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[tiwai-sound:for-next 73/83] sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared; did you mean 'chip'?
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: d41abde894830bfb77252653f606473728e930eb [73/83] ALSA: sparc: Use standard print API config: sparc64-defconfig (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) compiler: sparc64-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090648.J2EAijjH-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090648.J2EAijjH-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from include/linux/device.h:15, from include/linux/platform_device.h:13, from sound/sparc/cs4231.c:21: sound/sparc/cs4231.c: In function 'snd_cs4231_ebus_create': >> sound/sparc/cs4231.c:2018:25: error: 'chpi' undeclared (first use in this function); did you mean 'chip'? 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ sound/sparc/cs4231.c:2018:25: note: each undeclared identifier is reported only once for each function it appears in 2018 | dev_dbg(chpi->card->dev, | ^~~~ include/linux/dev_printk.h:139:44: note: in definition of macro 'dev_no_printk' 139 | _dev_printk(level, dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/sparc/cs4231.c:2018:17: note: in expansion of macro 'dev_dbg' 2018 | dev_dbg(chpi->card->dev, | ^~~~~~~ vim +2018 sound/sparc/cs4231.c 1950 1951 static int snd_cs4231_ebus_create(struct snd_card *card, 1952 struct platform_device *op, 1953 int dev) 1954 { 1955 struct snd_cs4231 *chip = card->private_data; 1956 int err; 1957 1958 spin_lock_init(&chip->lock); 1959 spin_lock_init(&chip->c_dma.ebus_info.lock); 1960 spin_lock_init(&chip->p_dma.ebus_info.lock); 1961 mutex_init(&chip->mce_mutex); 1962 mutex_init(&chip->open_mutex); 1963 chip->flags |= CS4231_FLAG_EBUS; 1964 chip->op = op; 1965 memcpy(&chip->image, &snd_cs4231_original_image, 1966 sizeof(snd_cs4231_original_image)); 1967 strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); 1968 chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1969 chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; 1970 chip->c_dma.ebus_info.client_cookie = chip; 1971 chip->c_dma.ebus_info.irq = op->archdata.irqs[0]; 1972 strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); 1973 chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; 1974 chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; 1975 chip->p_dma.ebus_info.client_cookie = chip; 1976 chip->p_dma.ebus_info.irq = op->archdata.irqs[1]; 1977 1978 chip->p_dma.prepare = _ebus_dma_prepare; 1979 chip->p_dma.enable = _ebus_dma_enable; 1980 chip->p_dma.request = _ebus_dma_request; 1981 chip->p_dma.address = _ebus_dma_addr; 1982 1983 chip->c_dma.prepare = _ebus_dma_prepare; 1984 chip->c_dma.enable = _ebus_dma_enable; 1985 chip->c_dma.request = _ebus_dma_request; 1986 chip->c_dma.address = _ebus_dma_addr; 1987 1988 chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); 1989 chip->p_dma.ebus_info.regs = 1990 of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); 1991 chip->c_dma.ebus_info.regs = 1992 of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); 1993 if (!chip->port || !chip->p_dma.ebus_info.regs || 1994 !chip->c_dma.ebus_info.regs) { 1995 snd_cs4231_ebus_free(chip); 1996 dev_dbg(chip->card->dev, 1997 "cs4231-%d: Unable to map chip registers.\n", dev); 1998 return -EIO; 1999 } 2000 2001 if (ebus_dma_register(&chip->c_dma.ebus_info)) { 2002 snd_cs4231_ebus_free(chip); 2003 dev_dbg(chip->card->dev, 2004 "cs4231-%d: Unable to register EBUS capture DMA\n", 2005 dev); 2006 return -EBUSY; 2007 } 2008 if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) { 2009 snd_cs4231_ebus_free(chip); 2010 dev_dbg(chip->card->dev, 2011 "cs4231-%d: Unable to enable EBUS capture IRQ\n", 2012 dev); 2013 return -EBUSY; 2014 } 2015 2016 if (ebus_dma_register(&chip->p_dma.ebus_info)) { 2017 snd_cs4231_ebus_free(chip); > 2018 dev_dbg(chpi->card->dev, 2019 "cs4231-%d: Unable to register EBUS play DMA\n", 2020 dev); 2021 return -EBUSY; 2022 } 2023 if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) { 2024 snd_cs4231_ebus_free(chip); 2025 dev_dbg(chip->card->dev, 2026 "cs4231-%d: Unable to enable EBUS play IRQ\n", dev); 2027 return -EBUSY; 2028 } 2029 2030 if (snd_cs4231_probe(chip) < 0) { 2031 snd_cs4231_ebus_free(chip); 2032 return -ENODEV; 2033 } 2034 snd_cs4231_init(chip); 2035 2036 err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, 2037 chip, &snd_cs4231_ebus_dev_ops); 2038 if (err < 0) { 2039 snd_cs4231_ebus_free(chip); 2040 return err; 2041 } 2042 2043 return 0; 2044 } 2045 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 32/83] sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core'
by kernel test robot 09 Aug '24

09 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: c46fc83e3f3c89f18962e43890de90b1c304747a commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/83] ALSA: vx_core: Drop unused dev field config: x86_64-allyesconfig (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090607.58bm8VK5-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090607.58bm8VK5-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD c46fc83e3f3c89f18962e43890de90b1c304747a builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pcmcia/vx/vxpocket.c:208:8: error: no member named 'dev' in 'struct vx_core' 208 | chip->dev = &link->dev; | ~~~~ ^ 1 error generated. vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v2] ASoC: meson: axg-fifo: fix irq scheduling issue with PREEMPT_RT
by Jerome Brunet 08 Aug '24

08 Aug '24

With PREEMPT_RT enabled a spinlock_t becomes a sleeping lock. This is usually not a problem with spinlocks used in IRQ context since IRQ handlers get threaded. However, if IRQF_ONESHOT is set, the primary handler won't be force-threaded and runs always in hardirq context. This is a problem because spinlock_t requires a preemptible context on PREEMPT_RT. In this particular instance, regmap mmio uses spinlock_t to protect the register access and IRQF_ONESHOT is set on the IRQ. In this case, it is actually better to do everything in threaded handler and it solves the problem with PREEMPT_RT. Reported-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> Closes: https://lore.kernel.org/linux-amlogic/20240729131652.3012327-1-avkrasnov@sa… Suggested-by: Sebastian Andrzej Siewior <bigeasy(a)linutronix.de> Fixes: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 26 ++++++++++---------------- 1 file changed, 10 insertions(+), 16 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 7e6090af720b..75909196b769 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -207,25 +207,18 @@ static irqreturn_t axg_fifo_pcm_irq_block(int irq, void *dev_id) status = FIELD_GET(STATUS1_INT_STS, status); axg_fifo_ack_irq(fifo, status); - /* Use the thread to call period elapsed on nonatomic links */ - if (status & FIFO_INT_COUNT_REPEAT) - return IRQ_WAKE_THREAD; + if (status & ~FIFO_INT_COUNT_REPEAT) + dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", + status); - dev_dbg(axg_fifo_dev(ss), "unexpected irq - STS 0x%02x\n", - status); + if (status & FIFO_INT_COUNT_REPEAT) { + snd_pcm_period_elapsed(ss); + return IRQ_HANDLED; + } return IRQ_NONE; } -static irqreturn_t axg_fifo_pcm_irq_block_thread(int irq, void *dev_id) -{ - struct snd_pcm_substream *ss = dev_id; - - snd_pcm_period_elapsed(ss); - - return IRQ_HANDLED; -} - int axg_fifo_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *ss) { @@ -251,8 +244,9 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, if (ret) return ret; - ret = request_threaded_irq(fifo->irq, axg_fifo_pcm_irq_block, - axg_fifo_pcm_irq_block_thread, + /* Use the threaded irq handler only with non-atomic links */ + ret = request_threaded_irq(fifo->irq, NULL, + axg_fifo_pcm_irq_block, IRQF_ONESHOT, dev_name(dev), ss); if (ret) return ret; -- 2.43.0

3 2

[PATCH 1/4] ASoC: dt-bindings: qcom,wcd937x: Correct reset GPIO polarity in example
by Krzysztof Kozlowski 08 Aug '24

08 Aug '24

The reset GPIO of WCD9370/WCD9375 is active low and that's how it is routed on typical boards, so correct the example DTS to use expected polarity. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Cc: Alexey Klimov <alexey.klimov(a)linaro.org> --- Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml index de397d879acc..f94203798f24 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml @@ -42,7 +42,7 @@ examples: pinctrl-names = "default", "sleep"; pinctrl-0 = <&wcd_reset_n>; pinctrl-1 = <&wcd_reset_n_sleep>; - reset-gpios = <&tlmm 83 GPIO_ACTIVE_HIGH>; + reset-gpios = <&tlmm 83 GPIO_ACTIVE_LOW>; vdd-buck-supply = <&vreg_l17b_1p8>; vdd-rxtx-supply = <&vreg_l18b_1p8>; vdd-px-supply = <&vreg_l18b_1p8>; -- 2.43.0

2 4

[PATCH v2 0/3] ASoC: tas*: Fix up GPIO usage
by Linus Walleij 08 Aug '24

08 Aug '24

The TI TAS drivers use some legacy GPIO code and headers, this series fixes it up. The TAS2781 is a special case since it adds a handful of lines of deviating code to reconfigure a GPIO line for IRQ mode and then never actually use the IRQ obtained in the code. Is the line used by autonomous hardware? I'm puzzled by this. Anyways the patch suggest how to solve this properly by fixing the parent irqchip and I'm happy to help. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Rebased onto v6.11-rc1 - Link to v1: https://lore.kernel.org/r/20240701-asoc-tas-gpios-v1-0-d69ec5d79939@linaro.… --- Linus Walleij (3): ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 26 ++++---------------------- 9 files changed, 6 insertions(+), 37 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240701-asoc-tas-gpios-5c051d80d768 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

2 4

[PATCH] ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value
by Simon Trimmer 08 Aug '24

08 Aug '24

Device tuning files made with early revision tooling may contain configuration that can unmask IRQ signals that are owned by the host. Adding a safe default to the regmap patch ensures that the hardware matches the driver expectations. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..bd74fef33d49 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -36,6 +36,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, + { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, -- 2.43.0

2 1

[PATCH 0/2] ASoC: codecs: ES8326: button and input issue
by Zhang Yi 08 Aug '24

08 Aug '24

TO solve button and input issue, We have modified parts of the driver related to button and input. Zhang Yi (2): ASoC: codecs: ES8326: button detect issue ASoC: codecs: ES8326: input issue after init sound/soc/codecs/es8326.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.17.1

2 3

[PATCH] ASoC: amd: acp: add ZSC control register programming sequence
by Vijendar Mukunda 08 Aug '24

08 Aug '24

Add ZSC Control register programming sequence for ACP D0 and D3 state transitions for ACP7.0 onwards. This will allow ACP to enter low power state when ACP enters D3 state. When ACP enters D0 State, ZSC control should be disabled. Tested-by: Leo Li <sunpeng.li(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 5 +++++ sound/soc/amd/acp/amd.h | 2 ++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 4422cec81e3c..04bd605fdce3 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -321,6 +321,8 @@ int acp_init(struct acp_chip_info *chip) pr_err("ACP reset failed\n"); return ret; } + if (chip->acp_rev >= ACP70_DEV) + writel(0, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_init, SND_SOC_ACP_COMMON); @@ -336,6 +338,9 @@ int acp_deinit(struct acp_chip_info *chip) if (chip->acp_rev != ACP70_DEV) writel(0, chip->base + ACP_CONTROL); + + if (chip->acp_rev >= ACP70_DEV) + writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_deinit, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 87a4813783f9..c095a34a7229 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -103,6 +103,8 @@ #define ACP70_PGFSM_CONTROL ACP6X_PGFSM_CONTROL #define ACP70_PGFSM_STATUS ACP6X_PGFSM_STATUS +#define ACP_ZSC_DSP_CTRL 0x0001014 +#define ACP_ZSC_STS 0x0001018 #define ACP_SOFT_RST_DONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 0xffffffff -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver
by Vijendar Mukunda 08 Aug '24

08 Aug '24

As module device table added for AMD SoundWire machine driver MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD SoundWire machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 3419675e45a9..f7e4af850309 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -738,5 +738,4 @@ module_platform_driver(sof_sdw_driver); MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); MODULE_AUTHOR("Vijendar Mukunda <Vijendar.Mukunda(a)amd.com"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:amd_sof_sdw"); MODULE_IMPORT_NS(SND_SOC_SDW_UTILS); -- 2.34.1

4 3

[PATCH] ASoC: cs35l56: Use regmap_read_bypassed() to wake the device
by Simon Trimmer 08 Aug '24

08 Aug '24

Now that regmap_read_bypassed() has been added to the kernel it is preferable to wake the device with a read rather than a write as the utility function can be called at a time before the device has been identified. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..91b3c1c8575c 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -450,32 +450,23 @@ static const struct reg_sequence cs35l56_hibernate_seq[] = { REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_ALLOW_AUTO_HIBERNATE), }; -static const struct reg_sequence cs35l56_hibernate_wake_seq[] = { - REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_WAKEUP), -}; - static void cs35l56_issue_wake_event(struct cs35l56_base *cs35l56_base) { + unsigned int val; + /* * Dummy transactions to trigger I2C/SPI auto-wake. Issue two * transactions to meet the minimum required time from the rising edge * to the last falling edge of wake. * - * It uses bypassed write because we must wake the chip before + * It uses bypassed read because we must wake the chip before * disabling regmap cache-only. - * - * This can NAK on I2C which will terminate the write sequence so the - * single-write sequence is issued twice. */ - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); usleep_range(CS35L56_WAKE_HOLD_TIME_US, 2 * CS35L56_WAKE_HOLD_TIME_US); - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); cs35l56_wait_control_port_ready(); } -- 2.43.0

2 1

[tiwai-sound:for-next 66/82] sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: f8466d91f36d97e237ea1b3c6f19de980054a1b6 [66/82] ALSA: usx2y: Use standard print API config: x86_64-randconfig-161-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090249.8LE9qrae-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090249.8LE9qrae-lkp@intel.com/ All warnings (new ones prefixed by >>): In file included from sound/usb/usx2y/us122l.c:15: >> sound/usb/usx2y/usb_stream.c:668:17: warning: variable 'iters' set but not used [-Wunused-but-set-variable] 668 | int frame = 0, iters = 0; | ^ 1 warning generated. vim +/iters +668 sound/usb/usx2y/usb_stream.c 030a07e441296c Karsten Wiese 2008-07-30 664 030a07e441296c Karsten Wiese 2008-07-30 665 int usb_stream_start(struct usb_stream_kernel *sk) 030a07e441296c Karsten Wiese 2008-07-30 666 { 030a07e441296c Karsten Wiese 2008-07-30 667 struct usb_stream *s = sk->s; 030a07e441296c Karsten Wiese 2008-07-30 @668 int frame = 0, iters = 0; 030a07e441296c Karsten Wiese 2008-07-30 669 int u, err; 030a07e441296c Karsten Wiese 2008-07-30 670 int try = 0; 030a07e441296c Karsten Wiese 2008-07-30 671 030a07e441296c Karsten Wiese 2008-07-30 672 if (s->state != usb_stream_stopped) 030a07e441296c Karsten Wiese 2008-07-30 673 return -EAGAIN; 030a07e441296c Karsten Wiese 2008-07-30 674 030a07e441296c Karsten Wiese 2008-07-30 675 subs_set_complete(sk->inurb, i_capture_start); 030a07e441296c Karsten Wiese 2008-07-30 676 subs_set_complete(sk->outurb, i_playback_start); 030a07e441296c Karsten Wiese 2008-07-30 677 memset(sk->write_page, 0, s->write_size); 030a07e441296c Karsten Wiese 2008-07-30 678 dotry: 030a07e441296c Karsten Wiese 2008-07-30 679 s->insize_done = 0; 030a07e441296c Karsten Wiese 2008-07-30 680 s->idle_insize = 0; 030a07e441296c Karsten Wiese 2008-07-30 681 s->idle_outsize = 0; 030a07e441296c Karsten Wiese 2008-07-30 682 s->sync_packet = -1; 030a07e441296c Karsten Wiese 2008-07-30 683 s->inpacket_head = -1; 030a07e441296c Karsten Wiese 2008-07-30 684 sk->iso_frame_balance = 0; 030a07e441296c Karsten Wiese 2008-07-30 685 ++try; 030a07e441296c Karsten Wiese 2008-07-30 686 for (u = 0; u < 2; u++) { 030a07e441296c Karsten Wiese 2008-07-30 687 struct urb *inurb = sk->inurb[u]; 030a07e441296c Karsten Wiese 2008-07-30 688 struct urb *outurb = sk->outurb[u]; 4c0a58ef36f3de Takashi Iwai 2021-05-17 689 030a07e441296c Karsten Wiese 2008-07-30 690 playback_prep_freqn(sk, outurb); 030a07e441296c Karsten Wiese 2008-07-30 691 inurb->number_of_packets = outurb->number_of_packets; 030a07e441296c Karsten Wiese 2008-07-30 692 inurb->transfer_buffer_length = 030a07e441296c Karsten Wiese 2008-07-30 693 inurb->number_of_packets * 030a07e441296c Karsten Wiese 2008-07-30 694 inurb->iso_frame_desc[0].length; d0f3a2eb906256 Karsten Wiese 2011-12-30 695 030a07e441296c Karsten Wiese 2008-07-30 696 if (u == 0) { 030a07e441296c Karsten Wiese 2008-07-30 697 int now; 030a07e441296c Karsten Wiese 2008-07-30 698 struct usb_device *dev = inurb->dev; 4c0a58ef36f3de Takashi Iwai 2021-05-17 699 030a07e441296c Karsten Wiese 2008-07-30 700 frame = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 701 do { 030a07e441296c Karsten Wiese 2008-07-30 702 now = usb_get_current_frame_number(dev); 030a07e441296c Karsten Wiese 2008-07-30 703 ++iters; 030a07e441296c Karsten Wiese 2008-07-30 704 } while (now > -1 && now == frame); 030a07e441296c Karsten Wiese 2008-07-30 705 } 030a07e441296c Karsten Wiese 2008-07-30 706 err = usb_submit_urb(inurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 707 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 708 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 709 "%s: usb_submit_urb(sk->inurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 710 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 711 return err; 030a07e441296c Karsten Wiese 2008-07-30 712 } 030a07e441296c Karsten Wiese 2008-07-30 713 err = usb_submit_urb(outurb, GFP_ATOMIC); 030a07e441296c Karsten Wiese 2008-07-30 714 if (err < 0) { f8466d91f36d97 Takashi Iwai 2024-08-07 715 dev_err(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 716 "%s: usb_submit_urb(sk->outurb[%i]) returned %i\n", f8466d91f36d97 Takashi Iwai 2024-08-07 717 __func__, u, err); 030a07e441296c Karsten Wiese 2008-07-30 718 return err; 030a07e441296c Karsten Wiese 2008-07-30 719 } d0f3a2eb906256 Karsten Wiese 2011-12-30 720 030a07e441296c Karsten Wiese 2008-07-30 721 if (inurb->start_frame != outurb->start_frame) { f8466d91f36d97 Takashi Iwai 2024-08-07 722 dev_dbg(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 723 "%s: u[%i] start_frames differ in:%u out:%u\n", f8466d91f36d97 Takashi Iwai 2024-08-07 724 __func__, u, inurb->start_frame, outurb->start_frame); 030a07e441296c Karsten Wiese 2008-07-30 725 goto check_retry; 030a07e441296c Karsten Wiese 2008-07-30 726 } 030a07e441296c Karsten Wiese 2008-07-30 727 } 030a07e441296c Karsten Wiese 2008-07-30 728 try = 0; 030a07e441296c Karsten Wiese 2008-07-30 729 check_retry: 030a07e441296c Karsten Wiese 2008-07-30 730 if (try) { 030a07e441296c Karsten Wiese 2008-07-30 731 usb_stream_stop(sk); 030a07e441296c Karsten Wiese 2008-07-30 732 if (try < 5) { 030a07e441296c Karsten Wiese 2008-07-30 733 msleep(1500); f8466d91f36d97 Takashi Iwai 2024-08-07 734 dev_dbg(&sk->dev->dev, "goto dotry;\n"); 030a07e441296c Karsten Wiese 2008-07-30 735 goto dotry; 030a07e441296c Karsten Wiese 2008-07-30 736 } f8466d91f36d97 Takashi Iwai 2024-08-07 737 dev_warn(&sk->dev->dev, f8466d91f36d97 Takashi Iwai 2024-08-07 738 "%s: couldn't start all urbs on the same start_frame.\n", f8466d91f36d97 Takashi Iwai 2024-08-07 739 __func__); 030a07e441296c Karsten Wiese 2008-07-30 740 return -EFAULT; 030a07e441296c Karsten Wiese 2008-07-30 741 } 030a07e441296c Karsten Wiese 2008-07-30 742 030a07e441296c Karsten Wiese 2008-07-30 743 sk->idle_inurb = sk->inurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 744 sk->idle_outurb = sk->outurb[USB_STREAM_NURBS - 2]; 030a07e441296c Karsten Wiese 2008-07-30 745 sk->completed_inurb = sk->inurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 746 sk->completed_outurb = sk->outurb[USB_STREAM_NURBS - 1]; 030a07e441296c Karsten Wiese 2008-07-30 747 :::::: The code at line 668 was first introduced by commit :::::: 030a07e441296c372f946cd4065b5d831d8dc40c ALSA: Add USB US122L driver :::::: TO: Karsten Wiese <fzu(a)wemgehoertderstaat.de> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 69/82] sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared; did you mean 'p_dev'?
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 commit: 2acbb5e57230830016e0fb2d61886e7a8c7f59ce [69/82] ALSA: vxpocket: Use standard print API config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090110.t0mWbTyh-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090110.t0mWbTyh-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': sound/pcmcia/vx/vxpocket.c:207:13: error: 'struct vx_core' has no member named 'dev' 207 | chip->dev = &link->dev; | ^~ In file included from include/linux/device.h:15, from include/sound/core.h:10, from sound/pcmcia/vx/vxpocket.c:13: sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_probe': >> sound/pcmcia/vx/vxpocket.c:273:26: error: 'pdev' undeclared (first use in this function); did you mean 'p_dev'? 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ sound/pcmcia/vx/vxpocket.c:273:26: note: each undeclared identifier is reported only once for each function it appears in 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~ include/linux/dev_printk.h:110:25: note: in definition of macro 'dev_printk_index_wrap' 110 | _p_func(dev, fmt, ##__VA_ARGS__); \ | ^~~ sound/pcmcia/vx/vxpocket.c:273:17: note: in expansion of macro 'dev_err' 273 | dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); | ^~~~~~~ vim +273 sound/pcmcia/vx/vxpocket.c 247 248 249 /* 250 */ 251 static int vxpocket_probe(struct pcmcia_device *p_dev) 252 { 253 struct snd_card *card; 254 struct snd_vxpocket *vxp; 255 int i, err; 256 257 /* find an empty slot from the card list */ 258 for (i = 0; i < SNDRV_CARDS; i++) { 259 if (!(card_alloc & (1 << i))) 260 break; 261 } 262 if (i >= SNDRV_CARDS) { 263 dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); 264 return -EINVAL; 265 } 266 if (! enable[i]) 267 return -ENODEV; /* disabled explicitly */ 268 269 /* ok, create a card instance */ 270 err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 271 0, &card); 272 if (err < 0) { > 273 dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); 274 return err; 275 } 276 277 err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); 278 if (err < 0) { 279 snd_card_free(card); 280 return err; 281 } 282 card->private_data = vxp; 283 284 vxp->index = i; 285 card_alloc |= 1 << i; 286 287 vxp->p_dev = p_dev; 288 289 return vxpocket_config(p_dev); 290 } 291 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ALSA: snd-usb-caiaq: use snd_pcm_rate_to_rate_bit
by Jerome Brunet 08 Aug '24

