August 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: pcmtest: Add support for pcm pausing
by Ivan Orlov 24 Aug '23

24 Aug '23

Add pause push/release support to the virtual PCM test driver. Add 'suspend' boolean field to the pcmtst_buf_iter structure, so we can pause the timer without shutting it down. Update the trigger callback handler correspondingly. Extract buffer initialization to the 'reset_buf_iterator' function since it is used in multiple places now. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/pcmtest.c | 49 ++++++++++++++++++++++++++++++++++------- 1 file changed, 41 insertions(+), 8 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 27cbb9d38f08..b59b78a09224 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -108,6 +108,7 @@ struct pcmtst_buf_iter { size_t total_bytes; // Total bytes read/written size_t chan_block; // Bytes in one channel buffer when non-interleaved struct snd_pcm_substream *substream; + bool suspend; // We need to pause timer without shutting it down struct timer_list timer_instance; }; @@ -115,7 +116,8 @@ static struct snd_pcm_hardware snd_pcmtst_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_NONINTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, @@ -346,6 +348,9 @@ static void timer_timeout(struct timer_list *data) v_iter = from_timer(v_iter, data, timer_instance); substream = v_iter->substream; + if (v_iter->suspend) + return; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !v_iter->is_buf_corrupted) check_buf_block(v_iter, substream->runtime); else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -358,7 +363,9 @@ static void timer_timeout(struct timer_list *data) v_iter->period_pos %= v_iter->period_bytes; snd_pcm_period_elapsed(substream); } - mod_timer(&v_iter->timer_instance, jiffies + TIMER_INTERVAL + inject_delay); + + if (!v_iter->suspend) + mod_timer(&v_iter->timer_instance, jiffies + TIMER_INTERVAL + inject_delay); } static int snd_pcmtst_pcm_open(struct snd_pcm_substream *substream) @@ -373,19 +380,15 @@ static int snd_pcmtst_pcm_open(struct snd_pcm_substream *substream) if (!v_iter) return -ENOMEM; + v_iter->substream = substream; runtime->hw = snd_pcmtst_hw; runtime->private_data = v_iter; - v_iter->substream = substream; - v_iter->buf_pos = 0; - v_iter->is_buf_corrupted = false; - v_iter->period_pos = 0; - v_iter->total_bytes = 0; playback_capture_test = 0; ioctl_reset_test = 0; timer_setup(&v_iter->timer_instance, timer_timeout, 0); - mod_timer(&v_iter->timer_instance, jiffies + TIMER_INTERVAL); + return 0; } @@ -400,10 +403,40 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) return 0; } +static inline void reset_buf_iterator(struct pcmtst_buf_iter *v_iter) +{ + v_iter->buf_pos = 0; + v_iter->is_buf_corrupted = false; + v_iter->period_pos = 0; + v_iter->total_bytes = 0; +} + +static inline void start_pcmtest_timer(struct pcmtst_buf_iter *v_iter) +{ + v_iter->suspend = false; + mod_timer(&v_iter->timer_instance, jiffies + TIMER_INTERVAL); +} + static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct pcmtst_buf_iter *v_iter = substream->runtime->private_data; + if (inject_trigger_err) return -EINVAL; + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + reset_buf_iterator(v_iter); + start_pcmtest_timer(v_iter); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + start_pcmtest_timer(v_iter); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + // We can't call timer_shutdown_sync here, as it is forbidden to sleep here + v_iter->suspend = true; + break; + } return 0; } -- 2.34.1

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[PATCH] ALSA: hda/realtek: Add quirk for HP Victus 16-d1xxx to enable mute LED
by SungHwan Jung 24 Aug '23

24 Aug '23

This quirk enables mute LED on HP Victus 16-d1xxx (8A25) laptops, which use ALC245 codec. Signed-off-by: SungHwan Jung <onenowy(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc7b7a407..d8865f467 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4639,6 +4639,22 @@ static void alc236_fixup_hp_mute_led_coefbit2(struct hda_codec *codec, } } +static void alc245_fixup_hp_mute_led_coefbit(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0x0b; + spec->mute_led_coef.mask = 3 << 2; + spec->mute_led_coef.on = 2 << 2; + spec->mute_led_coef.off = 1 << 2; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static int coef_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -7231,6 +7247,7 @@ enum { ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + ALC245_FIXUP_HP_MUTE_LED_COEFBIT, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9309,6 +9326,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC245_FIXUP_HP_MUTE_LED_COEFBIT] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_mute_led_coefbit, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9582,6 +9603,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), -- 2.42.0

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[PATCH v1] ALSA: hda: cs35l41: Override the _DSD for HP Zbook Fury 17 G9 to correct boost type
by Stefan Binding 24 Aug '23

24 Aug '23

CS35L41 HDA driver requires ACPI to contain correct _DSD properties to correctly configure the device. Whilst the HP Zbook Fury 17 G9 contains valid _DSD properties, the boost type has been configured incorrectly in the _DSD for this laptop. We can override these properties to fix the boost type. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda_property.c | 32 ++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 9b5a1d61575e..b62a4e6968e2 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -43,6 +43,37 @@ static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *phy return 0; } +/* + * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. + * We can override the _DSD to correct the boost type here. + * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists + * in the ACPI. + */ +static int hp_vision_acpi_fix(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; + + dev_info(cs35l41->dev, "Adding DSD properties for %s\n", cs35l41->acpi_subsystem_id); + + cs35l41->index = id; + cs35l41->channel_index = 0; + cs35l41->reset_gpio = gpiod_get_index(physdev, NULL, 1, GPIOD_OUT_HIGH); + cs35l41->speaker_id = -ENOENT; + hw_cfg->spk_pos = cs35l41->index ? 1 : 0; // right:left + hw_cfg->gpio1.func = CS35L41_NOT_USED; + hw_cfg->gpio1.valid = true; + hw_cfg->gpio2.func = CS35L41_INTERRUPT; + hw_cfg->gpio2.valid = true; + hw_cfg->bst_type = CS35L41_INT_BOOST; + hw_cfg->bst_ind = 1000; + hw_cfg->bst_ipk = 4500; + hw_cfg->bst_cap = 24; + hw_cfg->valid = true; + + return 0; +} + struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -53,6 +84,7 @@ struct cs35l41_prop_model { static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, + { "CSC3551", "103C89C6", hp_vision_acpi_fix }, {} }; -- 2.34.1

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[PATCH 0/4] ALSA: ump: Improve legacy rawmidi & co
by Takashi Iwai 24 Aug '23

24 Aug '23

Hi, this is a patch set to fix / enhance the legacy rawmidi handling in UMP core and USB-audio driver, as well as the update of MIDI 2.0 documentation. Takashi === Takashi Iwai (4): ALSA: usb-audio: Attach legacy rawmidi after probing all UMP EPs ALSA: ump: Fill group names for legacy rawmidi substreams ALSA: ump: Don't create unused substreams for static blocks ALSA: documentation: Add description for USB MIDI 2.0 gadget driver Documentation/sound/designs/midi-2.0.rst | 188 +++++++++++++++++++++++ include/sound/ump.h | 1 + sound/core/ump.c | 58 ++++++- sound/usb/midi2.c | 15 +- 4 files changed, 250 insertions(+), 12 deletions(-) -- 2.35.3

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[GIT PULL] ASoC fixes for v6.5-rc7
by Mark Brown 24 Aug '23

