_spec adc2 = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ +static const struct dice_weiss_spec dac2_minerva = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ +static const struct dice_weiss_spec vesta = { + .tx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .rx_pcm_chs = {{2, 2, 2}, {0, 0, 0} }, + .has_midi = false +}; + +/* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ +static const struct dice_weiss_spec afi1 = { + .tx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .rx_pcm_chs = {{24, 16, 8}, {0, 0, 0} }, + .has_midi = false +}; + +int snd_dice_detect_weiss_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_weiss_spec *spec; + } *entry, entries[] = { + {0x000007, &dac202}, + {0x000008, &dac202}, /* Maya edition: same audio I/O as DAC202 */ + {0x000006, &int202}, + {0x00000a, &int203}, + {0x00000b, &man301}, + {0x000001, &adc2}, + {0x000003, &dac2_minerva}, + {0x000002, &vesta}, + {0x000004, &afi1}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 6c93e6e4982c..4bcd026a0b3f 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -396,6 +396,78 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* Weiss DAC202: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000007, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC202: 192kHz 2-channel DAC (Maya edition) */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss MAN301: 192kHz 2-channel music archive network player */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000b, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT202: 192kHz unidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000006, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss INT203: 192kHz bidirectional 2-channel digital Firewire interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x00000a, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss ADC2: 192kHz A/D converter with microphone preamps and line inputs */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000001, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss DAC2/Minerva: 192kHz 2-channel DAC */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000003, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss Vesta: 192kHz 2-channel Firewire to AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, + /* Weiss AFI1: 192kHz 24-channel Firewire to ADAT or AES/EBU interface */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_WEISS, + .model_id = 0x000004, + .driver_data = (kernel_ulong_t)snd_dice_detect_weiss_formats, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); base-commit: 9b4469410cf9a0fcbccc92c480fd42f7c815a745 -- 2.41.0

3 26

[PATCH] Convert the Imagination Technologies SPDIF Input Controllerto DT schema.
by Javier García 28 Feb '24

28 Feb '24

Convert the Imagination Technologies SPDIF Input Controllerto DT schema. Signed-off-by: Javier García <javier.garcia.ta(a)udima.es> --- .../bindings/sound/img,spdif-in.txt | 41 ---------- .../bindings/sound/img,spdif-in.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 41 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.yaml diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.txt b/Documentation/devicetree/bindings/sound/img,spdif-in.txt deleted file mode 100644 index f7ea8c87bf34..000000000000 --- a/Documentation/devicetree/bindings/sound/img,spdif-in.txt +++ /dev/null @@ -1,41 +0,0 @@ -Imagination Technologies SPDIF Input Controller - -Required Properties: - - - compatible : Compatible list, must contain "img,spdif-in" - - - #sound-dai-cells : Must be equal to 0 - - - reg : Offset and length of the register set for the device - - - dmas: Contains an entry for each entry in dma-names. - - - dma-names: Must include the following entry: - "rx" - - - clocks : Contains an entry for each entry in clock-names - - - clock-names : Includes the following entries: - "sys" The system clock - -Optional Properties: - - - resets: Should contain a phandle to the spdif in reset signal, if any - - - reset-names: Should contain the reset signal name "rst", if a - reset phandle is given - - - interrupts : Contains the spdif in interrupt, if present - -Example: - -spdif_in: spdif-in@18100e00 { - compatible = "img,spdif-in"; - reg = <0x18100E00 0x100>; - interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&mdc 15 0xffffffff 0>; - dma-names = "rx"; - clocks = <&cr_periph SYS_CLK_SPDIF_IN>; - clock-names = "sys"; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/img,spdif-in.yaml b/Documentation/devicetree/bindings/sound/img,spdif-in.yaml new file mode 100644 index 000000000000..d201293d63c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/img,spdif-in.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/img,spdif-in.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Imagination Technologies SPDIF Input Controller + +maintainers: + - Rob Herring <robh+dt(a)kernel.org> + - Krzysztof Kozlowski <krzysztof.kozlowski+dt(a)linaro.org> + +properties: + compatible: + enum: + - img,spdif-in + + reg: + description: + Offset and length of the register set for the device. + maxItems: 1 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + items: + - const: rx + + clocks: + items: + - description: The system clock + + clock-names: + items: + - const: sys + + '#sound-dai-cells': + const: 0 + + resets: + items: + - description: Should contain a phandle to the spdif in reset signal, if any + + reset-names: + items: + - const: rst + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + - '#sound-dai-cells' + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/clock/pistachio-clk.h> + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/interrupt-controller/mips-gic.h> + #include <dt-bindings/reset/pistachio-resets.h> + spdif_in: spdif-in@18100e00 { + compatible = "img,spdif-in"; + reg = <0x18100E00 0x100>; + interrupts = <GIC_SHARED 20 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&mdc 15 0xffffffff 0>; + dma-names = "rx"; + clocks = <&cr_periph SYS_CLK_SPDIF_IN>; + clock-names = "sys"; + + #sound-dai-cells = <0>; + }; -- 2.43.0