08 Aug '24

Use snd_pcm_rate_to_rate_bit() helper provided by Alsa instead re-implementing it. This reduce code duplication and helps when changing some Alsa definition is necessary. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/usb/caiaq/audio.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4981753652a7..e62a4ea1d19c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -174,14 +174,6 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) return 0; } -/* this should probably go upstream */ -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 -#error "Change this table" -#endif - -static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; - static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { int bytes_per_sample, bpp, ret, i; @@ -233,10 +225,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ - for (i=0; i < ARRAY_SIZE(rates); i++) - if (runtime->rate == rates[i]) - cdev->pcm_info.rates = 1 << i; - + cdev->pcm_info.rates = snd_pcm_rate_to_rate_bit(runtime->rate); snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; -- 2.43.0

2 1

[tiwai-sound:for-next 32/79] sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev'
by kernel test robot 08 Aug '24

08 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b88d0890ed9adab413ea991fac90842688e1017 commit: b426b3ba9f6fe17990893c5324727dd217d420b6 [32/79] ALSA: vx_core: Drop unused dev field config: i386-randconfig-141-20240808 (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240809/202408090016.kW0TA6fc-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202408090016.kW0TA6fc-lkp@intel.com/ All errors (new ones prefixed by >>): sound/pcmcia/vx/vxpocket.c: In function 'vxpocket_config': >> sound/pcmcia/vx/vxpocket.c:208:13: error: 'struct vx_core' has no member named 'dev' 208 | chip->dev = &link->dev; | ^~ vim +208 sound/pcmcia/vx/vxpocket.c 6d00a3127972e7 Takashi Iwai 2005-06-30 170 6d00a3127972e7 Takashi Iwai 2005-06-30 171 6d00a3127972e7 Takashi Iwai 2005-06-30 172 /* 6d00a3127972e7 Takashi Iwai 2005-06-30 173 * configuration callback 6d00a3127972e7 Takashi Iwai 2005-06-30 174 */ 6d00a3127972e7 Takashi Iwai 2005-06-30 175 15b99ac1729503 Dominik Brodowski 2006-03-31 176 static int vxpocket_config(struct pcmcia_device *link) 6d00a3127972e7 Takashi Iwai 2005-06-30 177 { af26367f69a474 Takashi Iwai 2005-11-17 178 struct vx_core *chip = link->priv; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 179 int ret; 6d00a3127972e7 Takashi Iwai 2005-06-30 180 6d00a3127972e7 Takashi Iwai 2005-06-30 181 snd_printdd(KERN_DEBUG "vxpocket_config called\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 182 6d00a3127972e7 Takashi Iwai 2005-06-30 183 /* redefine hardware record according to the VERSION1 string */ af2b3b503ad1b0 Dominik Brodowski 2006-10-25 184 if (!strcmp(link->prod_id[1], "VX-POCKET")) { 6d00a3127972e7 Takashi Iwai 2005-06-30 185 snd_printdd("VX-pocket is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 186 } else { 6d00a3127972e7 Takashi Iwai 2005-06-30 187 snd_printdd("VX-pocket 440 is detected\n"); 6d00a3127972e7 Takashi Iwai 2005-06-30 188 /* overwrite the hardware information */ 6d00a3127972e7 Takashi Iwai 2005-06-30 189 chip->hw = &vxp440_hw; 6d00a3127972e7 Takashi Iwai 2005-06-30 190 chip->type = vxp440_hw.type; 6d00a3127972e7 Takashi Iwai 2005-06-30 191 strcpy(chip->card->driver, vxp440_hw.name); 6d00a3127972e7 Takashi Iwai 2005-06-30 192 } 6d00a3127972e7 Takashi Iwai 2005-06-30 193 90abdc3b973229 Dominik Brodowski 2010-07-24 194 ret = pcmcia_request_io(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 195 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 196 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 197 db0a5214b8d6cc Takashi Iwai 2014-09-09 198 ret = request_threaded_irq(link->irq, snd_vx_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 199 snd_vx_threaded_irq_handler, db0a5214b8d6cc Takashi Iwai 2014-09-09 200 IRQF_SHARED, link->devname, link->priv); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 201 if (ret) db0a5214b8d6cc Takashi Iwai 2014-09-09 202 goto failed_preirq; 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 203 1ac71e5a35eebe Dominik Brodowski 2010-07-29 204 ret = pcmcia_enable_device(link); 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 205 if (ret) 7c5af6ffd69bb2 Dominik Brodowski 2009-10-24 206 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 207 dd2e5a156525f1 Dominik Brodowski 2009-11-03 @208 chip->dev = &link->dev; 6d00a3127972e7 Takashi Iwai 2005-06-30 209 9a017a910346af Dominik Brodowski 2010-07-24 210 if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, 9a017a910346af Dominik Brodowski 2010-07-24 211 link->irq) < 0) 6d00a3127972e7 Takashi Iwai 2005-06-30 212 goto failed; 6d00a3127972e7 Takashi Iwai 2005-06-30 213 79ca4f3f625e14 Takashi Iwai 2006-04-12 214 return 0; 6d00a3127972e7 Takashi Iwai 2005-06-30 215 6d00a3127972e7 Takashi Iwai 2005-06-30 216 failed: db0a5214b8d6cc Takashi Iwai 2014-09-09 217 free_irq(link->irq, link->priv); db0a5214b8d6cc Takashi Iwai 2014-09-09 218 failed_preirq: fba395eee7d3f3 Dominik Brodowski 2006-03-31 219 pcmcia_disable_device(link); 15b99ac1729503 Dominik Brodowski 2006-03-31 220 return -ENODEV; 6d00a3127972e7 Takashi Iwai 2005-06-30 221 } 6d00a3127972e7 Takashi Iwai 2005-06-30 222 :::::: The code at line 208 was first introduced by commit :::::: dd2e5a156525f11754d9b1e0583f6bb49c253d62 pcmcia: remove deprecated handle_to_dev() macro :::::: TO: Dominik Brodowski <linux(a)dominikbrodowski.net> :::::: CC: Dominik Brodowski <linux(a)dominikbrodowski.net> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 08 Aug '24

08 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support" Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: SOF: Intel: add initial support for PTL ASoC: Intel: soc-acpi: add PTL match tables include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 121 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 +++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 ++- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 9

[PATCH 2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 07 Aug '24

07 Aug '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. There is no reason that the UCM or user should need to interact with any of the ALSA controls for the firmware coefficients so they can be removed entirely. Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a51..757ade6373ed 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1095,6 +1095,11 @@ int cs35l56_system_resume(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_resume); +static int cs35l56_control_add_nop(struct wm_adsp *dsp, struct cs_dsp_coeff_ctl *cs_ctl) +{ + return 0; +} + static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) { struct wm_adsp *dsp; @@ -1117,6 +1122,12 @@ static int cs35l56_dsp_init(struct cs35l56_private *cs35l56) dsp->fw = 12; dsp->wmfw_optional = true; + /* + * None of the firmware controls need to be exported so add a no-op + * callback that suppresses creating an ALSA control. + */ + dsp->control_add = &cs35l56_control_add_nop; + dev_dbg(cs35l56->base.dev, "DSP system name: '%s'\n", dsp->system_name); ret = wm_halo_init(dsp); -- 2.43.0

1 0

[PATCH alsa-lib] seq: Add snd_seq_{get|set}_ump_is_midi1() API functions
by Takashi Iwai 07 Aug '24

07 Aug '24

Implement the API function calls corresponding to the new sequencer port flag bit that has been added recently to the kernel. A UMP MIDI 2.0 device allow to have an optional MIDI 1.0 port while speaking in MIDI 2.0 protocol for other UMP Groups. The new seq port flag indicates that. This is rather a minor difference, and since ALSA sequencer core covers the all conversions, usually you don't have to worry about it at all. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 2 ++ include/sound/uapi/asequencer.h | 2 ++ src/Versions.in.in | 2 ++ src/seq/seq.c | 29 +++++++++++++++++++++++++++++ 4 files changed, 35 insertions(+) diff --git a/include/seq.h b/include/seq.h index 4f03df438288..2eee95a643b4 100644 --- a/include/seq.h +++ b/include/seq.h @@ -318,6 +318,7 @@ int snd_seq_port_info_get_timestamp_real(const snd_seq_port_info_t *info); int snd_seq_port_info_get_timestamp_queue(const snd_seq_port_info_t *info); int snd_seq_port_info_get_direction(const snd_seq_port_info_t *info); int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info); +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info); void snd_seq_port_info_set_client(snd_seq_port_info_t *info, int client); void snd_seq_port_info_set_port(snd_seq_port_info_t *info, int port); @@ -334,6 +335,7 @@ void snd_seq_port_info_set_timestamp_real(snd_seq_port_info_t *info, int realtim void snd_seq_port_info_set_timestamp_queue(snd_seq_port_info_t *info, int queue); void snd_seq_port_info_set_direction(snd_seq_port_info_t *info, int direction); void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group); +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1); int snd_seq_create_port(snd_seq_t *handle, snd_seq_port_info_t *info); int snd_seq_delete_port(snd_seq_t *handle, int port); diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index 923dfdddfc76..b3e9df5894af 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -477,6 +477,8 @@ struct snd_seq_remove_events { #define SNDRV_SEQ_PORT_FLG_TIMESTAMP (1<<1) #define SNDRV_SEQ_PORT_FLG_TIME_REAL (1<<2) +#define SNDRV_SEQ_PORT_FLG_IS_MIDI1 (1<<3) /* Keep MIDI 1.0 protocol */ + /* port direction */ #define SNDRV_SEQ_PORT_DIR_UNKNOWN 0 #define SNDRV_SEQ_PORT_DIR_INPUT 1 diff --git a/src/Versions.in.in b/src/Versions.in.in index 90849277c983..298e610c2099 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -209,6 +209,8 @@ ALSA_1.2.13 { @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; @SYMBOL_PREFIX@snd_seq_has_tempo_base; + @SYMBOL_PREFIX@snd_seq_port_info_get_ump_is_midi1; + @SYMBOL_PREFIX@snd_seq_port_info_set_ump_is_midi1; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index d0ecedf0838f..347ff45544fa 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -2431,6 +2431,19 @@ int snd_seq_port_info_get_ump_group(const snd_seq_port_info_t *info) return info->ump_group; } +/** + * \brief Get the status of the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \return 1 if it's an optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_set_ump_is_midi1() + */ +int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t *info) +{ + assert(info); + return !!(info->flags & SNDRV_SEQ_PORT_FLG_IS_MIDI1); +} + /** * \brief Set the client id of a port_info container * \param info port_info container @@ -2635,6 +2648,22 @@ void snd_seq_port_info_set_ump_group(snd_seq_port_info_t *info, int ump_group) info->ump_group = ump_group; } +/** + * \brief Set the optional MIDI 1.0 port in MIDI 2.0 UMP Endpoint + * \param info port_info container + * \param is_midi1 non-zero for MIDI 1.0 port in MIDI 2.0 EP + * + * \sa snd_seq_get_port_info(), snd_seq_port_info_get_ump_is_midi1() + */ +void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t *info, int is_midi1) +{ + assert(info); + if (is_midi1) + info->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; + else + info->flags &= ~SNDRV_SEQ_PORT_FLG_IS_MIDI1; +} + /** * \brief create a sequencer port on the current client * \param seq sequencer handle -- 2.43.0

1 0

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 07 Aug '24

07 Aug '24

Hi, > Based on the inputs/suggestions from Tudor, i am planning to add a new > layer between the SPI-NOR and MTD layers to support stacked and parallel > configurations. This new layer will be part of the spi-nor and located in > mtd/spi-nor/ Will AMD submit to maintain this layer? What happens if the maintainer will leave AMD? TBH, personally, I don't like to maintain such a niche feature. I'd really like to see some use cases and performance reports for this, like actual boards (and no evaluation boards don't count). Why wouldn't someone just use an octal flash? And as already mentioned there is also mtdcat, which seems to duplicate some features? -michael

5 13

[PATCH v1] ALSA: ASoC/tas2781: fix wrong calibrated data order
by Shenghao Ding 07 Aug '24

07 Aug '24

From: Baojun Xu <baojun.xu(a)ti.com> Wrong calibration data order cause sound too low in some device. Fix wrong calibrated data order, add calibration data converssion by get_unaligned_be32() after reading from UEFI. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: - Change copyright date, and add new maintainer. - Add unaligned.h included for get_unaligned_be32(). - In tas2781_apply_calib(), change data address transfer directly to get data by get_unaligned_be32(), and send address to device. --- sound/pci/hda/tas2781_hda_i2c.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 49bd7097d..4dc3350d5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -2,10 +2,12 @@ // // TAS2781 HDA I2C driver // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Current maintainer: Baojun Xu <baojun.xu(a)ti.com> +#include <asm/unaligned.h> #include <linux/acpi.h> #include <linux/crc8.h> #include <linux/crc32.h> @@ -519,20 +521,21 @@ static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) static const unsigned char rgno_array[CALIB_MAX] = { 0x74, 0x0c, 0x14, 0x70, 0x7c, }; - unsigned char *data; - int i, j, rc; + int i, j, rc, data; + int offset = 0; for (i = 0; i < tas_priv->ndev; i++) { - data = tas_priv->cali_data.data + - i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; for (j = 0; j < CALIB_MAX; j++) { + data = get_unaligned_be32( + &tas_priv->cali_data.data[offset]); rc = tasdevice_dev_bulk_write(tas_priv, i, TASDEVICE_REG(0, page_array[j], rgno_array[j]), - &(data[4 * j]), 4); + (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); + offset += 4; } } } -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add Framework Laptop 13 (Intel Core Ultra) to quirks
by Dustin L. Howett 07 Aug '24

07 Aug '24

The Framework Laptop 13 (Intel Core Ultra) has an ALC285 that ships in a similar configuration to the ALC295 in previous models. It requires the same quirk for headset detection. --- Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- 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 1645d21d422f..480e82df7a4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10678,6 +10678,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. --- base-commit: de9c2c66ad8e787abec7c9d7eff4f8c3cdd28aed change-id: 20240606-alsa-hda-realtek-add-framework-laptop-13-intel-core-ultra-to-quirks-2af5081e2f2c Best regards, -- Dustin L. Howett <dustin(a)howett.net>

2 1

[PATCH v2] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest asoc/for-6.12 branch include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9ddc..1edcb8d6f6eea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2388,14 +2388,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2525,26 +2517,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", snd_pcm_direction_name(stream), fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2558,10 +2536,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2605,23 +2580,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 1

Re: [PATCH 0/2] ASoC: Extend wm_adsp so cs35l56 can suppress controls
by Mark Brown 06 Aug '24

06 Aug '24

On Mon, 05 Aug 2024 10:27:19 +0000, Simon Trimmer wrote: > This pair of patches extend wm_adsp to add a callback that can be used > to control whether ALSA controls are added and then tweak cs35l56 to use > it to suppress controls made from firmware coefficients. > > Simon Trimmer (2): > ASoC: wm_adsp: Add control_add callback and export > wm_adsp_control_add() > ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: wm_adsp: Add control_add callback and export wm_adsp_control_add() commit: 45b4acab4cac79503663f0a4be9eb3752db04d4b [2/2] ASoC: cs35l56: Stop creating ALSA controls for firmware coefficients commit: 2c3640b82213cf2beb7c1cc3cfce2ecf5349b0de All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

2 3

[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 122 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

2 3

[PATCH] ALSA: usb-audio: Add input gain and master output mixer controls for RME Babyface Pro
by Stefan Stistrup 06 Aug '24

06 Aug '24

Add missing input gain and master output mixer controls for RME Babyface Pro This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup <sstistrup(a)gmail.com> --- sound/usb/mixer_quirks.c | 162 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 161 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d771..26298e06b838 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,113 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + } else { + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + } + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2906,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2938,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2997,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) { + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + } else { + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + } + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, -- 2.45.2

2 2

[RFC PATCH 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Aug '24

06 Aug '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. Shengjiu Wang (6): ALSA: compress: add Sample Rate Converter codec support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_offload.h | 2 + include/uapi/sound/compress_params.h | 9 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 176 +++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 68 +++ sound/soc/fsl/fsl_asrc_m2m.c | 769 ++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 259 ++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 10 files changed, 1284 insertions(+), 8 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

5 23

[PATCH v1] ASoC: meson: axg-fifo: set option to use raw spinlock
by Arseniy Krasnov 06 Aug '24

06 Aug '24

Raw spinlock is needed here, because with enabled PREEMPT_RT, spinlock_t become preemptible, but this regmap lock could be acquired in IRQ handler. Found by lockdep: [ ] ============================= [ ] [ BUG: Invalid wait context ] [ ] 6.9.9-sdkernel #1 Tainted: G O [ ] ----------------------------- [ ] aplay/413 is trying to lock: [ ] ffff000003930018 (axg_fifo:356:(&axg_fifo_regmap_cfg)->lock){....}-{3:3},c [ ] other info that might help us debug this: [ ] context-{2:2} [ ] no locks held by aplay/413. [ ] stack backtrace: [ ] CPU: 0 PID: 413 Comm: aplay Tainted: G O 6.9.9-kernel #1 [ ] Hardware name: SberDevices SberBoom Mini (DT) [ ] Call trace: [ ] dump_backtrace+0x98/0xf0 [ ] show_stack+0x18/0x24 [ ] dump_stack_lvl+0x90/0xd0 [ ] dump_stack+0x18/0x24 [ ] __lock_acquire+0x9dc/0x1f10 [ ] lock_acquire.part.0+0xe8/0x228 [ ] lock_acquire+0x68/0x84 [ ] _raw_spin_lock_irqsave+0x60/0x88 [ ] regmap_lock_spinlock+0x18/0x2c [ ] regmap_read+0x3c/0x78 [ ] axg_fifo_pcm_irq_block+0x4c/0xc8 [ ] __handle_irq_event_percpu+0xa4/0x2f8 [ ] handle_irq_event+0x4c/0xbc [ ] handle_fasteoi_irq+0xa4/0x23c [ ] generic_handle_domain_irq+0x2c/0x44 [ ] gic_handle_irq+0x40/0xc4 [ ] call_on_irq_stack+0x24/0x4c [ ] do_interrupt_handler+0x80/0x84 [ ] el0_interrupt+0x5c/0x124 [ ] __el0_irq_handler_common+0x18/0x24 [ ] el0t_32_irq_handler+0x10/0x1c [ ] el0t_32_irq+0x194/0x198 Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-fifo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index ecb3eb7a9723d..a22298f74b35a 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -328,6 +328,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .val_bits = 32, .reg_stride = 4, .max_register = FIFO_CTRL2, + .use_raw_spinlock = true, }; int axg_fifo_probe(struct platform_device *pdev) -- 2.30.1