24 Aug '23

The following changes since commit 37aba3190891d4de189bd5192ee95220e295f34d: ASoC: rt1308-sdw: fix random louder sound (2023-08-13 18:16:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc7 for you to fetch changes up to 1613781d7e8a93618ff3a6b37f81f06769b53717: ASoC: cs35l41: Correct amp_gain_tlv values (2023-08-23 13:27:06 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A relatively large but generally not super urgent set of fixes for ASoC, including some quirks and a MAINTAINERS update. There's also an update to cs35l56 to change the firmware ABI, there are no current shipping systems which use the current interface and the sooner we get the new interface in the less likely it is that something will start. It'd be nice if these landed for v6.5 but not the end of the world if they wait till v6.6. ---------------------------------------------------------------- BrenoRCBrito (1): ASoC: amd: yc: Add VivoBook Pro 15 to quirks list for acp6x Chao Song (1): ASoC: SOF: ipc4-pcm: fix possible null pointer deference Charles Keepax (1): ASoC: cs35l41: Correct amp_gain_tlv values Kevin-Lu (1): MAINTAINERS: Add entries for TEXAS INSTRUMENTS ASoC DRIVERS Maciej Strozek (1): ASoC: cs35l56: Read firmware uuid from a device property instead of _SUB Mark Brown (1): ASoC: cs35l56: Update ACPI HID and property Shenghao Ding (1): ASoC: tas2781: fixed register access error when switching to other chips Simon Trimmer (1): ASoC: cs35l56: Add an ACPI match table MAINTAINERS | 33 +++++++++++++++++++++++++++++++++ sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs35l41.c | 2 +- sound/soc/codecs/cs35l56-i2c.c | 9 +++++++++ sound/soc/codecs/cs35l56-spi.c | 9 +++++++++ sound/soc/codecs/cs35l56.c | 31 ++++++++++++------------------- sound/soc/codecs/tas2781-comlib.c | 19 ++++++++++--------- sound/soc/sof/ipc4-pcm.c | 3 +++ 8 files changed, 84 insertions(+), 29 deletions(-)

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[PATCH v1 0/5] ASoC: sunxi: Add support for D1/T113s internal audio codec
by Maksim Kiselev 24 Aug '23

24 Aug '23

Hi, This is the series adding support for internal audio codec on Allwinner D1/T113s SoCs family. My work based on Icenowy Zheng's patches from sipeed repo that adds support for the R329 audio codec. But, unfortunately, these patches were not sent to upstream. So, I adapted them to fit the T113s. In the future, support for R329 could easily be added on top of this series. Because D1/T113s and R329 SoCs have almost the same audio codec IP, with slight differences in the analog part, plus the R329 has two additional ADCs. The series includes modifications for the existing sun4i-codec driver that brings support for the digital part of the D1 codec. And adds the new sun20i-d1-codec-analog driver for the analog part. I would be glad if someone have a look :) Cheers, Maksim Maksim Kiselev (5): ASoC: dt-bindings: sun4i-a10-codec: Add binding for Allwinner D1 SoC ASoC: dt-bindings: Add schema for "allwinner,sun20i-d1-codec-analog" ASoC: sunxi: sun4i-codec: add basic support for D1 audio codec ASoC: sunxi: Add new driver for Allwinner D1/T113s codec's analog path controls riscv: dts: allwinner: d1: Add device nodes for internal audio codec .../allwinner,sun20i-d1-codec-analog.yaml | 33 ++ .../sound/allwinner,sun4i-a10-codec.yaml | 64 ++- .../boot/dts/allwinner/sunxi-d1s-t113.dtsi | 22 ++ sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun20i-d1-codec-analog.c | 220 +++++++++++ sound/soc/sunxi/sun4i-codec.c | 364 +++++++++++++++--- 7 files changed, 643 insertions(+), 72 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun20i-d1-codec-analog.yaml create mode 100644 sound/soc/sunxi/sun20i-d1-codec-analog.c -- 2.39.2

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[PATCH] ASoC: cs42l43: Initialize ret in default case in cs42l43_pll_ev()
by Nathan Chancellor 24 Aug '23

24 Aug '23

clang warns (or errors with CONFIG_WERROR=y): sound/soc/codecs/cs42l43.c:1371:2: error: variable 'ret' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] 1371 | default: | ^~~~~~~ sound/soc/codecs/cs42l43.c:1377:9: note: uninitialized use occurs here 1377 | return ret; | ^~~ sound/soc/codecs/cs42l43.c:1349:9: note: initialize the variable 'ret' to silence this warning 1349 | int ret; | ^ | = 0 1 error generated. Initialize ret to 0 in the default case, as there was nothing to do for other event types. Closes: https://github.com/ClangBuiltLinux/linux/issues/1922 Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/cs42l43.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 55a79219af35..18178ffa82f2 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -1369,6 +1369,7 @@ static int cs42l43_pll_ev(struct snd_soc_dapm_widget *w, clk_disable_unprepare(priv->mclk); break; default: + ret = 0; break; } --- base-commit: 014ee0692f29da8b08fed5da0fa14e04698a50f7 change-id: 20230823-cs42l43_pll_ev-init-ret-fb1e47ea0605 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

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[PATCH] ASoC: audio-graph-card2: add comment for format property
by Kuninori Morimoto 23 Aug '23

23 Aug '23

We don't need to have "format" property on DT any more if CPU/Codec driver has .auto_selectable_formats settings on snd_soc_dai_ops. The sample dtsi doesn't have it. To avoid user confusion, this patch indicates it on comment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../soc/generic/audio-graph-card2-custom-sample.dtsi | 7 +++++++ sound/soc/generic/audio-graph-card2.c | 12 ++++++++++++ 2 files changed, 19 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 2ac0de3c21da..8acaa2ddb335 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -34,6 +34,13 @@ * ... * }; * + * + * Below sample doesn't use "format" property, + * because test-component driver (test-cpu/test-codec) is supporting + * snd_soc_dai_ops :: .auto_selectable_formats. + * see + * snd_soc_runtime_get_dai_fmt() + * linux/sound/soc/generic/test-component.c :: test_dai_formats */ / { /* diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 542c4a114940..1fae810a2833 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -47,6 +47,18 @@ see graph_parse_daifmt(). + "format" property is no longer needed on DT if both CPU/Codec drivers are + supporting snd_soc_dai_ops :: .auto_selectable_formats. + see + snd_soc_runtime_get_dai_fmt() + + sample driver + linux/sound/soc/sh/rcar/core.c + linux/sound/soc/codecs/ak4613.c + linux/sound/soc/codecs/pcm3168a.c + linux/sound/soc/soc-utils.c + linux/sound/soc/generic/test-component.c + ************************************ Normal Audio-Graph ************************************ -- 2.25.1

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[PATCH 0/5] ASoC: rsnd: tidyup ADG
by Kuninori Morimoto 23 Aug '23

23 Aug '23

Hi Mark ------ for git-log ------ 8< ------ 8< ------ 8< ------ Renesas Sound has ADG for clock control. Basically it needs accurately divisible external input clock. But sometimes sometimes it doesn't have to be accurate for some reason. We can use ADG clk_i for such case. It came from CPG as very high rate clock, but is not accurately divisible for 48kHz/44.1kHz rate, but enough for approximate rate. This patch set support such use case. ------ extra info ------ 8< ------ 8< ------ 8< ------ 8< ------ Kuninori Morimoto (5): ASoC: rsnd: enable clk_i approximate rate usage ASoC: rsnd: setup clock-out only when all conditions are right ASoC: rsnd: tidyup brga/brgb default value ASoC: rsnd: remove default division of clock out ASoC: rsnd: setup BRGCKR/BRRA/BRRB on rsnd_adg_clk_control() sound/soc/sh/rcar/adg.c | 68 +++++++++++++++++++++++++++++++++-------- 1 file changed, 55 insertions(+), 13 deletions(-) -- 2.25.1

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[PATCH] ALSA: ac97: Fix possible NULL dereference in snd_ac97_mixer
by Su Hui 23 Aug '23

23 Aug '23

smatch error: sound/pci/ac97/ac97_codec.c:2354 snd_ac97_mixer() error: we previously assumed 'rac97' could be null (see line 2072) remove redundant assignment, return error if rac97 is NULL. Fixes: da3cec35dd3c ("ALSA: Kill snd_assert() in sound/pci/*") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 9afc5906d662..80a65b8ad7b9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2069,8 +2069,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, .dev_disconnect = snd_ac97_dev_disconnect, }; - if (rac97) - *rac97 = NULL; + if (!rac97) + return -EINVAL; if (snd_BUG_ON(!bus || !template)) return -EINVAL; if (snd_BUG_ON(template->num >= 4)) -- 2.30.2