4 9

[PATCH v2] ASoC: meson: axg-fifo: use FIELD helpers
by Jerome Brunet 27 Feb '24

27 Feb '24

Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Changes since v1: [1] * Add missing include <linux/bitfield.h> causing build error with x86 compile test. [1]: http://lore.kernel.org/alsa-devel/20240223175116.2005407-7-jbrunet@baylibre… sound/soc/meson/axg-fifo.c | 25 +++++++++++++------------ sound/soc/meson/axg-fifo.h | 12 +++++------- sound/soc/meson/axg-frddr.c | 5 +++-- sound/soc/meson/axg-toddr.c | 22 ++++++++++------------ 4 files changed, 31 insertions(+), 33 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 65541fdb0038..bebee0ca8e38 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -3,6 +3,7 @@ // Copyright (c) 2018 BayLibre, SAS. // Author: Jerome Brunet <jbrunet(a)baylibre.com> +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -145,8 +146,8 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, /* Enable irq if necessary */ irq_en = runtime->no_period_wakeup ? 0 : FIFO_INT_COUNT_REPEAT; regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), - CTRL0_INT_EN(irq_en)); + CTRL0_INT_EN, + FIELD_PREP(CTRL0_INT_EN, irq_en)); return 0; } @@ -176,9 +177,9 @@ int axg_fifo_pcm_hw_free(struct snd_soc_component *component, { struct axg_fifo *fifo = axg_fifo_data(ss); - /* Disable the block count irq */ + /* Disable irqs */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), 0); + CTRL0_INT_EN, 0); return 0; } @@ -187,13 +188,13 @@ EXPORT_SYMBOL_GPL(axg_fifo_pcm_hw_free); static void axg_fifo_ack_irq(struct axg_fifo *fifo, u8 mask) { regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - CTRL1_INT_CLR(mask)); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, mask)); /* Clear must also be cleared */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_INT_CLR(FIFO_INT_MASK), - 0); + CTRL1_INT_CLR, + FIELD_PREP(CTRL1_INT_CLR, 0)); } static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) @@ -204,7 +205,7 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) regmap_read(fifo->map, FIFO_STATUS1, &status); - status = STATUS1_INT_STS(status) & FIFO_INT_MASK; + status = FIELD_GET(STATUS1_INT_STS, status); if (status & FIFO_INT_COUNT_REPEAT) snd_pcm_period_elapsed(ss); else @@ -254,15 +255,15 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, - CTRL1_STATUS2_SEL_MASK, - CTRL1_STATUS2_SEL(STATUS2_SEL_DDR_READ)); + CTRL1_STATUS2_SEL, + FIELD_PREP(CTRL1_STATUS2_SEL, STATUS2_SEL_DDR_READ)); /* Make sure the dma is initially disabled */ __dma_enable(fifo, false); /* Disable irqs until params are ready */ regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_INT_EN(FIFO_INT_MASK), 0); + CTRL0_INT_EN, 0); /* Clear any pending interrupt */ axg_fifo_ack_irq(fifo, FIFO_INT_MASK); diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h index a14c31eb06d8..4c48c0a08481 100644 --- a/sound/soc/meson/axg-fifo.h +++ b/sound/soc/meson/axg-fifo.h @@ -40,21 +40,19 @@ struct snd_soc_pcm_runtime; #define FIFO_CTRL0 0x00 #define CTRL0_DMA_EN