4 12

[PATCH] ASoC: codecs: lpass-macro: fix missing codec version
by Johan Hovold 05 Aug '24

05 Aug '24

Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- Cc: regressions(a)lists.linux.dev #regzbot introduced: 378918d59181 sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) -- 2.44.2

3 2

Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 05 Aug '24

05 Aug '24

Hi, > > You get twice more capacity based on that configuration. I can't answer the > > second question because not working with field. But both of that configurations > > are used by customers. Adding Neal if he wants to add something more to it. > > Just to add a comment as I work directly with our customers. The main reason > this support is important is for our older SoCs, zynq and zynqmp. > > Most of our customers are using QSPI flash as the first boot memory to get > from the boot ROM to u-boot. They then typically use other memories, such as > eMMC for the Linux kernel, OS and file system. Agreed and that's probably the most prominent use case for NOR flashes anyway. > The issue we have on the zynq and zynqmp SoCs is that the boot ROM (code that > cannot be changed) will not boot from an OSPI flash. It will only boot from a > QSPI flash. This is what is forcing many of our customers down the QSPI path. > Since many of these customers are interested in additional speed and memory > size, they then end up using a parallel or stacked configuration because they > cannot use an OSPI with zynq or zynqmp. Above you've said, the bootloader is stored on the NOR flash and the bulk storage is eMMC. So why do you need bigger NOR flashes (where even the biggest NOR flash isn't enough)? I also don't understand "the boot ROM will just boot from QSPI". First, you cannot connect an octal flash anyway, because you only have an QSPI controller, right? Secondly, normally the bootrom will (at least) boot the first stage using normal (single line) SPI commands. Is that not the case for zynq and zynqmp? > All of our newer SoCs can boot from OSPI. I agree with you that if someone > could choose OSPI for performance, they would, so I do not expect parallel or > stacked configurations with our newer SoCs. Ok, but then the argument with bigger flashes are void, because you are back to be bound to one OSPI flash. > I get why you see this configuration as a niche, but for us, it is a very > large niche because zynq and zynqmp are two of our most successful SoC > families. Fair enough. But please find a way to support it without butchering the whole core. -michael

1 0

Re: [PATCH v2] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 17:50:44 +0200, Stefan Binding wrote: > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls. > > These controls were never intended to be exposed to users but the > firmware doesn't mark them hidden, so make the driver ignore them. > > Any changes in the coefficients handled by these controls needs to > be matched to the individual system by SSID, which is already handled > using the tuning file, when firmware is loaded, so UCM should not be > setting these controls anyway. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Applied to for-linus branch now. Thanks. Takashi

1 0

Re: [PATCH v3] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Takashi Iwai 05 Aug '24

05 Aug '24

On Thu, 01 Aug 2024 16:31:39 +0200, Simon Trimmer wrote: > > A number of laptops have gone to market with old firmware versions that > export controls that have since been hidden, but we can't just install a > newer firmware because the firmware for each product is customized and > qualified by the OEM. The issue is that alsactl save and restore has no > idea what controls are good to persist which can lead to > misconfiguration. > > There is no reason that the UCM or user should need to interact with any > of the ALSA controls for the firmware coefficients so they can be > removed entirely, this also simplifies the driver. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > Changes in v3: > - Remove ability to add firmware coefficients as ALSA controls > entirely > > Changes in v2: > - v1 was accidentally the backport version for older kernels instead of > the one for v6.11 Applied now to for-linus branch. Thanks. Takashi

1 0

ucm2: sof-soundwire: add rt712 VA device
by GitHub pull_request - opened 05 Aug '24

05 Aug '24

alsa-project/alsa-ucm-conf pull request #435 was opened from shumingfan: Add support headphone/headset mic/speaker/dmic for rt712 VA Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

a52 plugin Unknown field card error
by GitHub issues - edited 04 Aug '24

04 Aug '24

alsa-project/alsa-lib issue #405 was edited from DocMAX: ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } a52:CARD=Adapter USB SPDIF Adapter Plugin to convert multichannel stream to A52 (AC3) bitstream [root@desktop ~]# speaker-test -c6 -Da52 speaker-test 1.2.12 Playback device is a52 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise ALSA lib pcm_rate.c:1701:(_snd_pcm_rate_open) Unknown field card Playback open error: -22,Invalid argument ``` how to fix this error? running arch linux all updated Issue URL : https://github.com/alsa-project/alsa-lib/issues/405 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

a52 plugin Unknown field card error
by GitHub issues - opened 04 Aug '24

04 Aug '24

alsa-project/alsa-lib issue #405 was opened from DocMAX: ``` a52:CARD=Adapter USB SPDIF Adapter Plugin to convert multichannel stream to A52 (AC3) bitstream [root@desktop ~]# speaker-test -c6 -Da52 speaker-test 1.2.12 Playback device is a52 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise ALSA lib pcm_rate.c:1701:(_snd_pcm_rate_open) Unknown field card Playback open error: -22,Invalid argument ``` how to fix this error? running arch linux all updated Issue URL : https://github.com/alsa-project/alsa-lib/issues/405 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 04 Aug '24

04 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change to Shenghao Ding and Baojun Xu, drop entries for the audio converter and add entries for both haptics drivers and middle- power audio amplifier drivers. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 8766f3e5e87e..34639103e463 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22567,37 +22567,30 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS AUDIO (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml -F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c F: include/sound/tas2*.h -F: include/sound/tlv320*.h +F: include/sound/tas5086.h F: include/sound/tpa6130a2-plat.h F: sound/pci/hda/tas2781_hda_i2c.c -F: sound/soc/codecs/pcm1681.c -F: sound/soc/codecs/pcm1789*.* -F: sound/soc/codecs/pcm179x*.* -F: sound/soc/codecs/pcm186x*.* -F: sound/soc/codecs/pcm3008.* -F: sound/soc/codecs/pcm3060*.* -F: sound/soc/codecs/pcm3168a*.* -F: sound/soc/codecs/pcm5102a.c -F: sound/soc/codecs/pcm512x*.* F: sound/soc/codecs/tas2*.* -F: sound/soc/codecs/tlv320*.* +F: sound/soc/codecs/tas5*.* F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS -- 2.34.1

1 0

[PATCH 0/4] ASoC: SOF: reshuffle and optimize structures
by Pierre-Louis Bossart 03 Aug '24

03 Aug '24

pahole reports a number of sub-optimal structure definitions with holes and cache-line alignment problems, this patchset optimizes a couple of structure frequently used. No functionality change, only re-ordering of structure members. Pierre-Louis Bossart (4): ASoC: SOF: sof-priv.h: optimize snd_sof_platform_stream_params ASoC: SOF: sof-priv.h: optimize snd_sof_mailbox ASoC: SOF: sof-priv.h: optimize snd_sof_ipc_msg ASoC: SOF: sof-audio.h: optimize snd_sof_pcm_stream_pipeline_list sound/soc/sof/sof-audio.h | 4 ++-- sound/soc/sof/sof-priv.h | 16 +++++++++------- 2 files changed, 11 insertions(+), 9 deletions(-) -- 2.43.0

2 5

[PATCH v4 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 03 Aug '24

03 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. All the dependencies required in the initial versions are now available in ASoc for-next branch. v4: topology name simplification for rt722 v3: add configuration with rt722 on Link3 v2: changed order of patches to fix git bisect error reported by Intel kbuild bot https://lore.kernel.org/oe-kbuild-all/202406131144.L6gW0I47-lkp@intel.com/ Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 145 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 ++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 ++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 290 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

2 6

[PATCH] ASoC: cs43130: Constify snd_soc_component_driver struct
by Christophe JAILLET 02 Aug '24

02 Aug '24

In order to constify `snd_soc_component_driver` struct, duplicate `soc_component_dev_cs43130` into a `soc_component_dev_cs43130_digital` and `soc_component_dev_cs43130_analog`. These 2 new structures share the same .dapm_widgets and .dapm_routes arrays but differ for .num_dapm_widgets and .num_dapm_routes. In the digital case, the last entries are not taken into account. Doing so has several advantages: - `snd_soc_component_driver` can be declared as const to move their declarations to read-only sections. - code in the probe is simpler. There is no need to concatenate some arrays to handle the "analog" case - this saves some memory because all_hp_widgets and analog_hp_routes can be removed. Before : ====== text data bss dec hex filename 53965 8265 4512 66742 104b6 sound/soc/codecs/cs43130.o After : ===== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. See discussion at [1]. [1]: https://lore.kernel.org/all/ZqNawRmAqBRLIoQq@opensource.cirrus.com/ --- sound/soc/codecs/cs43130.c | 73 +++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index be4037890fdb..cb4ca80f36d2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1415,7 +1415,7 @@ static const char * const bypass_mux_text[] = { static SOC_ENUM_SINGLE_DECL(bypass_enum, SND_SOC_NOPM, 0, bypass_mux_text); static const struct snd_kcontrol_new bypass_ctrl = SOC_DAPM_ENUM("Switch", bypass_enum); -static const struct snd_soc_dapm_widget digital_hp_widgets[] = { +static const struct snd_soc_dapm_widget hp_widgets[] = { SND_SOC_DAPM_MUX("Bypass Switch", SND_SOC_NOPM, 0, 0, &bypass_ctrl), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -1447,19 +1447,16 @@ static const struct snd_soc_dapm_widget digital_hp_widgets[] = { CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD)), -}; -static const struct snd_soc_dapm_widget analog_hp_widgets[] = { +/* Some devices have some extra analog widgets */ +#define NUM_ANALOG_WIDGETS 1 + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), }; -static struct snd_soc_dapm_widget all_hp_widgets[ - ARRAY_SIZE(digital_hp_widgets) + - ARRAY_SIZE(analog_hp_widgets)]; - -static const struct snd_soc_dapm_route digital_hp_routes[] = { +static const struct snd_soc_dapm_route hp_routes[] = { {"ASPIN PCM", NULL, "ASP PCM Playback"}, {"ASPIN DoP", NULL, "ASP DoP Playback"}, {"XSPIN DoP", NULL, "XSP DoP Playback"}, @@ -1472,15 +1469,12 @@ static const struct snd_soc_dapm_route digital_hp_routes[] = { {"Bypass Switch", "Internal", "HiFi DAC"}, {"HPOUTA", NULL, "Bypass Switch"}, {"HPOUTB", NULL, "Bypass Switch"}, -}; -static const struct snd_soc_dapm_route analog_hp_routes[] = { +/* Some devices have some extra analog routes */ +#define NUM_ANALOG_ROUTES 1 {"Bypass Switch", "Alternative", "Analog Playback"}, }; -static struct snd_soc_dapm_route all_hp_routes[ - ARRAY_SIZE(digital_hp_routes) + - ARRAY_SIZE(analog_hp_routes)]; static const unsigned int cs43130_asp_src_rates[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 @@ -2398,7 +2392,23 @@ static int cs43130_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_cs43130 = { +static const struct snd_soc_component_driver soc_component_dev_cs43130_digital = { + .probe = cs43130_probe, + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .set_sysclk = cs43130_component_set_sysclk, + .set_pll = cs43130_set_pll, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + /* Don't take into account the ending analog widgets and routes */ + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets) - NUM_ANALOG_WIDGETS, + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes) - NUM_ANALOG_ROUTES, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs43130_analog = { .probe = cs43130_probe, .controls = cs43130_snd_controls, .num_controls = ARRAY_SIZE(cs43130_snd_controls), @@ -2407,6 +2417,10 @@ static struct snd_soc_component_driver soc_component_dev_cs43130 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets), + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes), }; static const struct regmap_config cs43130_regmap = { @@ -2479,6 +2493,7 @@ static int cs43130_handle_device_data(struct cs43130_private *cs43130) static int cs43130_i2c_probe(struct i2c_client *client) { + const struct snd_soc_component_driver *component_driver; struct cs43130_private *cs43130; int ret; unsigned int reg; @@ -2596,39 +2611,15 @@ static int cs43130_i2c_probe(struct i2c_client *client) switch (cs43130->dev_id) { case CS43130_CHIP_ID: case CS43131_CHIP_ID: - memcpy(all_hp_widgets, digital_hp_widgets, - sizeof(digital_hp_widgets)); - memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), - analog_hp_widgets, sizeof(analog_hp_widgets)); - memcpy(all_hp_routes, digital_hp_routes, - sizeof(digital_hp_routes)); - memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), - analog_hp_routes, sizeof(analog_hp_routes)); - - soc_component_dev_cs43130.dapm_widgets = - all_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(all_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - all_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(all_hp_routes); + component_driver = &soc_component_dev_cs43130_analog; break; case CS43198_CHIP_ID: case CS4399_CHIP_ID: - soc_component_dev_cs43130.dapm_widgets = - digital_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(digital_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - digital_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(digital_hp_routes); + component_driver = &soc_component_dev_cs43130_digital; break; } - ret = devm_snd_soc_register_component(cs43130->dev, - &soc_component_dev_cs43130, + ret = devm_snd_soc_register_component(cs43130->dev, component_driver, cs43130_dai, ARRAY_SIZE(cs43130_dai)); if (ret < 0) { dev_err(cs43130->dev, -- 2.45.2

2 1

Soundblaster AE-5 100% Muted on alsamixer every boot!!
by GitHub issues - opened 02 Aug '24

02 Aug '24

alsa-project/alsa-lib issue #404 was opened from DavidRLTG: I am on Linux Mint Wilma (22), 64-bit. Obviously, I have a Soundblaster AE-5 Sound card The issue I am encountering is, well, every boot I have to enter Alsamixer and unmute every channel on my Soundblaster AE-5. alsactl store does nothing, and yes I ran it as superuser. Also, I'm on pipewire! I think Wilma comes with it preinstalled. sudo alsactl store is also temporary, the config is lost at every reboot, and doing sudo alsactl restore does nothing. Any ideas for a fix? Issue URL : https://github.com/alsa-project/alsa-lib/issues/404 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RESEND PATCH v2] ASoc: tas2781: Fix a compiling warning reported by robot kernel test due to adding tas2563_dvc_table
by Shenghao Ding 02 Aug '24

02 Aug '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Fixes: 75ed63a5ab5d ("ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Modify the submission subject v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..faa3e194f73b --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0X7F, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 00fd4d449ff3..d87263e43fdb 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

2 1

[PATCH -next] ASoC: fsl: lpc3xxx: Make some symbols static
by Yue Haibing 02 Aug '24

02 Aug '24

These symbols are not used outside of the files, make them static to fix sparse warnings: sound/soc/fsl/lpc3xxx-i2s.c:261:30: warning: symbol 'lpc3xxx_i2s_dai_ops' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-i2s.c:271:27: warning: symbol 'lpc3xxx_i2s_dai_driver' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-pcm.c:55:39: warning: symbol 'lpc3xxx_soc_platform_driver' was not declared. Should it be static? Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 ++-- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 62ef624d6dd4..bdb320bb3719 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -258,7 +258,7 @@ static int lpc3xxx_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { +static const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .probe = lpc3xxx_i2s_dai_probe, .startup = lpc3xxx_i2s_startup, .shutdown = lpc3xxx_i2s_shutdown, @@ -268,7 +268,7 @@ const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .set_fmt = lpc3xxx_i2s_set_dai_fmt, }; -struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { +static struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { .playback = { .channels_min = 1, .channels_max = 2, diff --git a/sound/soc/fsl/lpc3xxx-pcm.c b/sound/soc/fsl/lpc3xxx-pcm.c index c0d499b9b8ba..e6abaf63895a 100644 --- a/sound/soc/fsl/lpc3xxx-pcm.c +++ b/sound/soc/fsl/lpc3xxx-pcm.c @@ -52,7 +52,7 @@ static const struct snd_dmaengine_pcm_config lpc3xxx_dmaengine_pcm_config = { .prealloc_buffer_size = 128 * 1024, }; -const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { +static const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { .name = "lpc32xx-pcm", }; -- 2.34.1

2 1

[PATCH 00/12] ALSA/ASoC: use snd_pcm_direction_name()
by Kuninori Morimoto 02 Aug '24

02 Aug '24

Hi Tatasi, Mark Many drivers are selecting strings "playback" / "capture" by own handling, but we have snd_pcm_direction_name() function for it. This patch use it. One note is that snd_pcm_direction_name() will select "Playback" and "Capture", instead of "playback" / "capture". Almost all drivers are using it as dev_dbg() or dev_err() so no problem. But some other drivers are using it as other purpose. It might be issue (?). For example ASoC debugfs dir name will be changed by this patch. Kuninori Morimoto (12): ALSA: pci: pcxhr: use snd_pcm_direction_name() ALSA: pci: rme9652: use snd_pcm_direction_name() ALSA: trace: use snd_pcm_direction_name() ALSA: aloop: use snd_pcm_direction_name() ALSA: pcm_timer: use snd_pcm_direction_name() ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() include/trace/events/asoc.h | 3 ++- sound/core/pcm_timer.c | 3 +-- sound/drivers/aloop.c | 3 +-- sound/pci/pcxhr/pcxhr_mix22.c | 2 +- sound/pci/rme9652/hdspm.c | 6 ++---- sound/soc/fsl/lpc3xxx-i2s.c | 3 +-- sound/soc/soc-dapm.c | 3 +-- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- sound/soc/sof/intel/hda-stream.c | 4 +--- sound/soc/sof/pcm.c | 2 +- sound/soc/stm/stm32_i2s.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 16 ++++++++-------- 12 files changed, 36 insertions(+), 43 deletions(-) -- 2.43.0

3 14

[PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Vijendar Mukunda 02 Aug '24

02 Aug '24

This patch series majorly consists of below changes. - Rename structures, macros and codec helper names used in Intel SoundWire generic driver to make it generic. - Move Intel SoundWire driver common structures, macros and codec helper functions to common placeholder so that it can be used by other platform machine driver. - Refactor few SoundWire common codec helper functions. - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. This work started a couple of months ago to avoid duplication of code that wasn't really Intel-specific in the "sof_sdw" machine driver. The code went through multiple iterations, was tested for multiple weeks and a couple of build issues reported by the Intel kbuild bots were corrected. This is the initial version of SoundWire machine driver for AMD platforms. Additional code refactoring will be done in the next step on the AMD side. Link: https://github.com/thesofproject/linux/pull/5068 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (31): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver include/sound/soc-acpi.h | 2 + include/sound/soc_sdw_utils.h | 216 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++ sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 9 - sound/soc/intel/boards/sof_sdw.c | 1142 ++--------------- sound/soc/intel/boards/sof_sdw_common.h | 166 +-- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../soc_sdw_dmic.c} | 10 +- .../soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../soc_sdw_rt700.c} | 12 +- .../soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 32 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 990 ++++++++++++++ sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + 38 files changed, 2475 insertions(+), 1357 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (87%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c -- 2.34.1

4 34

common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #413 was edited from ujfalusi: Hi, The PR will introduce a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

common: pcm/split: add support up to 32 / 8 channels
by GitHub pull_request - edited 02 Aug '24

02 Aug '24

alsa-project/alsa-ucm-conf pull request #417 was edited from perexg: **DO NOT MERGE - after 1.2.12!** MOTU 828 has up to 32 hardware channels. Add support for this configuration to common/pcm/split.conf. Also, add support for up to 8 channels for the abstact devices (like 7.1 surround). Note that this change requires recent alsa-lib (Syntax 7). Link: https://github.com/alsa-project/alsa-ucm-conf/pull/416 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/417.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lag between microphone and speaker
by Guillermo Bernaldo de Quiros Maraver 02 Aug '24