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[PATCH 1/7] ASoC: SOF: amd: remove unused sha dma interrupt code
by Vijendar Mukunda 23 Aug '23

23 Aug '23

During initial development time for RN platform, when SHA dma gets completed, SHA DMA engine used to raise the ACP interrupt. In ACP interrupt handler, SHA DMA interrupt got handled. Currently SHA DMA compleition is verified by checking transfer count using read poll time out logic. Remove unused SHA dma interrupt handling code. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/sof/amd/acp.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index b2e00a10a03e..630c2c5fe4c7 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -337,14 +337,7 @@ static irqreturn_t acp_irq_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); - unsigned int val, count = ACP_HW_SEM_RETRY_COUNT; - - val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->ext_intr_stat); - if (val & ACP_SHA_STAT) { - /* Clear SHA interrupt raised by PSP */ - snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, val); - return IRQ_HANDLED; - } + unsigned int count = ACP_HW_SEM_RETRY_COUNT; while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { /* Wait until acquired HW Semaphore lock or timeout */ -- 2.34.1

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[PATCH 1/2] ASoC: dt-bindings: nau8821: Add single-ended input feature
by Seven Lee 23 Aug '23

23 Aug '23

Add input with single-ended control. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index fc2f4ce4db88..3e54abd4ca74 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -89,6 +89,12 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32 default: 3072000 + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + '#sound-dai-cells': const: 0 @@ -114,6 +120,7 @@ examples: nuvoton,jkdet-pull-enable; nuvoton,jkdet-pull-up; nuvoton,key-enable; + nuvoton,left-input-single-end; nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; nuvoton,micbias-voltage = <6>; nuvoton,vref-impedance = <2>; -- 2.25.1

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[PATCH] ASoC: cs35l41: Correct amp_gain_tlv values
by Charles Keepax 23 Aug '23

23 Aug '23

The current analog gain TLV seems to have completely incorrect values in it. The gain starts at 0.5dB, proceeds in 1dB steps, and has no mute value, correct the control to match. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 6ac501f008eca..8a879b6f48290 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -168,7 +168,7 @@ static int cs35l41_get_fs_mon_config_index(int freq) static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), 1, 913, TLV_DB_MINMAX_ITEM(-10200, 1200)); -static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); +static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 50, 100, 0); static const struct snd_kcontrol_new dre_ctrl = SOC_DAPM_SINGLE("Switch", CS35L41_PWR_CTRL3, 20, 1, 0); -- 2.30.2