BIT(31) -#define CTRL0_INT_EN(x) ((x) << 16) +#define CTRL0_INT_EN GENMASK(23, 16) #define CTRL0_SEL_MASK GENMASK(2, 0) #define CTRL0_SEL_SHIFT 0 #define FIFO_CTRL1 0x04 -#define CTRL1_INT_CLR(x) ((x) << 0) -#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8) -#define CTRL1_STATUS2_SEL(x) ((x) << 8) +#define CTRL1_INT_CLR GENMASK(7, 0) +#define CTRL1_STATUS2_SEL GENMASK(11, 8) #define STATUS2_SEL_DDR_READ 0 -#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24) -#define CTRL1_FRDDR_DEPTH(x) ((x) << 24) +#define CTRL1_FRDDR_DEPTH GENMASK(31, 24) #define FIFO_START_ADDR 0x08 #define FIFO_FINISH_ADDR 0x0c #define FIFO_INT_ADDR 0x10 #define FIFO_STATUS1 0x14 -#define STATUS1_INT_STS(x) ((x) << 0) +#define STATUS1_INT_STS GENMASK(7, 0) #define FIFO_STATUS2 0x18 #define FIFO_INIT_ADDR 0x24 #define FIFO_CTRL2 0x28 diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index 98140f449eb3..e97d43ae7fd2 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -7,6 +7,7 @@ * This driver implements the frontend playback DAI of AXG and G12A based SoCs */ +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/regmap.h> #include <linux/module.h> @@ -59,8 +60,8 @@ static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, /* Trim the FIFO depth if the period is small to improve latency */ depth = min(period, fifo->depth); val = (depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH, + FIELD_PREP(CTRL1_FRDDR_DEPTH, val)); return 0; } diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index 32ee45cce7f8..e03a6e21c1c6 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -5,6 +5,7 @@ /* This driver implements the frontend capture DAI of AXG based SoCs */ +#include <linux/bitfield.h> #include <linux/clk.h> #include <linux/regmap.h> #include <linux/module.h> @@ -19,12 +20,9 @@ #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) #define CTRL0_TODDR_SYNC_CH BIT(27) -#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) -#define CTRL0_TODDR_TYPE(x) ((x) << 13) -#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) -#define CTRL0_TODDR_MSB_POS(x) ((x) << 8) -#define CTRL0_TODDR_LSB_POS_MASK GENMASK(7, 3) -#define CTRL0_TODDR_LSB_POS(x) ((x) << 3) +#define CTRL0_TODDR_TYPE GENMASK(15, 13) +#define CTRL0_TODDR_MSB_POS GENMASK(12, 8) +#define CTRL0_TODDR_LSB_POS GENMASK(7, 3) #define CTRL1_TODDR_FORCE_FINISH BIT(25) #define CTRL1_SEL_SHIFT 28 @@ -76,12 +74,12 @@ static int axg_toddr_dai_hw_params(struct snd_pcm_substream *substream, width = params_width(params); regmap_update_bits(fifo->map, FIFO_CTRL0, - CTRL0_TODDR_TYPE_MASK | - CTRL0_TODDR_MSB_POS_MASK | - CTRL0_TODDR_LSB_POS_MASK, - CTRL0_TODDR_TYPE(type) | - CTRL0_TODDR_MSB_POS(TODDR_MSB_POS) | - CTRL0_TODDR_LSB_POS(TODDR_MSB_POS - (width - 1))); + CTRL0_TODDR_TYPE | + CTRL0_TODDR_MSB_POS | + CTRL0_TODDR_LSB_POS, + FIELD_PREP(CTRL0_TODDR_TYPE, type) | + FIELD_PREP(CTRL0_TODDR_MSB_POS, TODDR_MSB_POS) | + FIELD_PREP(CTRL0_TODDR_LSB_POS, TODDR_MSB_POS - (width - 1))); return 0; } -- 2.43.0