02 Aug '24

Hi, good afternoon! First of all, I'm new to the mailing list so I'm not sure if this is the correct place to ask this question. I ask it here because I don't know a better place to ask it so, if this is not the correct place, please let me know. The thing is, I have a USB sound card plugged to a Beaglebone AI64. This works correctly (I mean, I can record my voice and I can play a wav file without problems. using the arecord and aplay tools) The problem comes with a software I'm writing which needs to read audio from the microphone, then do a process (which takes less than 20ms) and then, after that process, I have to write the audio to the speaker. The software, right now, consists of several files, but, the most important components of the software are Recorder and Player CPP classes which I show you at the end. The important part is the next: 1º I init the alsa context using the next calls: audioPlayer.OpenDevice("plughw:CARD=Device,DEV=0"); micRecorder.OpenDevice("plughw:CARD=Device,DEV=0"); 2º I'm using the sample rate of 16KHz, a sample size of 16bit and 1 channel. So, assuming I've done correctly, this should give me a total frame rate of 342 samples, then, if I multiply it by 2 bytes, I get 684 bytes of total size. 3º Finally, in the main, I do the next thing: while(1) { //bool keyPressed = id.IsKeyPressed(0); //printf("Key is pressed: %s\n", keyPressed ? "true" : "false"); const int8_t *pAudioBuffer = micRecorder.ReadAudio(); if(pAudioBuffer) { // Process a single frame of audio // Note: output_frame can be the same buffer as input_frame for in-place processing if(keyPressed) { // DO A PROCESS WHICH TAKES 19ms more or less } else { memcpy(buff_in,pAudioBuffer, buff_size); } // Opcionalmente, leemos/escribinos de/a fichero. //write(noisy_raw_fd, pAudioBuffer, micRecorder.GetBufferSize()); //write(clean_raw_fd, g_out_buffer, g_out_buffer_size); if(audioPlayer.WriteAudio((int8_t*)buff_in)<0) { return 1; } } //printf("Numero de frames escritos: %d del total de: %d\n", pcm, frames); } This works correctly, the problem is after some time I start to note that there is a difference of time between my speaking and my listening and I don't know why. I've enabled all traces in both the recorder and the player, but I don't see any error. Is just that there is a difference between my speaking and my listening. The difference is up to 10 seconds or more. Please, can anyone help me with this problem? PD: Please, be aware I'm just a software engineer, I'm not an expert in all technical stuff related to audio. I did this code based on some examples I found on internet which are: I used this code to implement the player class: https://gist.github.com/ghedo/963382/815c98d1ba0eda1b486eb9d80d9a91a81d9952… I used this code to implement the recorder class: https://gist.github.com/albanpeignier/104902 Here is the code // Player.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Player { public: Player(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Player(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); int WriteAudio(const int8_t *pAudioBuffer); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Player.cpp #include "Player.h" Player::Player(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Player::~Player() { CloseDevice(); } int Player::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } void Player::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); //free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } size_t Player::GetFrameSize() { return m_FrameBufferSize; } size_t Player::GetBufferSize() { return m_BufferSize; } int Player::WriteAudio(const int8_t *pAudioBuffer) { int pcm; if (pcm = snd_pcm_writei(m_AudioHandle, pAudioBuffer, FRAME_SIZE) == -EPIPE) { printf("XRUN.\n"); snd_pcm_prepare(m_AudioHandle); } else if (pcm < 0) { printf("ERROR. Can't write to PCM device. %s\n", snd_strerror(pcm)); return 0; } return 1; } int Player::InitAudioContext() { std::cout << "Iniciando Player" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "format width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(SND_PCM_FORMAT_S16_LE) / 8 * CHANNELS; return 0; } // Recorder.h #pragma once #include <iostream> #include <string> #include <alsa/asoundlib.h> #define SAMPLE_RATE 16000 #define FRAME_SIZE 342 #define CHANNELS 1 class Recorder { public: Recorder(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); ~Recorder(); int OpenDevice(const std::string &devname = "", int sampleRate = SAMPLE_RATE, int channels = CHANNELS); void CloseDevice(); size_t GetFrameSize(); size_t GetBufferSize(); const int8_t *ReadAudio(); private: int InitAudioContext(); private: int m_SampleRate; int m_Channels; std::string m_DevName; snd_pcm_t *m_AudioHandle; size_t m_FrameBufferSize; size_t m_BufferSize; int8_t *m_pAudioBuffer; }; // Recorder.cpp #include "Recorder.h" /* * Recuperacion ante errores de alsa. */ static int xrun_recovery(snd_pcm_t *handle, int err) { if (err == -EPIPE) { /* under-run */ printf("underrun!\n"); err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /* wait until the suspend flag is released */ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } Recorder::Recorder(const std::string &devname, int sampleRate, int channels): m_DevName{devname}, m_SampleRate{sampleRate}, m_Channels{channels}, m_AudioHandle{nullptr}, m_FrameBufferSize{FRAME_SIZE} { } Recorder::~Recorder() { CloseDevice(); } int Recorder::OpenDevice(const std::string &devname, int sampleRate, int channels) { if(!devname.empty()) m_DevName = devname; if(sampleRate != SAMPLE_RATE) m_SampleRate = sampleRate; if(channels != CHANNELS) m_Channels = channels; return InitAudioContext(); } size_t Recorder::GetFrameSize() { return m_FrameBufferSize; } /* * Tamaño total en bytes del buffer de memoria reservado */ size_t Recorder::GetBufferSize() { return m_BufferSize; } /* * Lee en el puntero de memoria resevado, las muestras de audio leidas de la capturadora de audio (microfono). */ const int8_t *Recorder::ReadAudio() { int err; if ((err = snd_pcm_readi(m_AudioHandle, m_pAudioBuffer, m_FrameBufferSize)) != m_FrameBufferSize) { // Si el error es de tipo PIPE o STRPIPE, tratamos de recuperar el dispositivo. if(xrun_recovery(m_AudioHandle, err) < 0) { std::cerr << "read from audio interface failed " << snd_strerror(err) << std::endl; return nullptr; } } return m_pAudioBuffer; } void Recorder::CloseDevice() { std::cout << "buffer freed" << std::endl; snd_pcm_close (m_AudioHandle); free(m_pAudioBuffer); std::cout << "audio interface closed" << std::endl; } int Recorder::InitAudioContext() { std::cout << "Iniciando Grabador" << std::endl; int i; int err; snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE; if ((err = snd_pcm_open (&m_AudioHandle, m_DevName.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) { std::cerr << "cannot open audio device " << m_DevName << " Error: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface opened" << std::endl; if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) { std::cerr << "cannot allocate hardware parameters structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params allocated" << std::endl; if ((err = snd_pcm_hw_params_any (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot initialize hardware parameter structure: " << snd_strerror(err) << std::endl; return (1); } std::cout << "hw_params initialized" << std::endl; if ((err = snd_pcm_hw_params_set_access (m_AudioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { std::cerr << "cannot set access type: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params access setted" << std::endl; if ((err = snd_pcm_hw_params_set_format (m_AudioHandle, hw_params, format)) < 0) { std::cerr << "cannot set sample format: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params format setted" << std::endl; if ((err = snd_pcm_hw_params_set_rate_near (m_AudioHandle, hw_params, (unsigned int *)&m_SampleRate, 0)) < 0) { std::cerr << "cannot set sample rate: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params rate setted" << std::endl; if ((err = snd_pcm_hw_params_set_channels (m_AudioHandle, hw_params, CHANNELS)) < 0) { std::cerr << "cannot set channel count: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params channels setted" << std::endl; if ((err = snd_pcm_hw_params (m_AudioHandle, hw_params)) < 0) { std::cerr << "cannot set parameters: " << snd_strerror (err) << std::endl; return (1); } std::cout << "hw_params setted" << std::endl; snd_pcm_hw_params_free (hw_params); std::cout << "hw_params freed" << std::endl; if ((err = snd_pcm_prepare (m_AudioHandle)) < 0) { std::cerr << "cannot prepare audio interface for use: " << snd_strerror (err) << std::endl; return (1); } std::cout << "audio interface prepared" << std::endl; std::cout << "formt width: " << snd_pcm_format_width(format) << std::endl; // Tamaño total en bytes del buffer de memoria reservado m_BufferSize = m_FrameBufferSize * snd_pcm_format_width(format) / 8 * CHANNELS; // Puntero al buffer de memoria reservado. m_pAudioBuffer = (int8_t *)malloc(m_BufferSize); return 0; }

1 0

[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 01 Aug '24

01 Aug '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 24ae1d4959be..1c6e035fd313 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static const struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.46.0.rc2.264.g509ed76dc8-goog

3 2

[PATCH] slimbus: generate MODULE_ALIAS() from MODULE_DEVICE_TABLE()
by Masahiro Yamada 01 Aug '24

01 Aug '24

Commit 9e663f4811c6 ("slimbus: core: add support to uevent") added the MODALIAS=slim:* uevent variable, but modpost does not generate the corresponding MODULE_ALIAS(). To support automatic module loading, slimbus drivers still need to manually add MODULE_ALIAS("slim:<manf_id>:<prod_code>:*"), as seen in sound/soc/codecs/wcd9335.c. To automate this, make modpost generate the proper MODULE_ALIAS() from MODULE_DEVICE_TABLE(slim, ). Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- scripts/mod/devicetable-offsets.c | 4 ++++ scripts/mod/file2alias.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index 518200813d4e..9c7b404defbd 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -153,6 +153,10 @@ int main(void) DEVID_FIELD(i3c_device_id, part_id); DEVID_FIELD(i3c_device_id, extra_info); + DEVID(slim_device_id); + DEVID_FIELD(slim_device_id, manf_id); + DEVID_FIELD(slim_device_id, prod_code); + DEVID(spi_device_id); DEVID_FIELD(spi_device_id, name); diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 5d1c61fa5a55..99dce93a4188 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -960,6 +960,16 @@ static int do_i3c_entry(const char *filename, void *symval, return 1; } +static int do_slim_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD(symval, slim_device_id, manf_id); + DEF_FIELD(symval, slim_device_id, prod_code); + + sprintf(alias, "slim:%x:%x:*", manf_id, prod_code); + + return 1; +} + /* Looks like: spi:S */ static int do_spi_entry(const char *filename, void *symval, char *alias) @@ -1555,6 +1565,7 @@ static const struct devtable devtable[] = { {"rpmsg", SIZE_rpmsg_device_id, do_rpmsg_entry}, {"i2c", SIZE_i2c_device_id, do_i2c_entry}, {"i3c", SIZE_i3c_device_id, do_i3c_entry}, + {"slim", SIZE_slim_device_id, do_slim_entry}, {"spi", SIZE_spi_device_id, do_spi_entry}, {"dmi", SIZE_dmi_system_id, do_dmi_entry}, {"platform", SIZE_platform_device_id, do_platform_entry}, -- 2.43.0

2 2

[PATCH 0/3] slimbus: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 01 Aug '24

01 Aug '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (3): slimbus: messaging: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ctrl: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ngd-ctrl: use 'time_left' variable with wait_for_completion_timeout() drivers/slimbus/messaging.c | 9 +++++---- drivers/slimbus/qcom-ctrl.c | 7 ++++--- drivers/slimbus/qcom-ngd-ctrl.c | 29 ++++++++++++++++------------- 3 files changed, 25 insertions(+), 20 deletions(-) -- 2.43.0

3 6

Re: [PATCH RESEND 00/31] AMD SOF based generic SoundWire machine driver
by Mukunda,Vijendar 01 Aug '24

01 Aug '24

On 01/08/24 16:27, Charles Keepax wrote: > On Thu, Aug 01, 2024 at 02:44:15PM +0530, Vijendar Mukunda wrote: >> This patch series majorly consists of below changes. >> - Rename structures, macros and codec helper names used in Intel >> SoundWire generic driver to make it generic. >> - Move Intel SoundWire driver common structures, macros and codec >> helper functions to common placeholder so that it can be used by >> other platform machine driver. >> - Refactor few SoundWire common codec helper functions. >> - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant. >> >> This work started a couple of months ago to avoid duplication of code >> that wasn't really Intel-specific in the "sof_sdw" machine driver. >> The code went through multiple iterations, was tested for multiple weeks >> and a couple of build issues reported by the Intel kbuild bots were >> corrected. >> >> This is the initial version of SoundWire machine driver for AMD >> platforms. Additional code refactoring will be done in the next step on >> the AMD side. >> >> Link: https://github.com/thesofproject/linux/pull/5068 >> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> >> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> >> > Would be good to also include the linux-sound mailing list on the > cover letter, if not the individuals given they are basically the > same on each patch. > > Also assuming your mail timed out again, you should be able to do > a separate git send-email for the later patches using the > --in-reply-to option to give the Message-ID of the cover letter > and they should append to the same chain. > > Thanks, > Charles Thanks, Charles, for your suggestion. I have shared the rest of the Patch series. - Vijendar

1 0

[PATCH 00/31] AMD SOF based generic SoundWire machine driver
by Vijendar Mukunda 01 Aug '24

01 Aug '24

This patch series majorly consists of below changes. - Rename structures, macros and codec helper names used in Intel SoundWire generic driver to make it generic. - Move Intel SoundWire driver common structures, macros and codec helper functions to common placeholder so that it can be used by other platform machine driver. - Refactor few SoundWire common codec helper functions. - AMD SOF based generic SoundWire machine driver for ACP 6.3 variant Link: https://github.com/thesofproject/linux/pull/5068 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Vijendar Mukunda (31): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver include/sound/soc-acpi.h | 2 + include/sound/soc_sdw_utils.h | 216 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++ sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 9 - sound/soc/intel/boards/sof_sdw.c | 1142 ++--------------- sound/soc/intel/boards/sof_sdw_common.h | 166 +-- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../soc_sdw_dmic.c} | 10 +- .../soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../soc_sdw_rt700.c} | 12 +- .../soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 32 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 990 ++++++++++++++ sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp63.c | 1 + 38 files changed, 2475 insertions(+), 1357 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (87%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c -- 2.34.1

3 24

gitcompile: restore ACLOCAL_FLAGS ability to use ../alsa-lib/utils/
by GitHub pull_request - edited 01 Aug '24

01 Aug '24

alsa-project/alsa-utils pull request #272 was edited from marc-hb: Fixes cryptic "error macro 'AM_PATH_ALSA' not found in library configure.ac:22" when attempting a local install as described at https://thesofproject.github.io/latest/getting_started/build-guide/build-fr… Fixes giant commit f9e6010d5ec7 ("topology: plugins - add Intel nhlt encoder plugin") which added a second invocation of `aclocal` in 2021 but forgot the extra `$alsa_m4_flags`. To avoid duplication, do not fix by adding the missing `$alsa_m4_flags` to the second aclocal invocation. Instead, drop that variable and append `-I ../alsa-lib/utils` directly to $ACLOCAL_FLAGS Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/272 Patch URL : https://github.com/alsa-project/alsa-utils/pull/272.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

gitcompile: restore ACLOCAL_FLAGS ability to use ../alsa-lib/utils/
by GitHub pull_request - opened 01 Aug '24

01 Aug '24

alsa-project/alsa-utils pull request #272 was opened from marc-hb: Fixes cryptic "error macro 'AM_PATH_ALSA' not found in library configure.ac:22" when attempting a local install as described at https://thesofproject.github.io/latest/getting_started/build-guide/build-fr… Fixes giant commit f9e6010d5ec7 ("topology: plugins - add Intel nhlt encoder plugin") which added a second invocation of `aclocal` in 2021 but forgot the extra `$alsa_m4_flags`. To avoid duplication, do not fix by adding the missing `$alsa_m4_flags` to the second aclocal invocation. Instead, drop that variable and append `-I ../alsa-lib/utils` directly to $ACLOCAL_FLAGS Request URL : https://github.com/alsa-project/alsa-utils/pull/272 Patch URL : https://github.com/alsa-project/alsa-utils/pull/272.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 31 Jul '24

31 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- Changes in v2: - v1 was accidentally the backport version for older kernels instead of the one for v6.11 sound/pci/hda/cs35l56_hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..f2f623f713d1 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -613,7 +617,7 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - else + else if (expose_dsp_controls) add_dsp_controls_required = true; cs35l56->base.fw_patched = false; -- 2.43.0

1 1

[PATCH][LINUX][ALSA][USB AUDIO CLASS 1] wrong channel ids for surround
by Sylvain BERTRAND 31 Jul '24

31 Jul '24

Hi, In the USB audio class 1 specifications, page 33 (3.7.2.3 Audio Channel Cluster Format): Surround is spatially defined as [rear], hence left surround should be SNDRV_CHMAP_RL and not SND_CHMAP_SL (Side). Same for right surround channel. Right? I am not a mailing list subscriber, CC me please. regards, -- Sylvain BERTRAND --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -244,8 +244,8 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, SNDRV_CHMAP_FR, /* right front */ SNDRV_CHMAP_FC, /* center front */ SNDRV_CHMAP_LFE, /* LFE */ - SNDRV_CHMAP_SL, /* left surround */ - SNDRV_CHMAP_SR, /* right surround */ + SNDRV_CHMAP_RL, /* left surround */ + SNDRV_CHMAP_RR, /* right surround */ SNDRV_CHMAP_FLC, /* left of center */ SNDRV_CHMAP_FRC, /* right of center */ SNDRV_CHMAP_RC, /* surround */

2 3

[PATCH -next] ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate'
by Yue Haibing 31 Jul '24

31 Jul '24

The variable savedbitclkrate is assigned and never used, so can be removed. sound/soc/fsl/lpc3xxx-i2s.c:42:13: warning: variable ‘savedbitclkrate’ set but not used [-Wunused-but-set-variable] Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index af995ca081a3..62ef624d6dd4 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -39,7 +39,7 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 { u32 i2srate; u32 idxx, idyy; - u32 savedbitclkrate, diff, trate, baseclk; + u32 diff, trate, baseclk; /* Adjust rate for sample size (bits) and 2 channels and offset for * divider in clock output @@ -53,14 +53,12 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 /* Find the best divider */ *clkx = *clky = 0; - savedbitclkrate = 0; diff = ~0; for (idxx = 1; idxx < 0xFF; idxx++) { for (idyy = 1; idyy < 0xFF; idyy++) { trate = (baseclk * idxx) / idyy; if (abs(trate - i2srate) < diff) { diff = abs(trate - i2srate); - savedbitclkrate = trate; *clkx = idxx; *clky = idyy; } -- 2.34.1

3 2

[PATCH] ASoC: rsnd: use pcm_dmaengine code
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd is implementing own DMAEngine code, but we can replace it with pcm_dmaengine code, because these are almost same. Let's use existing and stable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 17 --------- sound/soc/sh/rcar/dma.c | 75 ++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 2 +- 5 files changed, 9 insertions(+), 87 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7bddfd5e38d6c..426632996a0a3 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_RCAR depends on COMMON_CLK depends on OF select SND_SIMPLE_CARD_UTILS + select SND_DMAENGINE_PCM select REGMAP_MMIO help This option enables R-Car SRU/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6bc7027ed4dbf..233128c8ff145 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -660,23 +660,6 @@ static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) return rsnd_rdai_get(priv, dai->id); } -/* - * rsnd_soc_dai functions - */ -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io) -{ - struct snd_pcm_substream *substream = io->substream; - - /* - * this function should be called... - * - * - if rsnd_dai_pointer_update() returns true - * - without spin lock - */ - - snd_pcm_period_elapsed(substream); -} - static void rsnd_dai_stream_init(struct rsnd_dai_stream *io, struct snd_pcm_substream *substream) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7b499eee50806..2342bbb6fe92e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -7,6 +7,7 @@ #include <linux/delay.h> #include <linux/of_dma.h> +#include <sound/dmaengine_pcm.h> #include "rsnd.h" /* @@ -22,8 +23,6 @@ struct rsnd_dmaen { struct dma_chan *chan; - dma_cookie_t cookie; - unsigned int dma_len; }; struct rsnd_dmapp { @@ -66,20 +65,6 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -static void __rsnd_dmaen_complete(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - if (rsnd_io_is_working(io)) - rsnd_dai_period_elapsed(io); -} - -static void rsnd_dmaen_complete(void *data) -{ - struct rsnd_mod *mod = data; - - rsnd_mod_interrupt(mod, __rsnd_dmaen_complete); -} - static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_dai_stream *io, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) @@ -98,13 +83,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - - if (dmaen->chan) - dmaengine_terminate_async(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_STOP); } static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, @@ -120,7 +99,7 @@ static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, * Let's call it under prepare */ if (dmaen->chan) - dma_release_channel(dmaen->chan); + snd_dmaengine_pcm_close_release_chan(io->substream); dmaen->chan = NULL; @@ -153,7 +132,7 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, return -EIO; } - return 0; + return snd_dmaengine_pcm_open(io->substream, dmaen->chan); } static int rsnd_dmaen_start(struct rsnd_mod *mod, @@ -162,12 +141,9 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - int is_play = rsnd_io_is_play(io); int ret; /* @@ -195,7 +171,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, } } - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.direction = snd_pcm_substream_to_dma_direction(io->substream); cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = buswidth; @@ -209,32 +185,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return -EIO; - } - - desc->callback = rsnd_dmaen_complete; - desc->callback_param = rsnd_mod_get(dma); - - dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); - - dmaen->cookie = dmaengine_submit(desc); - if (dmaen->cookie < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return -EIO; - } - - dma_async_issue_pending(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_START); } struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, @@ -307,19 +258,7 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct dma_tx_state state; - enum dma_status status; - unsigned int pos = 0; - - status = dmaengine_tx_status(dmaen->chan, dmaen->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - if (state.residue > 0 && state.residue <= dmaen->dma_len) - pos = dmaen->dma_len - state.residue; - } - *pointer = bytes_to_frames(runtime, pos); + *pointer = snd_dmaengine_pcm_pointer(io->substream); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..dba684e4334b1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -576,7 +576,6 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, #define rsnd_rdai_width_get(rdai) \ rsnd_rdai_width_ctrl(rdai, 0) int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8d2a86383ae01..b3d4e8ae07eff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -706,7 +706,7 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, spin_unlock(&priv->lock); if (elapsed) - rsnd_dai_period_elapsed(io); + snd_pcm_period_elapsed(io->substream); if (stop) snd_pcm_stop_xrun(io->substream); -- 2.43.0