3 2

[PATCH v2] ASoC: Delete UDA134x/L3 audio codec
by Linus Walleij 23 Aug '23

23 Aug '23

This codec was used by the deleted S3C board sound/soc/samsung/s3c24xx_uda134x.c. Fixes: 503278c12701 ("ASoC: samsung: remove unused drivers") Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Properly delete the L3 codec Kconfig and Makefile entries :/ - Test with x86_64 allmodconfig - Link to v1: https://lore.kernel.org/r/20230821-delete-l3-v1-1-26d9cd32e7a2@linaro.org --- include/sound/l3.h | 28 --- include/sound/uda134x.h | 24 -- sound/soc/codecs/Kconfig | 8 - sound/soc/codecs/Makefile | 4 - sound/soc/codecs/l3.c | 132 ---------- sound/soc/codecs/uda134x.c | 587 --------------------------------------------- sound/soc/codecs/uda134x.h | 33 --- 7 files changed, 816 deletions(-) diff --git a/include/sound/l3.h b/include/sound/l3.h deleted file mode 100644 index b6f58072237a..000000000000 --- a/include/sound/l3.h +++ /dev/null @@ -1,28 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef _L3_H_ -#define _L3_H_ 1 - -struct l3_pins { - void (*setdat)(struct l3_pins *, int); - void (*setclk)(struct l3_pins *, int); - void (*setmode)(struct l3_pins *, int); - - int gpio_data; - int gpio_clk; - int gpio_mode; - int use_gpios; - - int data_hold; - int data_setup; - int clock_high; - int mode_hold; - int mode; - int mode_setup; -}; - -struct device; - -int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); -int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap); - -#endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h deleted file mode 100644 index db82516da162..000000000000 --- a/include/sound/uda134x.h +++ /dev/null @@ -1,24 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * uda134x.h -- UDA134x ALSA SoC Codec driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - */ - -#ifndef _UDA134X_H -#define _UDA134X_H - -#include <sound/l3.h> - -struct uda134x_platform_data { - struct l3_pins l3; - void (*power) (int); - int model; -#define UDA134X_UDA1340 1 -#define UDA134X_UDA1341 2 -#define UDA134X_UDA1344 3 -#define UDA134X_UDA1345 4 -}; - -#endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ef279a27eb51..8964616275df 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,7 +15,6 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST imply SND_SOC_88PM860X - imply SND_SOC_L3 imply SND_SOC_AB8500_CODEC imply SND_SOC_AC97_CODEC imply SND_SOC_AD1836 @@ -268,7 +267,6 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TWL4030 imply SND_SOC_TWL6040 imply SND_SOC_UDA1334 - imply SND_SOC_UDA134X imply SND_SOC_UDA1380 imply SND_SOC_WCD9335 imply SND_SOC_WCD934X @@ -994,9 +992,6 @@ config SND_SOC_JZ4770_CODEC This driver can also be built as a module. If so, the module will be called snd-soc-jz4770-codec. -config SND_SOC_L3 - tristate - config SND_SOC_DA7210 tristate depends on SND_SOC_I2C_AND_SPI @@ -1953,9 +1948,6 @@ config SND_SOC_UDA1334 and has basic features such as de-emphasis (at 44.1 kHz sampling rate) and mute. -config SND_SOC_UDA134X - tristate - config SND_SOC_UDA1380 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2bdc0ef29a9e..e89698e239dd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,7 +129,6 @@ snd-soc-jz4740-codec-objs := jz4740.o snd-soc-jz4725b-codec-objs := jz4725b.o snd-soc-jz4760-codec-objs := jz4760.o snd-soc-jz4770-codec-objs := jz4770.o -snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-lochnagar-sc-objs := lochnagar-sc.o @@ -301,7 +300,6 @@ snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda1334-objs := uda1334.o -snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o @@ -514,7 +512,6 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_JZ4725B_CODEC) += snd-soc-jz4725b-codec.o obj-$(CONFIG_SND_SOC_JZ4760_CODEC) += snd-soc-jz4760-codec.o obj-$(CONFIG_SND_SOC_JZ4770_CODEC) += snd-soc-jz4770-codec.o -obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_LOCHNAGAR_SC) += snd-soc-lochnagar-sc.o @@ -683,7 +680,6 @@ obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA1334) += snd-soc-uda1334.o -obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c deleted file mode 100644 index b84f6f1f6800..000000000000 --- a/sound/soc/codecs/l3.c +++ /dev/null @@ -1,132 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * L3 code - * - * Copyright (C) 2008, Christian Pellegrin <chripell(a)evolware.org> - * - * based on: - * - * L3 bus algorithm module. - * - * Copyright (C) 2001 Russell King, All Rights Reserved. - */ - -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/delay.h> -#include <linux/device.h> -#include <linux/gpio.h> - -#include <sound/l3.h> - -/* - * Send one byte of data to the chip. Data is latched into the chip on - * the rising edge of the clock. - */ -static void sendbyte(struct l3_pins *adap, unsigned int byte) -{ - int i; - - for (i = 0; i < 8; i++) { - adap->setclk(adap, 0); - udelay(adap->data_hold); - adap->setdat(adap, byte & 1); - udelay(adap->data_setup); - adap->setclk(adap, 1); - udelay(adap->clock_high); - byte >>= 1; - } -} - -/* - * Send a set of bytes to the chip. We need to pulse the MODE line - * between each byte, but never at the start nor at the end of the - * transfer. - */ -static void sendbytes(struct l3_pins *adap, const u8 *buf, - int len) -{ - int i; - - for (i = 0; i < len; i++) { - if (i) { - udelay(adap->mode_hold); - adap->setmode(adap, 0); - udelay(adap->mode); - } - adap->setmode(adap, 1); - udelay(adap->mode_setup); - sendbyte(adap, buf[i]); - } -} - -int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) -{ - adap->setclk(adap, 1); - adap->setdat(adap, 1); - adap->setmode(adap, 1); - udelay(adap->mode); - - adap->setmode(adap, 0); - udelay(adap->mode_setup); - sendbyte(adap, addr); - udelay(adap->mode_hold); - - sendbytes(adap, data, len); - - adap->setclk(adap, 1); - adap->setdat(adap, 1); - adap->setmode(adap, 0); - - return len; -} -EXPORT_SYMBOL_GPL(l3_write); - - -static void l3_set_clk(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_clk, val); -} - -static void l3_set_data(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_data, val); -} - -static void l3_set_mode(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_mode, val); -} - -int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap) -{ - int ret; - - if (!adap->use_gpios) - return -EINVAL; - - ret = devm_gpio_request_one(dev, adap->gpio_data, - GPIOF_OUT_INIT_LOW, "l3_data"); - if (ret < 0) - return ret; - adap->setdat = l3_set_data; - - ret = devm_gpio_request_one(dev, adap->gpio_clk, - GPIOF_OUT_INIT_LOW, "l3_clk"); - if (ret < 0) - return ret; - adap->setclk = l3_set_clk; - - ret = devm_gpio_request_one(dev, adap->gpio_mode, - GPIOF_OUT_INIT_LOW, "l3_mode"); - if (ret < 0) - return ret; - adap->setmode = l3_set_mode; - - return 0; -} -EXPORT_SYMBOL_GPL(l3_set_gpio_ops); - -MODULE_DESCRIPTION("L3 bit-banging driver"); -MODULE_AUTHOR("Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c deleted file mode 100644 index 1a62bec94005..000000000000 --- a/sound/soc/codecs/uda134x.c +++ /dev/null @@ -1,587 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * uda134x.c -- UDA134X ALSA SoC Codec driver - * - * Modifications by Christian Pellegrin <chripell(a)evolware.org> - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie - */ - -#include <linux/module.h> -#include <linux/delay.h> -#include <linux/slab.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/initval.h> - -#include <sound/uda134x.h> -#include <sound/l3.h> - -#include "uda134x.h" - - -#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 -#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) - -struct uda134x_priv { - int sysclk; - int dai_fmt; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; - - struct regmap *regmap; - struct uda134x_platform_data *pd; -}; - -static const struct reg_default uda134x_reg_defaults[] = { - { UDA134X_EA000, 0x04 }, - { UDA134X_EA001, 0x04 }, - { UDA134X_EA010, 0x04 }, - { UDA134X_EA011, 0x00 }, - { UDA134X_EA100, 0x00 }, - { UDA134X_EA101, 0x00 }, - { UDA134X_EA110, 0x00 }, - { UDA134X_EA111, 0x00 }, - { UDA134X_STATUS0, 0x00 }, - { UDA134X_STATUS1, 0x03 }, - { UDA134X_DATA000, 0x00 }, - { UDA134X_DATA001, 0x00 }, - { UDA134X_DATA010, 0x00 }, - { UDA134X_DATA011, 0x00 }, - { UDA134X_DATA1, 0x00 }, -}; - -/* - * Write to the uda134x registers - * - */ -static int uda134x_regmap_write(void *context, unsigned int reg, - unsigned int value) -{ - struct uda134x_platform_data *pd = context; - int ret; - u8 addr; - u8 data = value; - - switch (reg) { - case UDA134X_STATUS0: - case UDA134X_STATUS1: - addr = UDA134X_STATUS_ADDR; - data |= (reg - UDA134X_STATUS0) << 7; - break; - case UDA134X_DATA000: - case UDA134X_DATA001: - case UDA134X_DATA010: - case UDA134X_DATA011: - addr = UDA134X_DATA0_ADDR; - data |= (reg - UDA134X_DATA000) << 6; - break; - case UDA134X_DATA1: - addr = UDA134X_DATA1_ADDR; - break; - default: - /* It's an extended address register */ - addr = (reg | UDA134X_EXTADDR_PREFIX); - - ret = l3_write(&pd->l3, - UDA134X_DATA0_ADDR, &addr, 1); - if (ret != 1) - return -EIO; - - addr = UDA134X_DATA0_ADDR; - data = (value | UDA134X_EXTDATA_PREFIX); - break; - } - - ret = l3_write(&pd->l3, - addr, &data, 1); - if (ret != 1) - return -EIO; - - return 0; -} - -static inline void uda134x_reset(struct snd_soc_component *component) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - unsigned int mask = 1<<6; - - regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, mask); - msleep(1); - regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0); -} - -static int uda134x_mute(struct snd_soc_dai *dai, int mute, int direction) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component); - unsigned int mask = 1<<2; - unsigned int val; - - pr_debug("%s mute: %d\n", __func__, mute); - - if (mute) - val = mask; - else - val = 0; - - return regmap_update_bits(uda134x->regmap, UDA134X_DATA010, mask, val); -} - -static int uda134x_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct snd_pcm_runtime *master_runtime; - - if (uda134x->master_substream) { - master_runtime = uda134x->master_substream->runtime; - - pr_debug("%s constraining to %d bits at %d\n", __func__, - master_runtime->sample_bits, - master_runtime->rate); - - snd_pcm_hw_constraint_single(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate); - - snd_pcm_hw_constraint_single(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits); - - uda134x->slave_substream = substream; - } else - uda134x->master_substream = substream; - - return 0; -} - -static void uda134x_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - if (uda134x->master_substream == substream) - uda134x->master_substream = uda134x->slave_substream; - - uda134x->slave_substream = NULL; -} - -static int uda134x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - unsigned int hw_params = 0; - - if (substream == uda134x->slave_substream) { - pr_debug("%s ignoring hw_params for slave substream\n", - __func__); - return 0; - } - - pr_debug("%s sysclk: %d, rate:%d\n", __func__, - uda134x->sysclk, params_rate(params)); - - /* set