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Audio dsp recovery using remoteproc
by noman pouigt 27 Feb '24

27 Feb '24

mailbox to dsp_1 is currently modeling platform pcm driver. mailbox to dsp_2 is also doing the same. Platform driver callbacks cause IPC to be sent to dsp's. Lifecycle of two dsp's are managed by separate remoteproc drivers. Single sound card is exposed. Separate watchdog interrupts from the corresponding dsp's are connected to remoteproc to manage crashing of the individual dsp's. How can I restart both the dsp when either of them crashes using the kernel device model? Remoteproc driver currently only restarts the crashed dsp instead of restarting both the dsp. It is needed to bring up the hardware in a consistent state as both the dsp's are connected to a common hardware. I thought of making a virtual parent remoteproc device and then managing individual dsp as a subdevice of that parent device but remoteproc device node is associated with the individual elf file i.e. it can manage only a single dsp. How can I model remoteproc drivers using linux device model so that when either of them crashes both the dsp's get reloaded by the remoteproc framework. MailBox ---- DSP_1 --- | | Linux ------ common_hw -> speaker/mic | | MailBox ---- DSP_2 ---

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[PATCH 00/13] irqchip: Convert to platform remove callback returning void
by Uwe Kleine-König 27 Feb '24

27 Feb '24

this series converts all drivers below drivers/irqchip to use .remove_new(). See commit 5c5a7680e67b ("platform: Provide a remove callback that returns no value") for an extended explanation and the eventual goal. The TL;DR; is to make it harder for driver authors to leak resources. The drivers touched here are all fine though and don't return early in .remove(). So all conversions in this series are trivial. Best regards Uwe Uwe Kleine-König (13): irqchip/imgpdc: Convert to platform remove callback returning void irqchip/imx-intmux: Convert to platform remove callback returning void irqchip/imx-irqsteer: Convert to platform remove callback returning void irqchip/keystone: Convert to platform remove callback returning void irqchip/ls-scfg-msi: Convert to platform remove callback returning void irqchip/madera: Convert to platform remove callback returning void irqchip/mvebu-pic: Convert to platform remove callback returning void irqchip/pruss-intc: Convert to platform remove callback returning void irqchip/renesas-intc-irqpin: Convert to platform remove callback returning void irqchip/renesas-irqc: Convert to platform remove callback returning void irqchip/renesas-rza1: Convert to platform remove callback returning void irqchip/stm32-exti: Convert to platform remove callback returning void irqchip/ts4800: Convert to platform remove callback returning void drivers/irqchip/irq-imgpdc.c | 5 ++--- drivers/irqchip/irq-imx-intmux.c | 6 ++---- drivers/irqchip/irq-imx-irqsteer.c | 6 ++---- drivers/irqchip/irq-keystone.c | 5 ++--- drivers/irqchip/irq-ls-scfg-msi.c | 6 ++---- drivers/irqchip/irq-madera.c | 6 ++---- drivers/irqchip/irq-mvebu-pic.c | 6 ++---- drivers/irqchip/irq-pruss-intc.c | 6 ++---- drivers/irqchip/irq-renesas-intc-irqpin.c | 5 ++--- drivers/irqchip/irq-renesas-irqc.c | 5 ++--- drivers/irqchip/irq-renesas-rza1.c | 5 ++--- drivers/irqchip/irq-stm32-exti.c | 5 ++--- drivers/irqchip/irq-ts4800.c | 6 ++---- 13 files changed, 26 insertions(+), 46 deletions(-) base-commit: 39676dfe52331dba909c617f213fdb21015c8d10 -- 2.42.0