2 1

[PATCH] ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd_mod_confirm_ssi() confirms mod sanity, it should always be confirmed, not only when DEBUG. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index afd69c6eb6544..0f190abf00e75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -262,7 +262,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; - rsnd_mod_confirm_src(src_mod); + rsnd_mod_make_sure(src_mod, RSND_MOD_SRC); rsnd_adg_get_timesel_ratio(priv, io, in_rate, out_rate, @@ -291,7 +291,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int shift = (id % 4) * 8; u32 mask = 0xFF << shift; - rsnd_mod_confirm_ssi(ssi_mod); + rsnd_mod_make_sure(ssi_mod, RSND_MOD_SSI); val = val << shift; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..8cf5d9001f437 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -871,15 +871,6 @@ void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); -#ifdef DEBUG -#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) -#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) -#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -#else -#define rsnd_mod_confirm_ssi(mssi) -#define rsnd_mod_confirm_src(msrc) -#define rsnd_mod_confirm_dvc(mdvc) -#endif /* * If you don't need interrupt status debug message, -- 2.43.0

2 1

[PATCH 0/2] ASoC: fsl_micfil: Check the difference for i.MX8 and i.MX9
by Shengjiu Wang 31 Jul '24

31 Jul '24

There are some register difference for i.MX8 and i.MX9 REG_MICFIL_FIFO_CTRL definition is updated. REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID, REG_MICFIL_PARAM are added from i.MX9 Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms sound/soc/fsl/fsl_micfil.c | 20 +++++++++++++++----- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 16 insertions(+), 6 deletions(-) -- 2.34.1

3 4

Re: [PATCH v1] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Jaroslav Kysela 31 Jul '24

31 Jul '24

On 30. 07. 24 16:37, Stefan Binding wrote: > Add a kernel parameter to allow coefficients to be exposed as ALSA controls. > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls by default. > > However, nothing in userspace currently uses them, and is unlikely to > do so in the future, therefore we don't need to create ASLA controls > for them. > > These controls can be useful for debug, so we can add a kernel > parameter to re-enable them if necessary. > > Disabling these controls would prevent userspace from trying to read > these controls when the CS35L41 is hibernating, which ordinarily > would result in an error message. This is probably not a right argument to add this code. The codec should be powered up when those controls are accessed or those controls should be cached by the driver. Although the controls have not been used yet, exposing them in this way is not ideal. Could you fix the driver (no I/O errors)? Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 3

[PATCH alsa-lib 0/4] Add API helper functions for creating UMP Endpoint and Blocks
by Takashi Iwai 31 Jul '24

31 Jul '24

Hi, this is a patch set to alsa-lib to add the new API functions for creating a virtual UMP Endpoint and UMP Blocks arbitrarily from a user-space, that is, to create a virtual UMP device. An example program is included in test directory. Takashi === Takashi Iwai (4): ump: Add missing *_set variants for snd_ump_endpoint_info and snd_ump_block_info seq: Add API helper functions for creating UMP Endpoint and Blocks test: Add an example program to create a virtual UMP Endpoint Add test/seq-ump-example to .gitignore .gitignore | 1 + include/seqmid.h | 7 + include/ump.h | 36 ++++- src/rawmidi/ump.c | 304 +++++++++++++++++++++++++++++++++++++++-- src/seq/seq.c | 6 +- src/seq/seq_local.h | 4 + src/seq/seqmid.c | 249 +++++++++++++++++++++++++++++++++ test/Makefile.am | 3 +- test/seq-ump-example.c | 187 +++++++++++++++++++++++++ 9 files changed, 783 insertions(+), 14 deletions(-) create mode 100644 test/seq-ump-example.c -- 2.43.0

2 8

[PATCH v4 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 31 Jul '24

31 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

3 4

Bug in tlv320adc3101 driver
by Terry Sanders 31 Jul '24

31 Jul '24

Hi, There is a bug in the code for the tlv320adc3101 driver when setting up the D value for PLL. The D value is set via I2C using page 0 registers 7 (MSB) and 8 (LSB). The data sheet has the following information for the registers: "Page 0 / Register 7 will be updated when Page 0 / Register 8 is written immediately after Page 0 / Register 7 is written" and: "Page 0 / Register 8 must be written immediately after writing to Page 0 / Register 7" The function that implements this is adc3xxx_setup_pll() in file soc/codecs/tlv320adc3xxx.c line 1172 - 1189. The function is shown below: static void adc3xxx_setup_pll(struct snd_soc_component *component, int div_entry) { int i = div_entry; /* P & R values */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); /* J value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); /* D value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); } In the function the LSB (register 8) is written first followed by the MSB (register 7). This is the wrong way around and should be swapped so that the MSB is written first. Regards Terry Sanders

1 0

[PATCH v1] ASoc: tas2781: Move tas2563_dvc_table into a separate Header file
by Shenghao Ding 30 Jul '24

30 Jul '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..cf6eedb98ad2 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

3 4

[PATCH v1] ASoc: tas2781: Add TAS2563 into the Header
by Shenghao Ding 30 Jul '24

30 Jul '24

Add TAS2563 into the Header in case of misunderstanding and add channel No information for error debug in tasdevice_dev_read. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header - Add channel no into the log in tasdevice_dev_read for error debug --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } -- 2.34.1

2 1

[PATCH v3 0/3] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 30 Jul '24

30 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (3): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" ALSA: firewire-lib: amdtp-stream work queue inline description sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

1 0

[PATCH] ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding
by Krzysztof Kozlowski 30 Jul '24

30 Jul '24

The APQ8016 SBC and MSM8916 QDSP6 sound cards are a bit different from others: they have additional IO muxing address space and pin control. Move them to separate schema, so the original qcom,sm8250.yaml will be easier to manage. New schema is going to grow for other platforms having more of IO muxing address spaces. Cc: Adam Skladowski <a39.skl(a)gmail.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++++++++++++++++++ .../bindings/sound/qcom,sm8250.yaml | 137 ------------ 2 files changed, 205 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000000..6ad451549036 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index c9076dcd44c1..1d3acdc0c733 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -27,9 +27,7 @@ properties: - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - - qcom,apq8016-sbc-sndcard - qcom,apq8096-sndcard - - qcom,msm8916-qdsp6-sndcard - qcom,qcm6490-idp-sndcard - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard @@ -58,18 +56,6 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User visible long sound card name - pin-switches: - description: List of widget names for which pin switches should be created. - $ref: /schemas/types.yaml#/definitions/string-array - - widgets: - description: User specified audio sound widgets. - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - - # Only valid for some compatibles (see allOf if below) - reg: true - reg-names: true - patternProperties: ".*-dai-link$": description: @@ -122,34 +108,6 @@ required: - compatible - model -allOf: - - if: - properties: - compatible: - contains: - enum: - - qcom,apq8016-sbc-sndcard - - qcom,msm8916-qdsp6-sndcard - then: - properties: - reg: - items: - - description: Microphone I/O mux register address - - description: Speaker I/O mux register address - reg-names: - items: - - const: mic-iomux - - const: spkr-iomux - required: - - compatible - - model - - reg - - reg-names - else: - properties: - reg: false - reg-names: false - additionalProperties: false examples: @@ -231,98 +189,3 @@ examples: }; }; }; - - - | - #include <dt-bindings/sound/qcom,lpass.h> - sound@7702000 { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "DB410c"; - audio-routing = - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1"; - - pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; - pinctrl-names = "default", "sleep"; - - quaternary-dai-link { - link-name = "ADV7533"; - cpu { - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; - - primary-dai-link { - link-name = "WCD"; - cpu { - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - tertiary-dai-link { - link-name = "WCD-Capture"; - cpu { - sound-dai = <&lpass MI2S_TERTIARY>; - }; - codec { - sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; - }; - }; - }; - - - | - #include <dt-bindings/sound/qcom,q6afe.h> - #include <dt-bindings/sound/qcom,q6asm.h> - sound@7702000 { - compatible = "qcom,msm8916-qdsp6-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "msm8916"; - widgets = - "Speaker", "Speaker", - "Headphone", "Headphones"; - pin-switches = "Speaker"; - audio-routing = - "Speaker", "Speaker Amp OUT", - "Speaker Amp IN", "HPH_R", - "Headphones", "HPH_L", - "Headphones", "HPH_R", - "AMIC1", "MIC BIAS Internal1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS Internal3"; - aux-devs = <&speaker_amp>; - - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&cdc_pdm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - primary-dai-link { - link-name = "Primary MI2S"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - platform { - sound-dai = <&q6routing>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - }; -- 2.43.0

6 5

[PATCH 0/6] ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports mask
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Incorrect mask of Soundwire ports - one bit too long/big - was passed. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Not adding Cc-stable because I don't think the issue can lead to real out-of-bounds read. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask sound/soc/codecs/wcd937x-sdw.c | 4 ++-- sound/soc/codecs/wcd938x-sdw.c | 4 ++-- sound/soc/codecs/wcd939x-sdw.c | 4 ++-- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- 6 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240726-asoc-wcd-wsa-swr-ports-genmask-868239f9682f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 8

[PATCH] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 29 Jul '24

29 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..1494383b22c9 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -758,6 +762,9 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas cs35l56_hda_create_controls(cs35l56); + if (expose_dsp_controls) + cs35l56_hda_add_dsp_controls(cs35l56); + #if IS_ENABLED(CONFIG_SND_DEBUG) cs35l56->debugfs_root = debugfs_create_dir(dev_name(cs35l56->base.dev), sound_debugfs_root); cs_dsp_init_debugfs(&cs35l56->cs_dsp, cs35l56->debugfs_root); -- 2.43.0

2 2

ASUS RoG Strix Z790-F: SPDIF (PCM) has wrong channels | how to get passthrough working?
by GitHub issues - opened 29 Jul '24

29 Jul '24

alsa-project/alsa-ucm-conf issue #434 was opened from Nightwing0815: Hi there, finally, [after a long journey,](https://github.com/alsa-project/alsa-ucm-conf/issues/350) yesterday I did an update from linuxmint to newest version 22, based on ubuntu 24.04 (I'll attach alsa-info and a system info). Now it's no big problem any more, to get SPDIF working: I just selected the right device and muted all others in alsamixer, except the pcm signal. But hey, there's a new problem: With a speaker-test -c6, I'm not able to hear my rear channels and the lfe channel, where they should be. Rear coming on front speakers... So, what do I have to do? Further, I'm asking for help to configure a pcm signal passthru for dts and (e)ac3 soundfiles. My teufel receiver recognices only a 5.1 stereo source, not an DD5.1 or DTS signal... [alsa-info.txt](https://github.com/user-attachments/files/16417700/alsa-info… [sysinfo.txt](https://github.com/user-attachments/files/16417724/sysinfo.txt) Thanks in advance for getting this problems solved, ToM Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/434 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/7] ASoC: codecs: wsa88xx and wcd93xx: Soundwire port non-functional cleanup
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups to make the code more robust or readable. No functional impact (compiled objects stay the same). Best regards, Krzysztof --- Krzysztof Kozlowski (7): ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- sound/soc/codecs/wcd938x.c | 18 ------------------ sound/soc/codecs/wcd938x.h | 4 ++-- sound/soc/codecs/wcd939x.h | 6 +++--- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 7 files changed, 88 insertions(+), 92 deletions(-) --- base-commit: 66df68b9536e7465aec703733ea91ca39b787ed8 change-id: 20240725-asoc-wsa88xx-port-arrays-c58c96cd888e Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH v2] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 29 Jul '24

29 Jul '24

This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Changes in v2: - Update changelog - Link to v1: https://lore.kernel.org/r/20240721-sti-audio-fix-v1-1-a8b1ecf61cb4@free.fr --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

1 0

[RESEND] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 29 Jul '24

29 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

1 0

[PATCH] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8351f0974e6a..4158657914fb 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2296,36 +2296,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); -- 2.43.0

2 1

[PATCH] ASoC: cs42l42: Convert comma to semicolon
by Chen Ni 29 Jul '24

29 Jul '24

Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; -- 2.25.1

4 3

[PATCH 0/4] ASoC: codecs: wsa88xx: Few cleanups
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups around wsa88xx codecs. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant sound/soc/codecs/wsa881x.c | 2 -- sound/soc/codecs/wsa883x.c | 33 ++++++++++++++++++--------------- sound/soc/codecs/wsa884x.c | 5 ++--- 3 files changed, 20 insertions(+), 20 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240710-asoc-wsa88xx-version-dd0039265245 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 5

Re: [PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by edmund.raile 29 Jul '24

29 Jul '24

> Thank you for your sending the revised patches, it looks better than the > previous one. However, I have an additional request. Allright, patch v3 it is. > [1] https://git-scm.com/docs/git-revert Should have known git has something like that, how handy! > $ git revert -s b5b519965c4c Yes, 5b5 can be removed via revert, but what is the difference in effect? Just time saving? > $ git revert -s 7ba5ca32fe6e This one I'd like to ask you about: The original inline comment in amdtp-stream.c amdtp_domain_stream_pcm_pointer() ``` // This function is called in software IRQ context of // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. // Therefore, no need to flush the queue in buffer furthermore. // // When the process context reach here, some packets will be // already queued in the buffer. These packets should be handled // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above ``` (let's call the above v1) was replaced with ``` // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. ``` on occasion of 7ba5ca32fe6e (let's call this v2). I sought to replace it with ``` // use wq to prevent deadlock between process context spin_lock // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and // softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() // in snd_pcm_period_elapsed() ``` to prevent this issue from occurring again (let's call this v3). Should I include v1, v3 or a combination of v1 and v3 in my next patch? > Just for safe, it is preferable to execute 'scripts/checkpatch.pl' in > kernel tree to check the patchset generated by send-email subcommand[3]. Absolutely should have done so, sorry. Thank you for your patience and guidance, Edmund Raile.

2 1

[PATCH AUTOSEL 5.15 09/13] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 089b6c7994f9a..cfa0c3eaffea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 10/15] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cf501baf6f143..9cd85ab19c553 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -387,6 +387,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/15] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d03de37e3578c..cf501baf6f143 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -267,6 +267,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 11/17] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index dc144cd7e0e3b..db1dcb9d70466 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -425,6 +425,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 10/17] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5980fce817976..dc144cd7e0e3b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -286,6 +286,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 15/23] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 965d46968f95a..0b3b2c88d31ed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -406,6 +406,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 14/23] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a878b6b3d1948..965d46968f95a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -277,6 +277,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 02/23] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 28 Jul '24

28 Jul '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e41b0d95e0ff7..a878b6b3d1948 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -505,6 +505,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

1 0

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/4] MSM8953/MSM8976 ASoC support
by Adam Skladowski 28 Jul '24

28 Jul '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Adam Skladowski (1): ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard Vladimir Lypak (3): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add cajon and cajon v2 support ASoC: qcom: apq8016_sbc: Add support for msm8953/msm8976 SoC .../sound/qcom,apq8016-sbc-sndcard.yaml | 51 +++++++++-- sound/soc/codecs/msm8916-wcd-analog.c | 63 +++++++++++++- sound/soc/qcom/apq8016_sbc.c | 84 ++++++++++++++++++- 3 files changed, 186 insertions(+), 12 deletions(-) -- 2.45.2

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[PATCH v23 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 26 Jul '24

26 Jul '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Wesley Cheng (32): ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/xhci-plat.c | 2 + .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 188 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 5705 insertions(+), 29 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

4 59

[PATCH 0/2] {ASoC,media}: constify snd_soc_component_driver struct
by Javier Carrasco 26 Jul '24

26 Jul '24

The `snd_soc_component_driver` struct is never modified after its declaration, and its only direct user `devm_snd_soc_register_component()` expects a constant value anyway. Declare `snd_soc_component_driver` as const to move their declarations to read-only sections. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

3 6

[GIT PULL] ASoC fixes for v6.11-merge-window
by Mark Brown 25 Jul '24

25 Jul '24

The following changes since commit c51cba4755609ad97ba97713210c16f043c73224: Fix the unbalanced pm_runtime_enable in wcd937x-sdw (2024-07-12 17:38:18 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-merge-window for you to fetch changes up to ab53dfdcdd1ec8df8729890aefa5b0e3c900afbb: ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components (2024-07-25 14:20:30 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A selection of routine fixes and quirks that came in since the merge window. The fsl-asoc-card change is a fix for systems with multiple cards where updating templates in place leaks data from one card to another. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: tegra: select CONFIG_SND_SIMPLE_CARD_UTILS Curtis Malainey (1): ASoC: Intel: Fix RT5650 SSP lookup Dan Carpenter (1): ASoC: TAS2781: Fix tasdev_load_calibrated_data() Daniel Baluta (1): ASoC: SOF: imx8m: Fix DSP control regmap retrieval Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Only handle dai_config with HW_PARAMS for ChainDMA ASoC: SOF: ipc4-topology: Preserve the DMA Link ID for ChainDMA on unprepare Pierre-Louis Bossart (2): ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable Shengjiu Wang (1): ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components Takashi Iwai (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E16 Gen 2 Venkata Prasad Potturu (1): ASoC: sof: amd: fix for firmware reload failure in Vangogh platform sound/soc/amd/yc/acp6x-mach.c | 7 ++++ sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++-------- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 9 +++++ sound/soc/intel/common/soc-intel-quirks.h | 2 +- sound/soc/sof/amd/pci-vangogh.c | 1 - sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/hda-loader.c | 20 ++++++---- sound/soc/sof/intel/hda.c | 17 ++++---- sound/soc/sof/ipc4-topology.c | 18 +++++++-- sound/soc/tegra/Kconfig | 1 + 11 files changed, 85 insertions(+), 40 deletions(-)

2 1

[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 25 Jul '24

25 Jul '24

Bisection revealed that the bitcrushing distortion with RME FireFace 800 was caused by 1d717123bb1a7555 ("ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning"). Reverting this commit yields restoration of clear audio output. I will send in a patch reverting this commit for now, soonTM. #regzbot introduced: 1d717123bb1a7555

4 6

[PATCH] ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components
by Shengjiu Wang 25 Jul '24

25 Jul '24

The static snd_soc_dai_link_components cause conflict for multiple instances of this generic driver. For example, when there is wm8962 and SPDIF case enabled together, the contaminated snd_soc_dai_link_components will cause another device probe fail. Fixes: 6d174cc4f224 ("ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++++++++++------------- 1 file changed, 29 insertions(+), 17 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 82df887b3af5..f6c3aeff0d8e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -306,27 +306,12 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_fe, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_be, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); - static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", .stream_name = "HiFi", .ops = &fsl_asoc_card_ops, - SND_SOC_DAILINK_REG(hifi), }, /* DPCM Link between Front-End and Back-End (Optional) */ { @@ -335,7 +320,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .dynamic = 1, - SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-BE", @@ -345,7 +329,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(hifi_be), }, }; @@ -637,6 +620,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev[2] = { NULL, NULL }; + struct snd_soc_dai_link_component *dlc; const char *codec_dai_name[2]; const char *codec_dev_name[2]; u32 asrc_fmt = 0; @@ -717,7 +701,35 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + /* + * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms + * "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms + * "DPCM Link Back-End": 1 cpus, 2 codecs + * totally 10 components + */ + dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + ret = -ENOMEM; + goto asrc_fail; + } + + priv->dai_link[0].cpus = &dlc[0]; + priv->dai_link[0].num_cpus = 1; + priv->dai_link[0].codecs = &dlc[1]; priv->dai_link[0].num_codecs = 1; + priv->dai_link[0].platforms = &dlc[3]; + priv->dai_link[0].num_platforms = 1; + + priv->dai_link[1].cpus = &dlc[4]; + priv->dai_link[1].num_cpus = 1; + priv->dai_link[1].codecs = &dlc[5]; + priv->dai_link[1].num_codecs = 0; /* dummy */ + priv->dai_link[1].platforms = &dlc[6]; + priv->dai_link[1].num_platforms = 1; + + priv->dai_link[2].cpus = &dlc[7]; + priv->dai_link[2].num_cpus = 1; + priv->dai_link[2].codecs = &dlc[8]; priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; -- 2.34.1