SYSCLK / fs ratio */ - switch (uda134x->sysclk / params_rate(params)) { - case 512: - break; - case 384: - hw_params |= (1<<4); - break; - case 256: - hw_params |= (1<<5); - break; - default: - printk(KERN_ERR "%s unsupported fs\n", __func__); - return -EINVAL; - } - - pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, - uda134x->dai_fmt, params_format(params)); - - /* set DAI format and word length */ - switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - break; - case SND_SOC_DAIFMT_RIGHT_J: - switch (params_width(params)) { - case 16: - hw_params |= (1<<1); - break; - case 18: - hw_params |= (1<<2); - break; - case 20: - hw_params |= ((1<<2) | (1<<1)); - break; - default: - printk(KERN_ERR "%s unsupported format (right)\n", - __func__); - return -EINVAL; - } - break; - case SND_SOC_DAIFMT_LEFT_J: - hw_params |= (1<<3); - break; - default: - printk(KERN_ERR "%s unsupported format\n", __func__); - return -EINVAL; - } - - return regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, - STATUS0_SYSCLK_MASK | STATUS0_DAIFMT_MASK, hw_params); -} - -static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_component *component = codec_dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, - clk_id, freq, dir); - - /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable - because the codec is slave. Of course limitations of the clock - master (the IIS controller) apply. - We'll error out on set_hw_params if it's not OK */ - if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { - uda134x->sysclk = freq; - return 0; - } - - printk(KERN_ERR "%s unsupported sysclk\n", __func__); - return -EINVAL; -} - -static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) -{ - struct snd_soc_component *component = codec_dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - pr_debug("%s fmt: %08X\n", __func__, fmt); - - /* codec supports only full consumer mode */ - if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { - printk(KERN_ERR "%s unsupported clocking mode\n", __func__); - return -EINVAL; - } - - /* no support for clock inversion */ - if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { - printk(KERN_ERR "%s unsupported clock inversion\n", __func__); - return -EINVAL; - } - - /* We can't setup DAI format here as it depends on the word bit num */ - /* so let's just store the value for later */ - uda134x->dai_fmt = fmt; - - return 0; -} - -static int uda134x_set_bias_level(struct snd_soc_component *component, - enum snd_soc_bias_level level) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct uda134x_platform_data *pd = uda134x->pd; - pr_debug("%s bias level %d\n", __func__, level); - - switch (level) { - case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - /* power on */ - if (pd->power) { - pd->power(1); - regcache_sync(uda134x->regmap); - } - break; - case SND_SOC_BIAS_STANDBY: - break; - case SND_SOC_BIAS_OFF: - /* power off */ - if (pd->power) { - pd->power(0); - regcache_mark_dirty(uda134x->regmap); - } - break; - } - return 0; -} - -static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", - "Minimum2", "Maximum"}; -static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const char *uda134x_mixmode[] = {"Differential", "Analog1", - "Analog2", "Both"}; - -static const struct soc_enum uda134x_mixer_enum[] = { -SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), -SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), -SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), -}; - -static const struct snd_kcontrol_new uda1341_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), -SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), -SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), -SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), - -SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), -SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), - -SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), -SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), - -SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), -SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), - -SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), -SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), -SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), - -SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), -SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), -SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), -SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), -SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -static const struct snd_kcontrol_new uda1340_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), - -SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), -SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), - -SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), - -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -static const struct snd_kcontrol_new uda1345_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), - -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), - -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -/* UDA1341 has the DAC/ADC power down in STATUS1 */ -static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), -}; - -/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ -static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), -}; - -/* Common DAPM widgets */ -static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("VINL1"), - SND_SOC_DAPM_INPUT("VINR1"), - SND_SOC_DAPM_INPUT("VINL2"), - SND_SOC_DAPM_INPUT("VINR2"), - SND_SOC_DAPM_OUTPUT("VOUTL"), - SND_SOC_DAPM_OUTPUT("VOUTR"), -}; - -static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { - { "ADC", NULL, "VINL1" }, - { "ADC", NULL, "VINR1" }, - { "ADC", NULL, "VINL2" }, - { "ADC", NULL, "VINR2" }, - { "VOUTL", NULL, "DAC" }, - { "VOUTR", NULL, "DAC" }, -}; - -static const struct snd_soc_dai_ops uda134x_dai_ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .mute_stream = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - .no_capture_mute = 1, -}; - -static struct snd_soc_dai_driver uda134x_dai = { - .name = "uda134x-hifi", - /* playback capabilities */ - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = UDA134X_RATES, - .formats = UDA134X_FORMATS, - }, - /* capture capabilities */ - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = UDA134X_RATES, - .formats = UDA134X_FORMATS, - }, - /* pcm operations */ - .ops = &uda134x_dai_ops, -}; - -static int uda134x_soc_probe(struct snd_soc_component *component) -{ - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct uda134x_platform_data *pd = uda134x->pd; - const struct snd_soc_dapm_widget *widgets; - unsigned int num_widgets; - int ret; - - printk(KERN_INFO "UDA134X SoC Audio Codec\n"); - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1341: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", - pd->model); - return -EINVAL; - } - - if (pd->power) - pd->power(1); - - uda134x_reset(component); - - if (pd->model == UDA134X_UDA1341) { - widgets = uda1341_dapm_widgets; - num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); - } else { - widgets = uda1340_dapm_widgets; - num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); - } - - ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets); - if (ret) { - printk(KERN_ERR "%s failed to register dapm controls: %d", - __func__, ret); - return ret; - } - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - ret = snd_soc_add_component_controls(component, uda1340_snd_controls, - ARRAY_SIZE(uda1340_snd_controls)); - break; - case UDA134X_UDA1341: - ret = snd_soc_add_component_controls(component, uda1341_snd_controls, - ARRAY_SIZE(uda1341_snd_controls)); - break; - case UDA134X_UDA1345: - ret = snd_soc_add_component_controls(component, uda1345_snd_controls, - ARRAY_SIZE(uda1345_snd_controls)); - break; - default: - printk(KERN_ERR "%s unknown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - if (ret < 0) { - printk(KERN_ERR "UDA134X: failed to register controls\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_component_driver soc_component_dev_uda134x = { - .probe = uda134x_soc_probe, - .set_bias_level = uda134x_set_bias_level, - .dapm_widgets = uda134x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), - .dapm_routes = uda134x_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), - .suspend_bias_off = 1, - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, -}; - -static const struct regmap_config uda134x_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - .max_register = UDA134X_DATA1, - .reg_defaults = uda134x_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(uda134x_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .reg_write = uda134x_regmap_write, -}; - -static int uda134x_codec_probe(struct platform_device *pdev) -{ - struct uda134x_platform_data *pd = pdev->dev.platform_data; - struct uda134x_priv *uda134x; - int ret; - - if (!pd) { - dev_err(&pdev->dev, "Missing L3 bitbang function\n"); - return -ENODEV; - } - - uda134x = devm_kzalloc(&pdev->dev, sizeof(*uda134x), GFP_KERNEL); - if (!uda134x) - return -ENOMEM; - - uda134x->pd = pd; - platform_set_drvdata(pdev, uda134x); - - if (pd->l3.use_gpios) { - ret = l3_set_gpio_ops(&pdev->dev, &uda134x->pd->l3); - if (ret < 0) - return ret; - } - - uda134x->regmap = devm_regmap_init(&pdev->dev, NULL, pd, - &uda134x_regmap_config); - if (IS_ERR(uda134x->regmap)) - return PTR_ERR(uda134x->regmap); - - return devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_uda134x, &uda134x_dai, 1); -} - -static struct platform_driver uda134x_codec_driver = { - .driver = { - .name = "uda134x-codec", - }, - .probe = uda134x_codec_probe, -}; - -module_platform_driver(uda134x_codec_driver); - -MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h deleted file mode 100644 index 664618c2571c..000000000000 --- a/sound/soc/codecs/uda134x.h +++ /dev/null @@ -1,33 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef _UDA134X_CODEC_H -#define _UDA134X_CODEC_H - -#define UDA134X_L3ADDR 5 -#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) -#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) -#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) - -#define UDA134X_EXTADDR_PREFIX 0xC0 -#define UDA134X_EXTDATA_PREFIX 0xE0 - -/* UDA134X registers */ -#define UDA134X_EA000 0 -#define UDA134X_EA001 1 -#define UDA134X_EA010 2 -#define UDA134X_EA011 3 -#define UDA134X_EA100 4 -#define UDA134X_EA101 5 -#define UDA134X_EA110 6 -#define UDA134X_EA111 7 -#define UDA134X_STATUS0 8 -#define UDA134X_STATUS1 9 -#define UDA134X_DATA000 10 -#define UDA134X_DATA001 11 -#define UDA134X_DATA010 12 -#define UDA134X_DATA011 13 -#define UDA134X_DATA1 14 - -#define STATUS0_DAIFMT_MASK (~(7<<1)) -#define STATUS0_SYSCLK_MASK (~(3<<4)) - -#endif --- base-commit: 17b9f4387ebabb19b871bbe2d06562e48e4e7130 change-id: 20230821-delete-l3-573f483201e3 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 1