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[PATCH] ASoC: cs35l56: Prevent bad sign extension in cs35l56_read_silicon_uid()
by Richard Fitzgerald 27 Feb '24

27 Feb '24

Cast u8 values to u32 when using them to build a 32-bit unsigned value that is then stored in a u64. This avoids the possibility of a bad sign extension where the u8 is implicitly extended to an int, thus changing it from an unsigned to a signed value. Whether this is a real problem is debatable, but it does no harm to ensure that the u8 are cast to a suitable type for shifting. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration") --- sound/soc/codecs/cs35l56-shared.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 20b6dbd3fbab..f3670bf85a95 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -659,9 +659,10 @@ static int cs35l56_read_silicon_uid(struct cs35l56_base *cs35l56_base, u64 *uid) return ret; } - unique_id = pte.lot[2] | (pte.lot[1] << 8) | (pte.lot[0] << 16); + unique_id = (u32)pte.lot[2] | ((u32)pte.lot[1] << 8) | ((u32)pte.lot[0] << 16); unique_id <<= 32; - unique_id |= pte.x | (pte.y << 8) | (pte.wafer_id << 16) | (pte.dvs << 24); + unique_id |= (u32)pte.x | ((u32)pte.y << 8) | ((u32)pte.wafer_id << 16) | + ((u32)pte.dvs << 24); dev_dbg(cs35l56_base->dev, "UniqueID = %#llx\n", unique_id); -- 2.39.2

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ALC285 reduce HP pop noise
by Kailang 27 Feb '24

27 Feb '24

Hi Takashi, Lenovo Thinkpad platform with ALC285 which had pop noise from Headphone port on reboot state. Attach patch will reduce it. BR, Kailang

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[PATCH 0/9] ALSA: cs35l56: Apply calibration from EFI
by Richard Fitzgerald 27 Feb '24

27 Feb '24

Factory calibration of the speakers stores the calibration information into an EFI variable. This set of patches adds support for applying speaker calibration data from that EFI variable. Patch #1 and #2 are bugfixes that are prerequisites. The HDA patches #7 and #8 depend on the ASoC patches. Patch #9 is a KUnit test for the new cs-amp-lib library. Richard Fitzgerald (8): ASoC: wm_adsp: Fix missing mutex_lock in wm_adsp_write_ctl() ASoC: wm_adsp: Add wm_adsp_start() and wm_adsp_stop() ASoC: cs-amp-lib: Add helpers for factory calibration data ASoC: cs35l56: Add helper functions for amp calibration ASoC: cs35l56: Apply amp calibration from EFI data ALSA: hda: hda_cs_dsp_ctl: Add helper function to write calibration ALSA: hda: cs35l56: Apply amp calibration from EFI data ASoC: cs-amp-lib: Add KUnit test for calibration helpers Simon Trimmer (1): ALSA: hda: hda_cs_dsp_ctl: Only call notify when a control has been added to a card include/sound/cs-amp-lib.h | 66 +++ include/sound/cs35l56.h | 10 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 36 +- sound/pci/hda/hda_cs_dsp_ctl.c | 52 ++- sound/pci/hda/hda_cs_dsp_ctl.h | 4 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs-amp-lib-test.c | 709 +++++++++++++++++++++++++++++ sound/soc/codecs/cs-amp-lib.c | 270 +++++++++++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 83 ++++ sound/soc/codecs/cs35l56.c | 43 +- sound/soc/codecs/wm_adsp.c | 30 +- sound/soc/codecs/wm_adsp.h | 2 + 15 files changed, 1328 insertions(+), 21 deletions(-) create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c -- 2.30.2

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