2 1

[PATCH v5 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 25 Jul '24

25 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 124 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 593 insertions(+), 73 deletions(-) base-commit: 472438c7e0e2261c6737a8321f46ef176eef1c8f -- 2.25.1

3 3

[RFC v2 00/xx] ALSA: ALSA: add snd_pcm_is_playback/capture()
by Kuninori Morimoto 25 Jul '24

25 Jul '24

Hi Iwai-san, Mark Cc Sakamoto-san, Amadeusz, Pierre-Louis It seems we can use _Generic() more simply, v2 RFC is using it. I'm not 100% sure but unfortunately we can't use bit-field with _Generic() in gcc (clang seems possible to handle it ?). pci/ac97 is the only user of bit-field direction. To avoid posting patch-bomb, I will post main patch and some sample patches as RFC. Please review it. Link: https://lore.kernel.org/r/87zfqel1g7.wl-kuninori.morimoto.gx@renesas.com Thank you for your help !! Best regards --- Kuninori Morimoto

4 9

[kernel][usb audio class 1] surround channels mapped to wrong alsa channel ids
by GitHub issues - opened 25 Jul '24

25 Jul '24

alsa-project/alsa-lib issue #403 was opened from sylware: In $linux_src/sound/usb/stream.c, function "convert_chmap", in the "uac1_maps" array, the "left surround" should be SNDRV_CHMAP_RL instead of SNDRV_CHMAP_SL, and the "right surround" should be SNDRV_CHMAP_RR instead of SNDDRV_CHMAP_SR. Issue URL : https://github.com/alsa-project/alsa-lib/issues/403 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 24 Jul '24

24 Jul '24

Commit b5b519965c4c ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. With RME Fireface 800, this lead to a regression since Kernels 5.14.0, causing a deadlock with eventual system freeze under ALSA operation: ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci Restore the work queue to prevent deadlock between ALSA substream process context spin_lock of snd_pcm_stream_lock_irq() in snd_pcm_status64() and OHCI 1394 IT softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed(). to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Closes: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… Fixes: 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") Signed-off-by: Edmund Raile <edmund.raile(a)proton.me> --- This is the follow-up patch to the 5.14.0 regression I reported: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Takashi Sakamoto explained the issue in his response to the regression: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() This leads me to believe this isn't just an issue limited to the RME Fireface driver (snd_fireface), though I can not test the other devices. sound/firewire/amdtp-stream.c | 32 +++++++++++++++++++++----------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 22 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index d35d0a420ee0..77b99a2117f4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -77,6 +77,8 @@ // overrun. Actual device can skip more, then this module stops the packet streaming. #define IR_JUMBO_PAYLOAD_MAX_SKIP_CYCLES 5 +static void pcm_period_work(struct work_struct *work); + /** * amdtp_stream_init - initialize an AMDTP stream structure * @s: the AMDTP stream to initialize @@ -105,6 +107,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); + INIT_WORK(&s->period_work, pcm_period_work); s->packet_index = 0; init_waitqueue_head(&s->ready_wait); @@ -347,6 +350,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload); */ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) { + cancel_work_sync(&s->period_work); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; } @@ -611,19 +615,21 @@ static void update_pcm_pointers(struct amdtp_stream *s, // The program in user process should periodically check the status of intermediate // buffer associated to PCM substream to process PCM frames in the buffer, instead // of receiving notification of period elapsed by poll wait. - if (!pcm->runtime->no_period_wakeup) { - if (in_softirq()) { - // In software IRQ context for 1394 OHCI. - snd_pcm_period_elapsed(pcm); - } else { - // In process context of ALSA PCM application under acquired lock of - // PCM substream. - snd_pcm_period_elapsed_under_stream_lock(pcm); - } - } + if (!pcm->runtime->no_period_wakeup) + queue_work(system_highpri_wq, &s->period_work); } } +static void pcm_period_work(struct work_struct *work) +{ + struct amdtp_stream *s = container_of(work, struct amdtp_stream, + period_work); + struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); + + if (pcm) + snd_pcm_period_elapsed(pcm); +} + static int queue_packet(struct amdtp_stream *s, struct fw_iso_packet *params, bool sched_irq) { @@ -1854,7 +1860,10 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. - if (!in_softirq()) + // workqueue serves to prevent deadlock between process context spinlock + // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and softIRQ spinlock + // of snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() + if ((!in_softirq()) && (current_work() != &s->period_work)) fw_iso_context_flush_completions(irq_target->context); } @@ -1910,6 +1919,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s) return; } + cancel_work_sync(&s->period_work); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index a1ed2e80f91a..775db3fc4959 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -191,6 +191,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; + struct work_struct period_work; snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; unsigned int pcm_frame_multiplier; -- 2.45.2

2 1

[bug report] ASoC: Intel: cht_bsw_rt5645: Set card.components string
by Dan Carpenter 23 Jul '24

23 Jul '24

Hello Hans de Goede, Commit f87b4402163b ("ASoC: Intel: cht_bsw_rt5645: Set card.components string") from Nov 26, 2023 (linux-next), leads to the following Smatch static checker warning: sound/soc/intel/boards/cht_bsw_rt5645.c:587 snd_cht_mc_probe() error: we previously assumed 'adev' could be null (see line 581) sound/soc/intel/boards/cht_bsw_rt5645.c 570 /* set correct codec name */ 571 for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) 572 if (cht_dailink[i].codecs->name && 573 !strcmp(cht_dailink[i].codecs->name, 574 "i2c-10EC5645:00")) { 575 dai_index = i; 576 break; 577 } 578 579 /* fixup codec name based on HID */ 580 adev = acpi_dev_get_first_match_dev(mach->id, NULL, -1); 581 if (adev) { ^^^^ The old code assumes adev can be NULL 582 snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), 583 "i2c-%s", acpi_dev_name(adev)); 584 cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; 585 } 586 /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ --> 587 codec_dev = acpi_get_first_physical_node(adev); ^^^^ Unchecked dereference 588 acpi_dev_put(adev); 589 if (!codec_dev) 590 return -EPROBE_DEFER; 591 592 snd_soc_card_chtrt5645.components = rt5645_components(codec_dev); 593 snd_soc_card_chtrt5650.components = rt5645_components(codec_dev); 594 regards, dan carpenter

2 2

[PATCH for-6.10 0/2] ASoC: cs35l56: Set correct upper volume limit
by Richard Fitzgerald 23 Jul '24

23 Jul '24

Patch series to limit the upper range of the CS35L56 volume control to +12 dB. These commits were not marked 'Fixes' because they were thought to be only a cosmetic issue. The user could reduce the volume to a usable value. But for some complex audio topologies with SOF Audio DSP + CS42L43 + multiple CS35L56 it has turned out to be not obvious to the user what the problem actually is and what to do to fix it. As support for these topologies went into 6.10 we would like this series to be applied to 6.10. Richard Fitzgerald (2): ASoC: cs35l56: Use header defines for Speaker Volume control definition ASoC: cs35l56: Limit Speaker Volume to +12dB maximum include/sound/cs35l56.h | 2 +- sound/soc/codecs/cs35l56.c | 6 +++++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.39.2

2 3

[PATCH 0/4] MSM8953/MSM8976 ASoC support
by Adam Skladowski 23 Jul '24

23 Jul '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. Adam Skladowski (1): ASoC: dt-bindings: qcom,sm8250: Add msm8953/msm8976-qdsp6-sndcard Vladimir Lypak (3): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add cajon and cajon v2 support ASoC: qcom: apq8016_sbc: Add support for msm8953/msm8976 SoC .../bindings/sound/qcom,sm8250.yaml | 28 ++++++- sound/soc/codecs/msm8916-wcd-analog.c | 63 +++++++++++++- sound/soc/qcom/apq8016_sbc.c | 82 ++++++++++++++++++- 3 files changed, 164 insertions(+), 9 deletions(-) -- 2.45.2

4 8

[RFC 00/xx] ALSA: ALSA: add snd_stream_is_playback/capture()
by Kuninori Morimoto 23 Jul '24

23 Jul '24

Hi Iwai-san, Mark Current many drivers are using below code to know its direction. if (direction == SNDRV_PCM_STREAM_PLAYBACK) I think it should be handled by function. But is it acceptable idea ? Because it will be too many patch-set, I want to know it was acceptable idea or not before posting patch-bomb. I will post main patch, and sample driver patches. Thank you for your help !! Best regards --- Kuninori Morimoto

5 17

[PATCH] ASoc: TAS2781: Fix tasdev_load_calibrated_data()
by Dan Carpenter 22 Jul '24

22 Jul '24

This function has a reversed if statement so it's either a no-op or it leads to a NULL dereference. Fixes: b195acf5266d ("ASoC: tas2781: Fix wrong loading calibrated data sequence") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 63626b982d04..8f9a3ae7153e 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2162,7 +2162,7 @@ static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) return; cal = cal_fmw->calibrations; - if (cal) + if (!cal) return; load_calib_data(priv, &cal->dev_data); -- 2.43.0

2 1

[PATCH] ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable
by Pierre-Louis Bossart 22 Jul '24

22 Jul '24

the Intel kbuild bot reports a link failure when IOSF_MBI is built-in but the Merrifield driver is configured as a module. The soc-intel-quirks.h is included for Merrifield platforms, but IOSF_MBI is not selected for that platform. ld.lld: error: undefined symbol: iosf_mbi_read >>> referenced by atom.c >>> sound/soc/sof/intel/atom.o:(atom_machine_select) in archive vmlinux.a This patch forces the use of the fallback static inline when IOSF_MBI is not reachable. Fixes: 536cfd2f375d ("ASoC: Intel: use common helpers to detect CPUs") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202407160704.zpdhJ8da-lkp@intel.com/ Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/common/soc-intel-quirks.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index de4e550c5b34..42bd51456b94 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,7 +11,7 @@ #include <linux/platform_data/x86/soc.h> -#if IS_ENABLED(CONFIG_X86) +#if IS_REACHABLE(CONFIG_IOSF_MBI) #include <linux/dmi.h> #include <asm/iosf_mbi.h> -- 2.43.0

2 1

[PATCH alsa-utils 00/10] Cleanup seq/* stuff and extend aseqsend
by Takashi Iwai 22 Jul '24

22 Jul '24

Hi, this is a series of cleanup / fix patches for alsa-utils/seq/* stuff, including the extension of aseqsend to support UMP packet bytes. Takashi === Takashi Iwai (10): configure: Requires the latest ALSA-lib release 1.2.12 aplaymidi: Drop ifdef for UMP support aplaymidi: Allow to pass 0 to -u option, too aconnect: Drop superfluous ifdefs aseqdump: Drop ifdef for UMP support aseqdump: Check the -u option value properly configure: Drop unused conditionals aseqdump: Show SysEx prefix to UMP SysEx data dump aseqsend: Support UMP mode aseqsend: Support long options configure.ac | 13 +------- seq/aconnect/aconnect.c | 31 ------------------ seq/aplaymidi/aplaymidi.c | 32 +++--------------- seq/aseqdump/aseqdump.c | 25 +++----------- seq/aseqsend/aseqsend.1 | 69 +++++++++++++++++++++------------------ seq/aseqsend/aseqsend.c | 54 +++++++++++++++++++++++++++++- 6 files changed, 100 insertions(+), 124 deletions(-) -- 2.43.0

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[PATCH v3] ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
by wangdicheng 22 Jul '24

22 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> Sonix HD USB Camera does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. (snip) [1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd [1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00 [1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [1.984127] usb 3-3: Product: USB 2.0 Camera [1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd. [5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v2 -> v3: Update the insertion order of ID 0c45 v1 -> v2: Arrange the ID in order --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 58156fbca02c..b42654033c33 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2125,6 +2125,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x0b0e, 0x0349, /* Jabra 550a */ QUIRK_FLAG_CTL_MSG_DELAY_1M), + DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x0ecb, 0x205c, /* JBL Quantum610 Wireless */ QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x0ecb, 0x2069, /* JBL Quantum810 Wireless */ -- 2.25.1

1 0

[PATCH] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 22 Jul '24

22 Jul '24

Restores the audio functionality that was broken since Linux version 6.6.y by adding the missing probe functions for the player and reader components. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Specifically, the probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. My patch correctly reinserts the missing probe entries, restoring the intended functionality. The patch modifies the following files: - `sound/soc/sti/sti_uniperif.c`: Changes the visibility of `sti_uniperiph_dai_probe` to non-static. - `sound/soc/sti/uniperif.h`: Adds the declaration of `sti_uniperiph_dai_probe`. - `sound/soc/sti/uniperif_player.c`: Adds `probe` function to `uni_player_dai_ops`. - `sound/soc/sti/uniperif_reader.c`: Adds `probe` function to `uni_reader_dai_ops`. This ensures the correct `probe` functions are utilized, thus fixing the audio regression. --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

1 0

[PATCH v2] ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
by wangdicheng 22 Jul '24

22 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> Sonix HD USB Camera does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. (snip) [1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd [1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00 [1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [1.984127] usb 3-3: Product: USB 2.0 Camera [1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd. [5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v1 -> v2: Arrange the ID in order --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 58156fbca02c..54b2d2f314ac 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2225,6 +2225,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

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aplay: Print '=== PAUSE ===' only if it is supported
by GitHub pull_request - opened 22 Jul '24

22 Jul '24

alsa-project/alsa-utils pull request #271 was opened from ujfalusi: Instead of printing the '=== PAUSE ===' unconditionally before calling do_pause(), move it to the function and only print it if the stream can be paused. If the stream cannot be paused that the '=== PAUSE ===' will be replaced by `PAUSE command ignored (no hw support)` immediately, which is not observable by users but automation scripts will catch the '=== PAUSE ===' and might think that the stream is indeed got paused. Move the print into do_pause() function after the snd_pcm_pause() have returned without error to make sure it is only printed if the stream is paused and we are waiting for the pause release from user to proceed. Request URL : https://github.com/alsa-project/alsa-utils/pull/271 Patch URL : https://github.com/alsa-project/alsa-utils/pull/271.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Fwd: [PATCH] ALSA: hda/realtek: Implement sound init sequence for Samsung Galaxy Book3 Pro 360
by Nick Weihs 22 Jul '24