[PATCH] ASoC: Delete UDA134x/L3 audio codec
by Linus Walleij 23 Aug '23

23 Aug '23

This codec was used by the deleted S3C board sound/soc/samsung/s3c24xx_uda134x.c. Fixes: 503278c12701 ("ASoC: samsung: remove unused drivers") Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- include/sound/l3.h | 28 --- include/sound/uda134x.h | 24 -- sound/soc/codecs/Kconfig | 4 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/l3.c | 132 ---------- sound/soc/codecs/uda134x.c | 587 --------------------------------------------- sound/soc/codecs/uda134x.h | 33 --- 7 files changed, 810 deletions(-) diff --git a/include/sound/l3.h b/include/sound/l3.h deleted file mode 100644 index b6f58072237a..000000000000 --- a/include/sound/l3.h +++ /dev/null @@ -1,28 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef _L3_H_ -#define _L3_H_ 1 - -struct l3_pins { - void (*setdat)(struct l3_pins *, int); - void (*setclk)(struct l3_pins *, int); - void (*setmode)(struct l3_pins *, int); - - int gpio_data; - int gpio_clk; - int gpio_mode; - int use_gpios; - - int data_hold; - int data_setup; - int clock_high; - int mode_hold; - int mode; - int mode_setup; -}; - -struct device; - -int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); -int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap); - -#endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h deleted file mode 100644 index db82516da162..000000000000 --- a/include/sound/uda134x.h +++ /dev/null @@ -1,24 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * uda134x.h -- UDA134x ALSA SoC Codec driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - */ - -#ifndef _UDA134X_H -#define _UDA134X_H - -#include <sound/l3.h> - -struct uda134x_platform_data { - struct l3_pins l3; - void (*power) (int); - int model; -#define UDA134X_UDA1340 1 -#define UDA134X_UDA1341 2 -#define UDA134X_UDA1344 3 -#define UDA134X_UDA1345 4 -}; - -#endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ef279a27eb51..520b4e13e08b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -268,7 +268,6 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TWL4030 imply SND_SOC_TWL6040 imply SND_SOC_UDA1334 - imply SND_SOC_UDA134X imply SND_SOC_UDA1380 imply SND_SOC_WCD9335 imply SND_SOC_WCD934X @@ -1953,9 +1952,6 @@ config SND_SOC_UDA1334 and has basic features such as de-emphasis (at 44.1 kHz sampling rate) and mute. -config SND_SOC_UDA134X - tristate - config SND_SOC_UDA1380 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2bdc0ef29a9e..34ebbb27f96a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -301,7 +301,6 @@ snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda1334-objs := uda1334.o -snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o @@ -683,7 +682,6 @@ obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA1334) += snd-soc-uda1334.o -obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c deleted file mode 100644 index b84f6f1f6800..000000000000 --- a/sound/soc/codecs/l3.c +++ /dev/null @@ -1,132 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * L3 code - * - * Copyright (C) 2008, Christian Pellegrin <chripell(a)evolware.org> - * - * based on: - * - * L3 bus algorithm module. - * - * Copyright (C) 2001 Russell King, All Rights Reserved. - */ - -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/delay.h> -#include <linux/device.h> -#include <linux/gpio.h> - -#include <sound/l3.h> - -/* - * Send one byte of data to the chip. Data is latched into the chip on - * the rising edge of the clock. - */ -static void sendbyte(struct l3_pins *adap, unsigned int byte) -{ - int i; - - for (i = 0; i < 8; i++) { - adap->setclk(adap, 0); - udelay(adap->data_hold); - adap->setdat(adap, byte & 1); - udelay(adap->data_setup); - adap->setclk(adap, 1); - udelay(adap->clock_high); - byte >>= 1; - } -} - -/* - * Send a set of bytes to the chip. We need to pulse the MODE line - * between each byte, but never at the start nor at the end of the - * transfer. - */ -static void sendbytes(struct l3_pins *adap, const u8 *buf, - int len) -{ - int i; - - for (i = 0; i < len; i++) { - if (i) { - udelay(adap->mode_hold); - adap->setmode(adap, 0); - udelay(adap->mode); - } - adap->setmode(adap, 1); - udelay(adap->mode_setup); - sendbyte(adap, buf[i]); - } -} - -int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) -{ - adap->setclk(adap, 1); - adap->setdat(adap, 1); - adap->setmode(adap, 1); - udelay(adap->mode); - - adap->setmode(adap, 0); - udelay(adap->mode_setup); - sendbyte(adap, addr); - udelay(adap->mode_hold); - - sendbytes(adap, data, len); - - adap->setclk(adap, 1); - adap->setdat(adap, 1); - adap->setmode(adap, 0); - - return len; -} -EXPORT_SYMBOL_GPL(l3_write); - - -static void l3_set_clk(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_clk, val); -} - -static void l3_set_data(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_data, val); -} - -static void l3_set_mode(struct l3_pins *adap, int val) -{ - gpio_set_value(adap->gpio_mode, val); -} - -int l3_set_gpio_ops(struct device *dev, struct l3_pins *adap) -{ - int ret; - - if (!adap->use_gpios) - return -EINVAL; - - ret = devm_gpio_request_one(dev, adap->gpio_data, - GPIOF_OUT_INIT_LOW, "l3_data"); - if (ret < 0) - return ret; - adap->setdat = l3_set_data; - - ret = devm_gpio_request_one(dev, adap->gpio_clk, - GPIOF_OUT_INIT_LOW, "l3_clk"); - if (ret < 0) - return ret; - adap->setclk = l3_set_clk; - - ret = devm_gpio_request_one(dev, adap->gpio_mode, - GPIOF_OUT_INIT_LOW, "l3_mode"); - if (ret < 0) - return ret; - adap->setmode = l3_set_mode; - - return 0; -} -EXPORT_SYMBOL_GPL(l3_set_gpio_ops); - -MODULE_DESCRIPTION("L3 bit-banging driver"); -MODULE_AUTHOR("Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c deleted file mode 100644 index 1a62bec94005..000000000000 --- a/sound/soc/codecs/uda134x.c +++ /dev/null @@ -1,587 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * uda134x.c -- UDA134X ALSA SoC Codec driver - * - * Modifications by Christian Pellegrin <chripell(a)evolware.org> - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie - */ - -#include <linux/module.h> -#include <linux/delay.h> -#include <linux/slab.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/initval.h> - -#include <sound/uda134x.h> -#include <sound/l3.h> - -#include "uda134x.h" - - -#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 -#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) - -struct uda134x_priv { - int sysclk; - int dai_fmt; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; - - struct regmap *regmap; - struct uda134x_platform_data *pd; -}; - -static const struct reg_default uda134x_reg_defaults[] = { - { UDA134X_EA000, 0x04 }, - { UDA134X_EA001, 0x04 }, - { UDA134X_EA010, 0x04 }, - { UDA134X_EA011, 0x00 }, - { UDA134X_EA100, 0x00 }, - { UDA134X_EA101, 0x00 }, - { UDA134X_EA110, 0x00 }, - { UDA134X_EA111, 0x00 }, - { UDA134X_STATUS0, 0x00 }, - { UDA134X_STATUS1, 0x03 }, - { UDA134X_DATA000, 0x00 }, - { UDA134X_DATA001, 0x00 }, - { UDA134X_DATA010, 0x00 }, - { UDA134X_DATA011, 0x00 }, - { UDA134X_DATA1, 0x00 }, -}; - -/* - * Write to the uda134x registers - * - */ -static int uda134x_regmap_write(void *context, unsigned int reg, - unsigned int value) -{ - struct uda134x_platform_data *pd = context; - int ret; - u8 addr; - u8 data = value; - - switch (reg) { - case UDA134X_STATUS0: - case UDA134X_STATUS1: - addr = UDA134X_STATUS_ADDR; - data |= (reg - UDA134X_STATUS0) << 7; - break; - case UDA134X_DATA000: - case UDA134X_DATA001: - case UDA134X_DATA010: - case UDA134X_DATA011: - addr = UDA134X_DATA0_ADDR; - data |= (reg - UDA134X_DATA000) << 6; - break; - case UDA134X_DATA1: - addr = UDA134X_DATA1_ADDR; - break; - default: - /* It's an extended address register */ - addr = (reg | UDA134X_EXTADDR_PREFIX); - - ret = l3_write(&pd->l3, - UDA134X_DATA0_ADDR, &addr, 1); - if (ret != 1) - return -EIO; - - addr = UDA134X_DATA0_ADDR; - data = (value | UDA134X_EXTDATA_PREFIX); - break; - } - - ret = l3_write(&pd->l3, - addr, &data, 1); - if (ret != 1) - return -EIO; - - return 0; -} - -static inline void uda134x_reset(struct snd_soc_component *component) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - unsigned int mask = 1<<6; - - regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, mask); - msleep(1); - regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0); -} - -static int uda134x_mute(struct snd_soc_dai *dai, int mute, int direction) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component); - unsigned int mask = 1<<2; - unsigned int val; - - pr_debug("%s mute: %d\n", __func__, mute); - - if (mute) - val = mask; - else - val = 0; - - return regmap_update_bits(uda134x->regmap, UDA134X_DATA010, mask, val); -} - -static int uda134x_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct snd_pcm_runtime *master_runtime; - - if (uda134x->master_substream) { - master_runtime = uda134x->master_substream->runtime; - - pr_debug("%s constraining to %d bits at %d\n", __func__, - master_runtime->sample_bits, - master_runtime->rate); - - snd_pcm_hw_constraint_single(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate); - - snd_pcm_hw_constraint_single(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits); - - uda134x->slave_substream = substream; - } else - uda134x->master_substream = substream; - - return 0; -} - -static void uda134x_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - if (uda134x->master_substream == substream) - uda134x->master_substream = uda134x->slave_substream; - - uda134x->slave_substream = NULL; -} - -static int uda134x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - unsigned int hw_params = 0; - - if (substream == uda134x->slave_substream) { - pr_debug("%s ignoring hw_params for slave substream\n", - __func__); - return 0; - } - - pr_debug("%s sysclk: %d, rate:%d\n", __func__, - uda134x->sysclk, params_rate(params)); - - /* set SYSCLK / fs ratio */ - switch (uda134x->sysclk / params_rate(params)) { - case 512: - break; - case 384: - hw_params |= (1<<4); - break; - case 256: - hw_params |= (1<<5); - break; - default: - printk(KERN_ERR "%s unsupported fs\n", __func__); - return -EINVAL; - } - - pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, - uda134x->dai_fmt, params_format(params)); - - /* set DAI format and word length */ - switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - break; - case SND_SOC_DAIFMT_RIGHT_J: - switch (params_width(params)) { - case 16: - hw_params |= (1<<1); - break; - case 18: - hw_params |= (1<<2); - break; - case 20: - hw_params |= ((1<<2) | (1<<1)); - break; - default: - printk(KERN_ERR "%s unsupported format (right)\n", - __func__); - return -EINVAL; - } - break; - case SND_SOC_DAIFMT_LEFT_J: - hw_params |= (1<<3); - break; - default: - printk(KERN_ERR "%s unsupported format\n", __func__); - return -EINVAL; - } - - return regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, - STATUS0_SYSCLK_MASK | STATUS0_DAIFMT_MASK, hw_params); -} - -static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_component *component = codec_dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__, - clk_id, freq, dir); - - /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable - because the codec is slave. Of course limitations of the clock - master (the IIS controller) apply. - We'll error out on set_hw_params if it's not OK */ - if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { - uda134x->sysclk = freq; - return 0; - } - - printk(KERN_ERR "%s unsupported sysclk\n", __func__); - return -EINVAL; -} - -static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) -{ - struct snd_soc_component *component = codec_dai->component; - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - - pr_debug("%s fmt: %08X\n", __func__, fmt); - - /* codec supports only full consumer mode */ - if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { - printk(KERN_ERR "%s unsupported clocking mode\n", __func__); - return -EINVAL; - } - - /* no support for clock inversion */ - if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { - printk(KERN_ERR "%s unsupported clock inversion\n", __func__); - return -EINVAL; - } - - /* We can't setup DAI format here as it depends on the word bit num */ - /* so let's just store the value for later */ - uda134x->dai_fmt = fmt; - - return 0; -} - -static int uda134x_set_bias_level(struct snd_soc_component *component, - enum snd_soc_bias_level level) -{ - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct uda134x_platform_data *pd = uda134x->pd; - pr_debug("%s bias level %d\n", __func__, level); - - switch (level) { - case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - /* power on */ - if (pd->power) { - pd->power(1); - regcache_sync(uda134x->regmap); - } - break; - case SND_SOC_BIAS_STANDBY: - break; - case SND_SOC_BIAS_OFF: - /* power off */ - if (pd->power) { - pd->power(0); - regcache_mark_dirty(uda134x->regmap); - } - break; - } - return 0; -} - -static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", - "Minimum2", "Maximum"}; -static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const char *uda134x_mixmode[] = {"Differential", "Analog1", - "Analog2", "Both"}; - -static const struct soc_enum uda134x_mixer_enum[] = { -SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), -SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), -SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), -}; - -static const struct snd_kcontrol_new uda1341_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), -SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), -SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), -SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), - -SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), -SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), - -SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), -SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), - -SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), -SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), - -SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), -SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), -SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), - -SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), -SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), -SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), -SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), -SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -static const struct snd_kcontrol_new uda1340_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), - -SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), -SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), - -SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), - -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -static const struct snd_kcontrol_new uda1345_snd_controls[] = { -SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), - -SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), - -SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), -}; - -/* UDA1341 has the DAC/ADC power down in STATUS1 */ -static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), -}; - -/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ -static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), -}; - -/* Common DAPM widgets */ -static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("VINL1"), - SND_SOC_DAPM_INPUT("VINR1"), - SND_SOC_DAPM_INPUT("VINL2"), - SND_SOC_DAPM_INPUT("VINR2"), - SND_SOC_DAPM_OUTPUT("VOUTL"), - SND_SOC_DAPM_OUTPUT("VOUTR"), -}; - -static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { - { "ADC", NULL, "VINL1" }, - { "ADC", NULL, "VINR1" }, - { "ADC", NULL, "VINL2" }, - { "ADC", NULL, "VINR2" }, - { "VOUTL", NULL, "DAC" }, - { "VOUTR", NULL, "DAC" }, -}; - -static const struct snd_soc_dai_ops uda134x_dai_ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .mute_stream = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - .no_capture_mute = 1, -}; - -static struct snd_soc_dai_driver uda134x_dai = { - .name = "uda134x-hifi", - /* playback capabilities */ - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = UDA134X_RATES, - .formats = UDA134X_FORMATS, - }, - /* capture capabilities */ - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = UDA134X_RATES, - .formats = UDA134X_FORMATS, - }, - /* pcm operations */ - .ops = &uda134x_dai_ops, -}; - -static int uda134x_soc_probe(struct snd_soc_component *component) -{ - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); - struct uda134x_platform_data *pd = uda134x->pd; - const struct snd_soc_dapm_widget *widgets; - unsigned int num_widgets; - int ret; - - printk(KERN_INFO "UDA134X SoC Audio Codec\n"); - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1341: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", - pd->model); - return -EINVAL; - } - - if (pd->power) - pd->power(1); - - uda134x_reset(component); - - if (pd->model == UDA134X_UDA1341) { - widgets = uda1341_dapm_widgets; - num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); - } else { - widgets = uda1340_dapm_widgets; - num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); - } - - ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets); - if (ret) { - printk(KERN_ERR "%s failed to register dapm controls: %d", - __func__, ret); - return ret; - } - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - ret = snd_soc_add_component_controls(component, uda1340_snd_controls, - ARRAY_SIZE(uda1340_snd_controls)); - break; - case UDA134X_UDA1341: - ret = snd_soc_add_component_controls(component, uda1341_snd_controls, - ARRAY_SIZE(uda1341_snd_controls)); - break; - case UDA134X_UDA1345: - ret = snd_soc_add_component_controls(component, uda1345_snd_controls, - ARRAY_SIZE(uda1345_snd_controls)); - break; - default: - printk(KERN_ERR "%s unknown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - if (ret < 0) { - printk(KERN_ERR "UDA134X: failed to register controls\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_component_driver soc_component_dev_uda134x = { - .probe = uda134x_soc_probe, - .set_bias_level = uda134x_set_bias_level, - .dapm_widgets = uda134x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), - .dapm_routes = uda134x_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), - .suspend_bias_off = 1, - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, -}; - -static const struct regmap_config uda134x_regmap_config = { - .reg_bits = 8, - .val_bits = 8, - .max_register = UDA134X_DATA1, - .reg_defaults = uda134x_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(uda134x_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .reg_write = uda134x_regmap_write, -}; - -static int uda134x_codec_probe(struct platform_device *pdev) -{ - struct uda134x_platform_data *pd = pdev->dev.platform_data; - struct uda134x_priv *uda134x; - int ret; - - if (!pd) { - dev_err(&pdev->dev, "Missing L3 bitbang function\n"); - return -ENODEV; - } - - uda134x = devm_kzalloc(&pdev->dev, sizeof(*uda134x), GFP_KERNEL); - if (!uda134x) - return -ENOMEM; - - uda134x->pd = pd; - platform_set_drvdata(pdev, uda134x); - - if (pd->l3.use_gpios) { - ret = l3_set_gpio_ops(&pdev->dev, &uda134x->pd->l3); - if (ret < 0) - return ret; - } - - uda134x->regmap = devm_regmap_init(&pdev->dev, NULL, pd, - &uda134x_regmap_config); - if (IS_ERR(uda134x->regmap)) - return PTR_ERR(uda134x->regmap); - - return devm_snd_soc_register_component(&pdev->dev, - &soc_component_dev_uda134x, &uda134x_dai, 1); -} - -static struct platform_driver uda134x_codec_driver = { - .driver = { - .name = "uda134x-codec", - }, - .probe = uda134x_codec_probe, -}; - -module_platform_driver(uda134x_codec_driver); - -MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h deleted file mode 100644 index 664618c2571c..000000000000 --- a/sound/soc/codecs/uda134x.h +++ /dev/null @@ -1,33 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -#ifndef _UDA134X_CODEC_H -#define _UDA134X_CODEC_H - -#define UDA134X_L3ADDR 5 -#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) -#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) -#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) - -#define UDA134X_EXTADDR_PREFIX 0xC0 -#define UDA134X_EXTDATA_PREFIX 0xE0 - -/* UDA134X registers */ -#define UDA134X_EA000 0 -#define UDA134X_EA001 1 -#define UDA134X_EA010 2 -#define UDA134X_EA011 3 -#define UDA134X_EA100 4 -#define UDA134X_EA101 5 -#define UDA134X_EA110 6 -#define UDA134X_EA111 7 -#define UDA134X_STATUS0 8 -#define UDA134X_STATUS1 9 -#define UDA134X_DATA000 10 -#define UDA134X_DATA001 11 -#define UDA134X_DATA010 12 -#define UDA134X_DATA011 13 -#define UDA134X_DATA1 14 - -#define STATUS0_DAIFMT_MASK (~(7<<1)) -#define STATUS0_SYSCLK_MASK (~(3<<4)) - -#endif --- base-commit: 17b9f4387ebabb19b871bbe2d06562e48e4e7130 change-id: 20230821-delete-l3-573f483201e3 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