22 Jul '24

Samsung Galaxy Book3 Pro 360 sends a large amount of data to the codec through hda processing coefficients. This data was captured using a modified version of QEMU, but the actual content of the data remains opaque to me. Elliding any part of the data seems to cause sound to not work. Signed-off-by: Nick Weihs <nick.weihs(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 305 ++++++++++++++++++++++++++++++++++ 1 file changed, 305 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa76d1c88589..cbe6d9689fc3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4796,6 +4796,304 @@ static void alc298_fixup_samsung_amp(struct hda_codec *codec, } } +static inline void alc298_samsung_write_coef_pack2(struct hda_codec *codec, + const unsigned short coefs[4]) +{ + int i; + + for (i = 0; i < 100; i++) { + if ((alc_read_coef_idx(codec, 0x26) & 0x0010) == 0) + break; + } + + alc_write_coef_idx(codec, 0x23, coefs[0]); + alc_write_coef_idx(codec, 0x24, coefs[1]); + alc_write_coef_idx(codec, 0x25, coefs[2]); + alc_write_coef_idx(codec, 0x26, coefs[3]); +} + +static inline void alc298_samsung_write_coef_pack_seq( + struct hda_codec *codec, + unsigned char target, + const unsigned short (*sequence)[4], + int count) +{ + alc_write_coef_idx(codec, 0x22, target); + for (int i = 0; i < count; i++) + alc298_samsung_write_coef_pack2(codec, sequence[i]); +} + +struct alc298_samsung_coeff_fixup_desc { + unsigned char coeff_idx; + unsigned short coeff_value; +}; + +static void alc298_fixup_samsung_amp2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + int i; + static const struct alc298_samsung_coeff_fixup_desc fixups1[] = { + { 0x99, 0x8000 }, { 0x82, 0x4408 }, { 0x32, 0x3f00 }, { 0x0e, 0x6f80 }, + { 0x10, 0x0e21 }, { 0x55, 0x8000 }, { 0x08, 0x2fcf }, { 0x08, 0x2fcf }, + { 0x2d, 0xc020 }, { 0x19, 0x0017 }, { 0x50, 0x1000 }, { 0x0e, 0x6f80 }, + { 0x08, 0x2fcf }, { 0x80, 0x0011 }, { 0x2b, 0x0c10 }, { 0x2d, 0xc020 }, + { 0x03, 0x0042 }, { 0x0f, 0x0062 }, { 0x08, 0x2fcf }, + }; + + static const unsigned short amp_0x38[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0001, 0xB011 }, { 0x201D, 0x0000, 0x0001, 0xB011 }, + { 0x201F, 0x0000, 0x00FE, 0xB011 }, { 0x2021, 0x0000, 0x0000, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x2399, 0x0000, 0x0003, 0xB011 }, + { 0x23A4, 0x0000, 0x00B5, 0xB011 }, { 0x23A5, 0x0000, 0x0001, 0xB011 }, + { 0x23BA, 0x0000, 0x0094, 0xB011 }, { 0x2100, 0x00D0, 0x950E, 0xB017 }, + { 0x2104, 0x0061, 0xD4E2, 0xB017 }, { 0x2108, 0x00D0, 0x950E, 0xB017 }, + { 0x210C, 0x0075, 0xF4E2, 0xB017 }, { 0x2110, 0x00B4, 0x4B0D, 0xB017 }, + { 0x2114, 0x000A, 0x1000, 0xB017 }, { 0x2118, 0x0015, 0x2000, 0xB017 }, + { 0x211C, 0x000A, 0x1000, 0xB017 }, { 0x2120, 0x0075, 0xF4E2, 0xB017 }, + { 0x2124, 0x00B4, 0x4B0D, 0xB017 }, { 0x2128, 0x0000, 0x0010, 0xB017 }, + { 0x212C, 0x0000, 0x0000, 0xB017 }, { 0x2130, 0x0000, 0x0000, 0xB017 }, + { 0x2134, 0x0000, 0x0000, 0xB017 }, { 0x2138, 0x0000, 0x0000, 0xB017 }, + { 0x213C, 0x0000, 0x0010, 0xB017 }, { 0x2140, 0x0000, 0x0000, 0xB017 }, + { 0x2144, 0x0000, 0x0000, 0xB017 }, { 0x2148, 0x0000, 0x0000, 0xB017 }, + { 0x214C, 0x0000, 0x0000, 0xB017 }, { 0x2150, 0x0000, 0x0010, 0xB017 }, + { 0x2154, 0x0000, 0x0000, 0xB017 }, { 0x2158, 0x0000, 0x0000, 0xB017 }, + { 0x215C, 0x0000, 0x0000, 0xB017 }, { 0x2160, 0x0000, 0x0000, 0xB017 }, + { 0x2164, 0x0000, 0x0010, 0xB017 }, { 0x2168, 0x0000, 0x0000, 0xB017 }, + { 0x216C, 0x0000, 0x0000, 0xB017 }, { 0x2170, 0x0000, 0x0000, 0xB017 }, + { 0x2174, 0x0000, 0x0000, 0xB017 }, { 0x2178, 0x0000, 0x0010, 0xB017 }, + { 0x217C, 0x0000, 0x0000, 0xB017 }, { 0x2180, 0x0000, 0x0000, 0xB017 }, + { 0x2184, 0x0000, 0x0000, 0xB017 }, { 0x2188, 0x0000, 0x0000, 0xB017 }, + { 0x218C, 0x0064, 0x5800, 0xB017 }, { 0x2190, 0x00C8, 0xB000, 0xB017 }, + { 0x2194, 0x0064, 0x5800, 0xB017 }, { 0x2198, 0x003D, 0x5BE7, 0xB017 }, + { 0x219C, 0x0054, 0x060A, 0xB017 }, { 0x21A0, 0x00C8, 0xA310, 0xB017 }, + { 0x21A4, 0x0029, 0x4DE5, 0xB017 }, { 0x21A8, 0x0032, 0x420C, 0xB017 }, + { 0x21AC, 0x0029, 0x4DE5, 0xB017 }, { 0x21B0, 0x00FA, 0xE50C, 0xB017 }, + { 0x21B4, 0x0000, 0x0010, 0xB017 }, { 0x21B8, 0x0000, 0x0000, 0xB017 }, + { 0x21BC, 0x0000, 0x0000, 0xB017 }, { 0x21C0, 0x0000, 0x0000, 0xB017 }, + { 0x21C4, 0x0000, 0x0000, 0xB017 }, { 0x21C8, 0x0056, 0xC50F, 0xB017 }, + { 0x21CC, 0x007B, 0xD7E1, 0xB017 }, { 0x21D0, 0x0077, 0xA70E, 0xB017 }, + { 0x21D4, 0x00E0, 0xBDE1, 0xB017 }, { 0x21D8, 0x0032, 0x530E, 0xB017 }, + { 0x2204, 0x00FB, 0x7E0F, 0xB017 }, { 0x2208, 0x000B, 0x02E1, 0xB017 }, + { 0x220C, 0x00FB, 0x7E0F, 0xB017 }, { 0x2210, 0x00D5, 0x17E1, 0xB017 }, + { 0x2214, 0x00C0, 0x130F, 0xB017 }, { 0x2218, 0x00E5, 0x0A00, 0xB017 }, + { 0x221C, 0x00CB, 0x1500, 0xB017 }, { 0x2220, 0x00E5, 0x0A00, 0xB017 }, + { 0x2224, 0x00D5, 0x17E1, 0xB017 }, { 0x2228, 0x00C0, 0x130F, 0xB017 }, + { 0x222C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2230, 0x0017, 0x48E2, 0xB017 }, + { 0x2234, 0x00F5, 0xDB0E, 0xB017 }, { 0x2238, 0x00EF, 0x5CE2, 0xB017 }, + { 0x223C, 0x00C1, 0xCC0D, 0xB017 }, { 0x2240, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2244, 0x0017, 0x48E2, 0xB017 }, { 0x2248, 0x00F5, 0xDB0E, 0xB017 }, + { 0x224C, 0x00EF, 0x5CE2, 0xB017 }, { 0x2250, 0x00C1, 0xCC0D, 0xB017 }, + { 0x2254, 0x00F5, 0xDB0E, 0xB017 }, { 0x2258, 0x0017, 0x48E2, 0xB017 }, + { 0x225C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2260, 0x00EF, 0x5CE2, 0xB017 }, + { 0x2264, 0x00C1, 0xCC0D, 0xB017 }, { 0x2268, 0x00F5, 0xDB0E, 0xB017 }, + { 0x226C, 0x0017, 0x48E2, 0xB017 }, { 0x2270, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2274, 0x00EF, 0x5CE2, 0xB017 }, { 0x2278, 0x00C1, 0xCC0D, 0xB017 }, + { 0x227C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2280, 0x0017, 0x48E2, 0xB017 }, + { 0x2284, 0x00F5, 0xDB0E, 0xB017 }, { 0x2288, 0x00EF, 0x5CE2, 0xB017 }, + { 0x228C, 0x00C1, 0xCC0D, 0xB017 }, { 0x22CC, 0x00E8, 0x8D00, 0xB017 }, + { 0x22D0, 0x0000, 0x0000, 0xB017 }, { 0x22D4, 0x0018, 0x72FF, 0xB017 }, + { 0x22D8, 0x00CE, 0x25E1, 0xB017 }, { 0x22DC, 0x002F, 0xE40E, 0xB017 }, + { 0x238E, 0x0000, 0x0099, 0xB011 }, { 0x238F, 0x0000, 0x0011, 0xB011 }, + { 0x2390, 0x0000, 0x0056, 0xB011 }, { 0x2391, 0x0000, 0x0004, 0xB011 }, + { 0x2392, 0x0000, 0x00BB, 0xB011 }, { 0x2393, 0x0000, 0x006D, 0xB011 }, + { 0x2394, 0x0000, 0x0010, 0xB011 }, { 0x2395, 0x0000, 0x0064, 0xB011 }, + { 0x2396, 0x0000, 0x00B6, 0xB011 }, { 0x2397, 0x0000, 0x0028, 0xB011 }, + { 0x2398, 0x0000, 0x000B, 0xB011 }, { 0x239A, 0x0000, 0x0099, 0xB011 }, + { 0x239B, 0x0000, 0x000D, 0xB011 }, { 0x23A6, 0x0000, 0x0064, 0xB011 }, + { 0x23A7, 0x0000, 0x0078, 0xB011 }, { 0x23B9, 0x0000, 0x0000, 0xB011 }, + { 0x23E0, 0x0000, 0x0021, 0xB011 }, { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x39[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0002, 0xB011 }, { 0x201D, 0x0000, 0x0002, 0xB011 }, + { 0x201F, 0x0000, 0x00FD, 0xB011 }, { 0x2021, 0x0000, 0x0001, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x2399, 0x0000, 0x0003, 0xB011 }, + { 0x23A4, 0x0000, 0x00B5, 0xB011 }, { 0x23A5, 0x0000, 0x0001, 0xB011 }, + { 0x23BA, 0x0000, 0x0094, 0xB011 }, { 0x2100, 0x00D0, 0x950E, 0xB017 }, + { 0x2104, 0x0061, 0xD4E2, 0xB017 }, { 0x2108, 0x00D0, 0x950E, 0xB017 }, + { 0x210C, 0x0075, 0xF4E2, 0xB017 }, { 0x2110, 0x00B4, 0x4B0D, 0xB017 }, + { 0x2114, 0x000A, 0x1000, 0xB017 }, { 0x2118, 0x0015, 0x2000, 0xB017 }, + { 0x211C, 0x000A, 0x1000, 0xB017 }, { 0x2120, 0x0075, 0xF4E2, 0xB017 }, + { 0x2124, 0x00B4, 0x4B0D, 0xB017 }, { 0x2128, 0x0000, 0x0010, 0xB017 }, + { 0x212C, 0x0000, 0x0000, 0xB017 }, { 0x2130, 0x0000, 0x0000, 0xB017 }, + { 0x2134, 0x0000, 0x0000, 0xB017 }, { 0x2138, 0x0000, 0x0000, 0xB017 }, + { 0x213C, 0x0000, 0x0010, 0xB017 }, { 0x2140, 0x0000, 0x0000, 0xB017 }, + { 0x2144, 0x0000, 0x0000, 0xB017 }, { 0x2148, 0x0000, 0x0000, 0xB017 }, + { 0x214C, 0x0000, 0x0000, 0xB017 }, { 0x2150, 0x0000, 0x0010, 0xB017 }, + { 0x2154, 0x0000, 0x0000, 0xB017 }, { 0x2158, 0x0000, 0x0000, 0xB017 }, + { 0x215C, 0x0000, 0x0000, 0xB017 }, { 0x2160, 0x0000, 0x0000, 0xB017 }, + { 0x2164, 0x0000, 0x0010, 0xB017 }, { 0x2168, 0x0000, 0x0000, 0xB017 }, + { 0x216C, 0x0000, 0x0000, 0xB017 }, { 0x2170, 0x0000, 0x0000, 0xB017 }, + { 0x2174, 0x0000, 0x0000, 0xB017 }, { 0x2178, 0x0000, 0x0010, 0xB017 }, + { 0x217C, 0x0000, 0x0000, 0xB017 }, { 0x2180, 0x0000, 0x0000, 0xB017 }, + { 0x2184, 0x0000, 0x0000, 0xB017 }, { 0x2188, 0x0000, 0x0000, 0xB017 }, + { 0x218C, 0x0064, 0x5800, 0xB017 }, { 0x2190, 0x00C8, 0xB000, 0xB017 }, + { 0x2194, 0x0064, 0x5800, 0xB017 }, { 0x2198, 0x003D, 0x5BE7, 0xB017 }, + { 0x219C, 0x0054, 0x060A, 0xB017 }, { 0x21A0, 0x00C8, 0xA310, 0xB017 }, + { 0x21A4, 0x0029, 0x4DE5, 0xB017 }, { 0x21A8, 0x0032, 0x420C, 0xB017 }, + { 0x21AC, 0x0029, 0x4DE5, 0xB017 }, { 0x21B0, 0x00FA, 0xE50C, 0xB017 }, + { 0x21B4, 0x0000, 0x0010, 0xB017 }, { 0x21B8, 0x0000, 0x0000, 0xB017 }, + { 0x21BC, 0x0000, 0x0000, 0xB017 }, { 0x21C0, 0x0000, 0x0000, 0xB017 }, + { 0x21C4, 0x0000, 0x0000, 0xB017 }, { 0x21C8, 0x0056, 0xC50F, 0xB017 }, + { 0x21CC, 0x007B, 0xD7E1, 0xB017 }, { 0x21D0, 0x0077, 0xA70E, 0xB017 }, + { 0x21D4, 0x00E0, 0xBDE1, 0xB017 }, { 0x21D8, 0x0032, 0x530E, 0xB017 }, + { 0x2204, 0x00FB, 0x7E0F, 0xB017 }, { 0x2208, 0x000B, 0x02E1, 0xB017 }, + { 0x220C, 0x00FB, 0x7E0F, 0xB017 }, { 0x2210, 0x00D5, 0x17E1, 0xB017 }, + { 0x2214, 0x00C0, 0x130F, 0xB017 }, { 0x2218, 0x00E5, 0x0A00, 0xB017 }, + { 0x221C, 0x00CB, 0x1500, 0xB017 }, { 0x2220, 0x00E5, 0x0A00, 0xB017 }, + { 0x2224, 0x00D5, 0x17E1, 0xB017 }, { 0x2228, 0x00C0, 0x130F, 0xB017 }, + { 0x222C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2230, 0x0017, 0x48E2, 0xB017 }, + { 0x2234, 0x00F5, 0xDB0E, 0xB017 }, { 0x2238, 0x00EF, 0x5CE2, 0xB017 }, + { 0x223C, 0x00C1, 0xCC0D, 0xB017 }, { 0x2240, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2244, 0x0017, 0x48E2, 0xB017 }, { 0x2248, 0x00F5, 0xDB0E, 0xB017 }, + { 0x224C, 0x00EF, 0x5CE2, 0xB017 }, { 0x2250, 0x00C1, 0xCC0D, 0xB017 }, + { 0x2254, 0x00F5, 0xDB0E, 0xB017 }, { 0x2258, 0x0017, 0x48E2, 0xB017 }, + { 0x225C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2260, 0x00EF, 0x5CE2, 0xB017 }, + { 0x2264, 0x00C1, 0xCC0D, 0xB017 }, { 0x2268, 0x00F5, 0xDB0E, 0xB017 }, + { 0x226C, 0x0017, 0x48E2, 0xB017 }, { 0x2270, 0x00F5, 0xDB0E, 0xB017 }, + { 0x2274, 0x00EF, 0x5CE2, 0xB017 }, { 0x2278, 0x00C1, 0xCC0D, 0xB017 }, + { 0x227C, 0x00F5, 0xDB0E, 0xB017 }, { 0x2280, 0x0017, 0x48E2, 0xB017 }, + { 0x2284, 0x00F5, 0xDB0E, 0xB017 }, { 0x2288, 0x00EF, 0x5CE2, 0xB017 }, + { 0x228C, 0x00C1, 0xCC0D, 0xB017 }, { 0x22CC, 0x00E8, 0x8D00, 0xB017 }, + { 0x22D0, 0x0000, 0x0000, 0xB017 }, { 0x22D4, 0x0018, 0x72FF, 0xB017 }, + { 0x22D8, 0x00CE, 0x25E1, 0xB017 }, { 0x22DC, 0x002F, 0xE40E, 0xB017 }, + { 0x238E, 0x0000, 0x0099, 0xB011 }, { 0x238F, 0x0000, 0x0011, 0xB011 }, + { 0x2390, 0x0000, 0x0056, 0xB011 }, { 0x2391, 0x0000, 0x0004, 0xB011 }, + { 0x2392, 0x0000, 0x00BB, 0xB011 }, { 0x2393, 0x0000, 0x006D, 0xB011 }, + { 0x2394, 0x0000, 0x0010, 0xB011 }, { 0x2395, 0x0000, 0x0064, 0xB011 }, + { 0x2396, 0x0000, 0x00B6, 0xB011 }, { 0x2397, 0x0000, 0x0028, 0xB011 }, + { 0x2398, 0x0000, 0x000B, 0xB011 }, { 0x239A, 0x0000, 0x0099, 0xB011 }, + { 0x239B, 0x0000, 0x000D, 0xB011 }, { 0x23A6, 0x0000, 0x0064, 0xB011 }, + { 0x23A7, 0x0000, 0x0078, 0xB011 }, { 0x23B9, 0x0000, 0x0000, 0xB011 }, + { 0x23E0, 0x0000, 0x0021, 0xB011 }, { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x3c[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0001, 0xB011 }, { 0x201D, 0x0000, 0x0001, 0xB011 }, + { 0x201F, 0x0000, 0x00FE, 0xB011 }, { 0x2021, 0x0000, 0x0000, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x23BA, 0x0000, 0x008D, 0xB011 }, + { 0x2128, 0x0005, 0x460D, 0xB017 }, { 0x212C, 0x00F6, 0x73E5, 0xB017 }, + { 0x2130, 0x0005, 0x460D, 0xB017 }, { 0x2134, 0x00C0, 0xE9E5, 0xB017 }, + { 0x2138, 0x00D5, 0x010B, 0xB017 }, { 0x213C, 0x009D, 0x7809, 0xB017 }, + { 0x2140, 0x00C5, 0x0EED, 0xB017 }, { 0x2144, 0x009D, 0x7809, 0xB017 }, + { 0x2148, 0x00C4, 0x4EF0, 0xB017 }, { 0x214C, 0x003A, 0x3106, 0xB017 }, + { 0x2150, 0x00AF, 0x750E, 0xB017 }, { 0x2154, 0x008C, 0x1FF1, 0xB017 }, + { 0x2158, 0x009E, 0x360C, 0xB017 }, { 0x215C, 0x008C, 0x1FF1, 0xB017 }, + { 0x2160, 0x004D, 0xAC0A, 0xB017 }, { 0x2164, 0x007D, 0xA00F, 0xB017 }, + { 0x2168, 0x00E1, 0x9CE3, 0xB017 }, { 0x216C, 0x00E8, 0x590E, 0xB017 }, + { 0x2170, 0x00E1, 0x9CE3, 0xB017 }, { 0x2174, 0x0066, 0xFA0D, 0xB017 }, + { 0x2178, 0x0000, 0x0010, 0xB017 }, { 0x217C, 0x0000, 0x0000, 0xB017 }, + { 0x2180, 0x0000, 0x0000, 0xB017 }, { 0x2184, 0x0000, 0x0000, 0xB017 }, + { 0x2188, 0x0000, 0x0000, 0xB017 }, { 0x218C, 0x0000, 0x0010, 0xB017 }, + { 0x2190, 0x0000, 0x0000, 0xB017 }, { 0x2194, 0x0000, 0x0000, 0xB017 }, + { 0x2198, 0x0000, 0x0000, 0xB017 }, { 0x219C, 0x0000, 0x0000, 0xB017 }, + { 0x21A0, 0x0000, 0x0010, 0xB017 }, { 0x21A4, 0x0000, 0x0000, 0xB017 }, + { 0x21A8, 0x0000, 0x0000, 0xB017 }, { 0x21AC, 0x0000, 0x0000, 0xB017 }, + { 0x21B0, 0x0000, 0x0000, 0xB017 }, { 0x21B4, 0x0000, 0x0010, 0xB017 }, + { 0x21B8, 0x0000, 0x0000, 0xB017 }, { 0x21BC, 0x0000, 0x0000, 0xB017 }, + { 0x21C0, 0x0000, 0x0000, 0xB017 }, { 0x21C4, 0x0000, 0x0000, 0xB017 }, + { 0x23B9, 0x0000, 0x0000, 0xB011 }, { 0x23E0, 0x0000, 0x0020, 0xB011 }, + { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_0x3d[][4] = { + { 0x2000, 0x0000, 0x0001, 0xB011 }, { 0x23FF, 0x0000, 0x0000, 0xB011 }, + { 0x203A, 0x0000, 0x0080, 0xB011 }, { 0x23E1, 0x0000, 0x0000, 0xB011 }, + { 0x2012, 0x0000, 0x006F, 0xB011 }, { 0x2014, 0x0000, 0x0000, 0xB011 }, + { 0x201B, 0x0000, 0x0002, 0xB011 }, { 0x201D, 0x0000, 0x0002, 0xB011 }, + { 0x201F, 0x0000, 0x00FD, 0xB011 }, { 0x2021, 0x0000, 0x0001, 0xB011 }, + { 0x2022, 0x0000, 0x0010, 0xB011 }, { 0x203D, 0x0000, 0x0005, 0xB011 }, + { 0x203F, 0x0000, 0x0003, 0xB011 }, { 0x2050, 0x0000, 0x002C, 0xB011 }, + { 0x2076, 0x0000, 0x000E, 0xB011 }, { 0x207C, 0x0000, 0x004A, 0xB011 }, + { 0x2081, 0x0000, 0x0003, 0xB011 }, { 0x23BA, 0x0000, 0x008D, 0xB011 }, + { 0x2128, 0x0005, 0x460D, 0xB017 }, { 0x212C, 0x00F6, 0x73E5, 0xB017 }, + { 0x2130, 0x0005, 0x460D, 0xB017 }, { 0x2134, 0x00C0, 0xE9E5, 0xB017 }, + { 0x2138, 0x00D5, 0x010B, 0xB017 }, { 0x213C, 0x009D, 0x7809, 0xB017 }, + { 0x2140, 0x00C5, 0x0EED, 0xB017 }, { 0x2144, 0x009D, 0x7809, 0xB017 }, + { 0x2148, 0x00C4, 0x4EF0, 0xB017 }, { 0x214C, 0x003A, 0x3106, 0xB017 }, + { 0x2150, 0x00AF, 0x750E, 0xB017 }, { 0x2154, 0x008C, 0x1FF1, 0xB017 }, + { 0x2158, 0x009E, 0x360C, 0xB017 }, { 0x215C, 0x008C, 0x1FF1, 0xB017 }, + { 0x2160, 0x004D, 0xAC0A, 0xB017 }, { 0x2164, 0x007D, 0xA00F, 0xB017 }, + { 0x2168, 0x00E1, 0x9CE3, 0xB017 }, { 0x216C, 0x00E8, 0x590E, 0xB017 }, + { 0x2170, 0x00E1, 0x9CE3, 0xB017 }, { 0x2174, 0x0066, 0xFA0D, 0xB017 }, + { 0x2178, 0x0000, 0x0010, 0xB017 }, { 0x217C, 0x0000, 0x0000, 0xB017 }, + { 0x2180, 0x0000, 0x0000, 0xB017 }, { 0x2184, 0x0000, 0x0000, 0xB017 }, + { 0x2188, 0x0000, 0x0000, 0xB017 }, { 0x218C, 0x0000, 0x0010, 0xB017 }, + { 0x2190, 0x0000, 0x0000, 0xB017 }, { 0x2194, 0x0000, 0x0000, 0xB017 }, + { 0x2198, 0x0000, 0x0000, 0xB017 }, { 0x219C, 0x0000, 0x0000, 0xB017 }, + { 0x21A0, 0x0000, 0x0010, 0xB017 }, { 0x21A4, 0x0000, 0x0000, 0xB017 }, + { 0x21A8, 0x0000, 0x0000, 0xB017 }, { 0x21AC, 0x0000, 0x0000, 0xB017 }, + { 0x21B0, 0x0000, 0x0000, 0xB017 }, { 0x21B4, 0x0000, 0x0010, 0xB017 }, + { 0x21B8, 0x0000, 0x0000, 0xB017 }, { 0x21BC, 0x0000, 0x0000, 0xB017 }, + { 0x21C0, 0x0000, 0x0000, 0xB017 }, { 0x21C4, 0x0000, 0x0000, 0xB017 }, + { 0x23B9, 0x0000, 0x0000, 0xB011 }, { 0x23E0, 0x0000, 0x0020, 0xB011 }, + { 0x23E1, 0x0000, 0x0001, 0xB011 }, + }; + + static const unsigned short amp_seq1[][4] = { + { 0x23FF, 0x0000, 0x0000, 0xB011 }, { 0x203A, 0x0000, 0x0080, 0xB011 }, + }; + + static const struct alc298_samsung_coeff_fixup_desc fixups2[] = { + { 0x4f, 0xb029 }, { 0x05, 0x2be0 }, { 0x30, 0x2421 }, + }; + + + static const unsigned short amp_seq2[][4] = { + { 0x203A, 0x0000, 0x0081, 0xB011 }, { 0x23FF, 0x0000, 0x0001, 0xB011 }, + }; + + if (action != HDA_FIXUP_ACT_INIT) + return; + + ///// First set of fixups + for (i = 0; i < ARRAY_SIZE(fixups1); i++) + alc_write_coef_idx(codec, fixups1[i].coeff_idx, fixups1[i].coeff_value); + + ///// First set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_0x38, ARRAY_SIZE(amp_0x38)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_0x39, ARRAY_SIZE(amp_0x39)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_0x3c, ARRAY_SIZE(amp_0x3c)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_0x3d, ARRAY_SIZE(amp_0x3d)); + + ///// Second set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq1, ARRAY_SIZE(amp_seq1)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq1, ARRAY_SIZE(amp_seq1)); + + ////// Second set of fixups + for (i = 0; i < ARRAY_SIZE(fixups2); i++) + alc_write_coef_idx(codec, fixups2[i].coeff_idx, fixups2[i].coeff_value); + + ///// Third set of writes + alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq2, ARRAY_SIZE(amp_seq2)); + alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq2, ARRAY_SIZE(amp_seq2)); + + ///// Final fixup + alc_write_coef_idx(codec, 0x10, 0x0F21); +} + #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) @@ -7426,6 +7724,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC298_FIXUP_SAMSUNG_AMP, + ALC298_FIXUP_SAMSUNG_AMP2, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -9021,6 +9320,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET }, + [ALC298_FIXUP_SAMSUNG_AMP2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_samsung_amp2 + }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10352,6 +10655,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG-KB1US)", ALC298_FIXUP_SAMSUNG_AMP2), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -10781,6 +11085,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_AMP, .name = "alc298-samsung-amp"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP2, .name = "alc298-samsung-amp2"}, {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, -- 2.45.2