3 3

[PATCH] ALSA: ymfpci: Fix the missing snd_card_free() call at probe error
by Takashi Iwai 23 Aug '23

23 Aug '23

Like a few other drivers, YMFPCI driver needs to clean up with snd_card_free() call at an error path of the probe; otherwise the other devres resources are released before the card and it results in the UAF. This patch uses the helper for handling the probe error gracefully. Fixes: f33fc1576757 ("ALSA: ymfpci: Create card with device-managed snd_devm_card_new()") Cc: <stable(a)vger.kernel.org> Reported-and-tested-by: Takashi Yano <takashi.yano(a)nifty.ne.jp> Closes: https://lore.kernel.org/r/20230823135846.1812-1-takashi.yano@nifty.ne.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ymfpci/ymfpci.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index b033bd290940..48444dda44de 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -152,8 +152,8 @@ static inline int snd_ymfpci_create_gameport(struct snd_ymfpci *chip, int dev, i void snd_ymfpci_free_gameport(struct snd_ymfpci *chip) { } #endif /* SUPPORT_JOYSTICK */ -static int snd_card_ymfpci_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +static int __snd_card_ymfpci_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) { static int dev; struct snd_card *card; @@ -348,6 +348,12 @@ static int snd_card_ymfpci_probe(struct pci_dev *pci, return 0; } +static int snd_card_ymfpci_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + return snd_card_free_on_error(&pci->dev, __snd_card_ymfpci_probe(pci, pci_id)); +} + static struct pci_driver ymfpci_driver = { .name = KBUILD_MODNAME, .id_table = snd_ymfpci_ids, -- 2.35.3

1 0

[PATCH v1] Revert "ASoC: Intel: Add rpl_nau8318_8825 driver"
by Ajye Huang 23 Aug '23

23 Aug '23

This reverts commit 6b8b6892e43419f5339c47d307b1baac0eea036d. Report from kernel test robot <lkp(a)intel.com>, that in broonie/ci.git file0yZ50U, it's redundant commit 6b8b6892e434 ("ASoC: Intel: Add rpl_nau8318_8825 driver") because commit 273bc8bf2227 ("ASoC: Intel: Add rpl_nau8318_8825 driver") was committed at 2023-08-14. All errors (new ones prefixed by >>): >> sound/soc/intel/common/soc-acpi-intel-rpl-match.c:369:41: error: redefinition of 'rpl_nau8318_amp' static const struct snd_soc_acpi_codecs rpl_nau8318_amp = { ^ sound/soc/intel/common/soc-acpi-intel-rpl-match.c:359:41: note: previous definition is here static const struct snd_soc_acpi_codecs rpl_nau8318_amp = { Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308221955.FkglAuTF-lkp@intel.com/ Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> --- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 103490ac7c838..122673c1dae27 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -366,11 +366,6 @@ static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = { .codecs = {"RTL1019"} }; -static const struct snd_soc_acpi_codecs rpl_nau8318_amp = { - .num_codecs = 1, - .codecs = {"NVTN2012"} -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { { .comp_ids = &rpl_rt5682_hp, -- 2.25.1

2 4

snd_seq_ev_set_ump_data() stack smashing
by GitHub issues - opened 23 Aug '23

23 Aug '23

alsa-project/alsa-lib issue #346 was opened from symdeb: snd_seq_ev_set_ump_data(&umpev, buffer, len); When called with len > 4 causes stack smashing. A ump messages is maximum four 32bit values. Proposal solution, return error when len > 4 ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/346 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] fix: Add quirk for Asus Zenbook Pro 14 UX6404 laptop
by Rohit Pidaparthi 23 Aug '23

23 Aug '23

From 4f14cdad4f526f45747c4f0b71e24a46ebe3885c Mon Sep 17 00:00:00 2001 From: Rohit Pidaparthi <rohitpid(a)gmail.com> Date: Sat, 19 Aug 2023 02:15:16 -0700 Subject: [PATCH] fix: Add quirk for Asus Zenbook Pro 14 UX6404 laptop The Asus Zenbook Pro 14 UX6404 (1043:1863) needs binding to the CS35L41 codec over spi1 rather than spi0. This is similar to existing quirks for many ASUS Laptops. Signed-off-by: Rohit Pidaparthi <rohitpid(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc7b7a407638..c44464b1619f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9681,6 +9681,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), -- 2.39.2

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aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (if longer, the issue is not reproduceable) 6. Audio stops and soon after the error appears stopping the video. pavuctl screen goes blank and reinitializes itself. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 2 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (if longer, the issue may not be reproduceable) 6. Audio stops and soon after the error appears stopping the video. pavuctl screen goes blank and reinitializes itself. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 2 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (if longer, the issue may not be reproduceable) 6. Audio stops and soon after the error appears stopping the video. pavuctl screen goes blank and seems the reinitialize itself. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 2 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (if longer, the issue may not be reproduceable) 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 2 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (if longer, the issue may not be reproduceable) 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. play a video with the "video" application. 3. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. run aplaymidi -l or aplaymidi --version within about 30 seconds of plugging in the device. (longer the issue may not be reproduceable) 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0