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[PATCH v10] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by Baojun Xu 21 Jul '24

21 Jul '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. The code support Realtek as the primary codec. In patch version-10, add multi devices firmware binary support, to compatble with windows driver, they can share same firmware binary. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v10: - Remove project id 103c8b93 in Patch_realtek.c, as it was used for test only, offcial project id is 103c8daa (Gemtree). - Add TAS2781_REG_CLK_CONFIG for check if AMP has been reset. - Add table for pre-reset and post-reset. - Perform pre-reset config before reset AMP. - Perform post-reset config after reset AMP. - Fix wrong offset in calibration data issue. - Remove tuning switch after firmware download. - Force stop play in runtime suspend. - Apply patch for component change in bind & unbind. - Remove program download and calibration data apply, switch tuning only in runtime resume. - Check if AMP has reseted in system resume, and do firmware download and calibration data apply. - Change flag 0xCx to 0x8x for coeff data force download. - Add device index for multi device firmware binary file compatible. - Check device index before registers write during firmware download. - Force config data download after program download. - Remove config download in tuning switch. v9: - Add parentheses for variables in micro define. - Add define for calibration registers address identify. - Change variable name from calibration_data to cali_data. - Add variable for calibration registers address in struct tasdevice_priv. - Change format for alignment to parentheses. - Remove tas2781_reg_defaults as it will be over write in firmware download. - Remove RESET for second device in tas2781_spi_reset(), as it will cause no sound on second device. - Add calibration registers address initial in tasdevice_spi_init(). - Remove unnecessary parentheses. - Remove exception in tas2781_hda_playback_hook(). - Remove hard code for calibration registers address. - Add support for calibration data format V3 for registers address identify, as new firmware tool will define calibration registers address dynamic, so can't hard code it, need read it from BIOS with calibration data. - Remove some debug print. - Change hard code value (3) to ARRAY_SIZE(tas2781_snd_controls)/2. - New patch apply for change position of component_del(). - Change label from out, out1 to config_err and block_err. - Remove unnecessary parentheses in tas2781_spi_fwlib.c. - Remove tasdevice_spi_prmg_calibdata_load(). v8: - Move *-y position in sound/pci/hda/Makefile. - Change + to | for bit operation in macro define. - Change TASDEVICE_Checksum to TASDEVICE_CHECKSUM. v7: - Add new project Gemtree (0x103c8daa) support for tas2781 spi driver. v6: - Remove client in tasdevice_priv struct as it was not used. - Change format for tasdevice_spi_dev_write() in tas2781_hda_spi.c. - Change update bits function to read & write mode, as tas2781 use last bit to mask read & write, it cause regmap_update_bits work not as our expected. - Add element in tas2781_snd_controls, tas2781_prof_ctrl, tas2781_dsp_conf_ctrl, to support second device. Previous version add sound card for first index only. - Change calibration registers address declaire to TASDEVICE_REG() mode. - Add snd_ctl_new1() for second device. - Change format for tas2781_hda_unbind() to fit 79 chars in one line. - Remove client for spi as it was not used. - Remove unused variables. - Remove cur_prog clean in runtime_suspend(). - Remove unused variables. - Add tasdevice_spi_tuning_switch() in runtime_resume. - Remove cur_prog clean in system_resume(). - Add tuning switch if playback started in system_resume(). - Change TASDEVICE_PAGE_REG to TASDEVICE_REG_ID in tas2781_spi_fwlib.c. - Remove calibration binary load functions as data will be loaded from EFI. v5: - Combined all modification into one patch for pass the compile. - Remove old hardware id "TIAS2781" as no production on market. - Change max page size to 256 in tas2781-spi.h. - Change book reg from 127 to GENMASK(7, 1), as one bit shift is needed. - Change register address define as one bit shift needed for offset. - Change block check sum comments to Firmware block from I2C. - Change define of (page + reg) to TASDEVICE_PAGE_REG from TASDEVICE_PGRG. - Change to lower case in defined value. - Add registers default value table in tas2781_hda_spi.c - Change rang_max to 256 * 256. - Add zero_flag_mask in regmap_config. - Change max_register to 256 * 256. - Add registers default table into structure regmap_config. - Change parameter from book number to whole register address. - Remove page 0 switch as SPI driver will do it. - Add bit(0) for read function as tas2781 will check last bit of address, to determine operation mode is read (1) or write (0). - Change switch case to if check in tas2781_hda_playback_hook(), as it may cause compile problem in clang-19 building. - Change variable declaration position. - Change order in if check. - Remove old hardware id ("TIAS2781") support. - Remove debug message in runtime suspend and resume. - Remove regmap.h include in tas2781_spi_fwlib.c. - Remove MODULE_ in fwlib as this file will work with tas2781_hda_spi.c only. v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs - Move include of sound/tas2781-dsp.h to c source file tas2781_hda_spi.c, tas2781_spi_fwlib.c, as it's not needed in header file tas2781-spi.h - Use page and book number define in other micro define - Change 4000000 to (4 * HZ_PER_MHZ) - Add define for calibration variable name and size in UEFI. - Remove structure tasdevice & tasdevice_irqinfo, and move variables in to structure tasdevice_priv - Remove some unused variables in sttructure tasdevice_priv - Remove function declare of tascodec_init(), use tascodec_spi_init() - Remove function declare of tasdevice_remove() - Add array_size.h included in tas2781_hda_spi.c - Add cleanup.h included for change mutex to guard - Add units.h included - Include tas2781-dsp.h - Remove unused variables in tas2781_hda structure - Change 0xff to GENMASK(7, 0) - Change 128 to define of max page size - Change code as all variables was moved to tasdevice_priv structure - Change comments for error in book write, or page write - Remove initial of ret, and return 0 directly - Change comments for wrong typo, add space in front and end - Add check for bulk read, max size should be a page size - Change usleep_rang() to fsleep() - Change mutex_lock to guard(mutex), and remove mutex_unlock() - Change tasdevice_spi_init() to void from int as no value return required - Change second parameter to sizeof - Remove tasdevice_clamp(), use clamp() - Add compatible with old hardware id "TIAS2781" - Remove cs_gpio and spi device in tas2781_read_acpi() as use default CS - Change dev-index check, return -EINVAL if it's <= 0 - Change 0xFF to U8_MAX - Remove GPIO get for chip-select, use default from SPI device - Perform RESET in first AMP only as all AMPs share same RESET pin - return 0 directly in some simple routine to avoid variable initial - Change comments for function which was used by system - Change "ON" "OFF" to function str_on_off(...) - Change format of GUID - Add temp buffer for first efi.get_variable() - Free data if it was allocated - Change format of debug message of calibration date time print - Remove total_sz = 0, as it's not needed - Move save_calibration to after tuning_switch to avoid calibration data overwrite - Remove dev from structure tasdevice_hda, use dev in tasdevice_priv - Remove 0 from { "tas2781-hda", 0}, & {"TXNW2781", 0}, - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v3: - Move HID up to above /* Non-conforming _HID ... */ in scan.c, for avoid misunderstanding. - Move HID up to above /* Non-conforming _HID ... */ in serial-multi-instantiate.c, for avoid misunderstanding. - Change objs to y for snd-hda-scodec-tas2781-spi- in Makefile. - Change format for some define in tas2781-spi.h. - Add define for TASDEVICE_MAX_BOOK_NUM, TASDEVICE_MAX_PAGE. - Move declare of "enum calib_data {" to header file from source file. - Remove "enum device_catlog_id {" as only one platform was supported now. - Remove "struct calidata", as we will save pure calibration data only. - Remove "struct calidata", "enum device_catlog_id" in tasdevice_priv. - Add calibration_data in tasdevice_priv to save pure calibration data. - Remove declare of tasdevice_save_calibration() and tasdevice_apply_calibration(), as it will be used within same c file. - Add below header files included in tas2781_hda_spi.c: bits.h, mutex.h, property.h, time.h, types.h. - Move two micro define to header file tas2781-spi.h. - Change format of some micro define. - Change format of some structure. - Remove irq in tas2781_hda, as have it already in tasdevice_priv. - Remove some local functions declare as not necessarily. - Return error if regmap_write() failed. - Change fixed value 2 to sizeof(data). - Remove all of EXPORT_SYMBOL_GPL, as all of function was called within same module. - Remove empty line after last return in some functions. - Remove some variable initialyzation. - Remove variable sub, store acpi_subsystem_id directly. - Remove wrong comments for device index, it's must, can't be NULL. - Remove some local variables, use elements in structure directly. - Append commas in last element of an array. - Change calibration data process, didn't save all of calibration data in EFI, get in local, and save pure calibration data in tasdevice_priv. - Call tasdevice_save_calibration() from function pointer in tasdevice_priv; - Remove subid as one platform was supported only, needn't check. - Add initialyzation for local variable. - Move regmap initialyzation to before acpi read. - Call tasdevice_apply_calibration() from function pointer in tasdevice_priv; - Remove MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB), as all functions was performed within same module. - Update format and variables declare order. - Change format of multi conditions for if. - Remove casting which is not needed. - Change variables type to avoid casting. - Remove Unneeded parentheses. - Change if check to avoid goto. - Add {} for keep same style. - Remove some local variables, use elements in structure directly. v2: - Change depends on SPI to SPI_MASTER in sound/pci/hda/Kconfig - Add select SND_HDA_SCODEC_COMPONENT in sound/pci/hda/Kconfig - Change comp_generic_fixup() 5th parameter from "-%s:00" to "-%s:00-tas2781-hda.%d" in sound/pci/hda/patch_realtek.c - Change chain_id from ALC285_FIXUP_THINKPAD_HEADSET_JACK to ALC285_FIXUP_HP_GPIO_LED in sound/pci/hda/patch_realtek.c - Remove project name "Gemtree" - Update information in file header of tas2781-spi.h. - Remove useless micro define. - Change TASDEVICE_I2CChecksum to TASDEVICE_Checksum - Remove enum control_bus as current code support SPI only. - Remove device define as current code support TAS2781 only. - Remove spi_device **spi_devs in structure tasdevice_priv. - Remove cal_binaryname in structure tasdevice_priv. - Remove ndev in structure tasdevice_priv. - Change isacpi and isspi, replace by control_bus(I2C or SPI). - Remove void tasdevice_spi_dsp_remove(void *context). - Add acpi_device and irq in structure tas2781_hda in tas2781_hda_spi.c. - Remove parameter chn in all of registers access functions. - Add tascodec_spi_init(). - Add chip select gpio set for SPI device. - Change tasdevice_tuning_switch() to tasdevice_spi_tuning_switch(). - Change device offset from tas_priv->ndev to tas_priv->index. - Change tasdevice_dsp_remove() to tasdevice_spi_dsp_remove(). - tasdevice_prmg_load() to tasdevice_spi_prmg_load(). - Change tasdevice_config_info_remove() to tasdevice_spi_config_info_remove(). - Add one dummy byte shift when read from page 2~127, or not book 0. - Change tasdevice_spi_change_chn_book() to tasdevice_spi_switch_book() - Perform put_device(physdev) before error return. - Change calibration data id to 2781 from 2783 in EFI - Change calibration data structure name in comments. - Remove file name in tas2781_spi_fwlib.c. - Remove #include <linux/i2c.h> as it not needed in SPI mode. - Change TAB to speace in micro define in tas2781_spi_fwlib.c. - Change to up-case for DSP. - Change all of multi-line comments format, empty first line. - Change all of sizeof(struct xx) to sizeof(*xx). - Remove Explicit casting for variables compare to avoid possible fault. - Return directly without goto. - Change 1 << xx to BIT(xx). - Remove deviceNumber[] as current SPI device will support one device only. - Add memory free before return error. - Change "buf[offset]; offset += 1" to buf[offset++]. - Remove some debug information print. - Change firmware binary to single device mode. - Change all bexx_to_cpup() to get_unaligned_bexx(). - Remove ndev process as current SPI device support single device only. - Remove chn and chnend for single device support only. - Change all of registers read/write function, remove parameter chn. - Create new function for single write, burst write, delay, field write in tasdevice_process_block(). --- drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 158 ++ sound/pci/hda/tas2781_hda_spi.c | 1317 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2115 +++++++++++++++++ 8 files changed, 3632 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 59771412686b..b1a8bf1043da 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1769,6 +1769,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 3be016cfe601..b15c819023d0 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -368,6 +368,17 @@ static const struct smi_node cs35l57_hda = { .bus_type = SMI_AUTO_DETECT, }; +static const struct smi_node tas2781_hda = { + .instances = { + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + {} + }, + .bus_type = SMI_AUTO_DETECT, +}; + /* * Note new device-ids must also be added to ignore_serial_bus_ids in * drivers/acpi/scan.c: acpi_device_enumeration_by_parent(). @@ -380,6 +391,7 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "CSC3556", (unsigned long)&cs35l56_hda }, { "CSC3557", (unsigned long)&cs35l57_hda }, { "INT3515", (unsigned long)&int3515_data }, + { "TXNW2781", (unsigned long)&tas2781_hda }, /* Non-conforming _HID for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, { "CLSA0101", (unsigned long)&cs35l41_hda }, diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb15a0248250..62cd9ba408e2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -206,6 +206,20 @@ config SND_HDA_SCODEC_TAS2781_I2C comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m +config SND_HDA_SCODEC_TAS2781_SPI + tristate "Build TAS2781 HD-audio side codec support for SPI Bus" + depends on SPI_MASTER + depends on ACPI + depends on EFI + depends on SND_SOC + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 SPI HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_SPI=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 80210f845df2..210c406dfbc5 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -40,6 +40,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -72,6 +73,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_SPI) += snd-hda-scodec-tas2781-spi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c3a86a99f8c6..51707270943a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,6 +6924,11 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } +static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7501,6 +7506,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, @@ -9726,6 +9732,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC245_FIXUP_TAS2781_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_spi, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -10263,6 +10275,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8daa, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h new file mode 100644 index 000000000000..0cadfd18bde2 --- /dev/null +++ b/sound/pci/hda/tas2781-spi.h @@ -0,0 +1,158 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for TAS2781 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2781_SPI_H__ +#define __TAS2781_SPI_H__ + +#define TASDEVICE_RATES \ + (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TASDEVICE_MAX_BOOK_NUM 256 +#define TASDEVICE_MAX_PAGE 256 + +#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) + +/* PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) +#define TASDEVICE_BOOK_ID(reg) (((reg) & GENMASK(24, 16)) >> 16) +#define TASDEVICE_PAGE_ID(reg) (((reg) & GENMASK(15, 8)) >> 8) +#define TASDEVICE_REG_ID(reg) (((reg) & GENMASK(7, 1)) >> 1) +#define TASDEVICE_PAGE_REG(reg) ((reg) & GENMASK(15, 1)) +#define TASDEVICE_REG(book, page, reg) \ + (((book) << 16) | ((page) << 8) | ((reg) << 1)) + +/* Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* System Reset Check Register */ +#define TAS2781_REG_CLK_CONFIG TASDEVICE_REG(0x0, 0x0, 0x5c) +#define TAS2781_REG_CLK_CONFIG_RESET (0x19) + +/* Block Checksum */ +#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* Flag of calibration registers address. */ +#define TASDEVICE_CALIBRATION_REG_ADDRESS BIT(7) + +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEVICE_CALIBRATION_DATA_SIZE 256 + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +struct tasdevice_priv { + struct tasdevice_fw *cali_data_fmw; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + const char *acpi_subsystem_id; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + unsigned int err_code; + bool is_loading; + bool is_loaderr; + int cur_book; + int cur_prog; + int cur_conf; + int fw_state; + int index; + int irq; + void *codec; + bool force_fwload_status; + bool playback_started; + unsigned int cali_data[CALIB_MAX]; + unsigned int cali_reg_array[CALIB_MAX]; + + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, + int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, + int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); +}; + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int *value); +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int value); +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, + unsigned int reg, unsigned int mask, + unsigned int value); + +void tasdevice_spi_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_spi_config_info_remove(void *context); +int tasdevice_spi_dsp_parser(void *context); +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_spi_dsp_remove(void *context); +void tasdevice_spi_calbin_remove(void *context); +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, int cfg_no, + int rca_conf_no); +int tasdevice_spi_prmg_load(void *context, int prm_no); +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_spi_tuning_switch(void *context, int state); +int tas2781_spi_load_calibration(void *context, char *file_name, + unsigned short i); +#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c new file mode 100644 index 000000000000..dd979cb46fce --- /dev/null +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -0,0 +1,1317 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <linux/acpi.h> +#include <linux/array_size.h> +#include <linux/bits.h> +#include <linux/cleanup.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/pm_runtime.h> +#include <linux/property.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/time.h> +#include <linux/types.h> +#include <linux/units.h> + +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2781-spi.h" + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +struct tas2781_hda { + struct tasdevice_priv *priv; + struct acpi_device *dacpi; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *prof_ctl; +}; + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = TASDEVICE_MAX_SIZE, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = GENMASK(7, 0), + .selector_shift = 0, + .window_start = 0, + .window_len = TASDEVICE_MAX_PAGE, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .zero_flag_mask = true, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = TASDEVICE_MAX_SIZE, +}; + +struct tasdevice_table { + int reg; + unsigned char data[8]; +}; + +static struct tasdevice_table tas2781_pre_reset_table[] = { + /* Values setting for pre-reset. */ + { TASDEVICE_REG(0x00, 0x00, 0x02), {0x01, 0x1a, },}, + { TASDEVICE_REG(0x00, 0x01, 0x37), {0x01, 0x3a, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xc1, },}, + { TASDEVICE_REG(0x00, 0x01, 0x19), {0x01, 0xe0, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xd5, },}, + { TASDEVICE_REG(0x00, 0x00, 0x00), {0x01, 0x00, },}, +}; + +static struct tasdevice_table tas2781_post_reset_table[] = { + /* Values setting for post-reset. */ + { TASDEVICE_REG(0x00, 0x01, 0x37), {0x01, 0x3a, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0d), {0x01, 0x0d, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xc1, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x06), {0x01, 0xd5, },}, + { TASDEVICE_REG(0x00, 0xfd, 0x0a), {0x01, 0x01, },}, + { TASDEVICE_REG(0x00, 0x00, 0x00), {0x01, 0x00, },}, +}; + +static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) { + dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); + return ret; + } + tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); + } + return ret; +} + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int *val) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char data[2]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, sizeof(data)); + *val = data[1]; + } else { + ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned char *data, + unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + if (len > TASDEVICE_MAX_PAGE) + len = TASDEVICE_MAX_PAGE; + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[TASDEVICE_MAX_PAGE+1]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, buf, + len + 1); + memcpy(data, buf + 1, len); + } else { + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, data, + len); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned int reg, + unsigned int mask, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret, val; + + /* + * In our TAS2781 SPI mode, read/write was masked in last bit of + * address, it cause regmap_update_bits() not work as expected. + */ + ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) +{ + int ret; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + fsleep(800); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } + for (int j = 0; j >= 0; j++) { + if (tas2781_pre_reset_table[j].reg == 0) + break; + if (tas2781_pre_reset_table[j].data[0] > 1) { + tasdevice_spi_dev_bulk_write(tas_dev, + tas2781_pre_reset_table[j].reg, + &tas2781_pre_reset_table[j].data[1], + tas2781_pre_reset_table[j].data[0]); + } else { + tasdevice_spi_dev_write(tas_dev, + tas2781_pre_reset_table[j].reg, + tas2781_pre_reset_table[j].data[1]); + } + } + ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + fsleep(1000); + for (int j = 0; j >= 0; j++) { + if (tas2781_post_reset_table[j].reg == 0) + break; + if (tas2781_post_reset_table[j].data[0] > 1) { + tasdevice_spi_dev_bulk_write(tas_dev, + tas2781_post_reset_table[j].reg, + &tas2781_post_reset_table[j].data[1], + tas2781_post_reset_table[j].data[0]); + } else { + tasdevice_spi_dev_write(tas_dev, + tas2781_post_reset_table[j].reg, + tas2781_post_reset_table[j].data[1]); + } + } +} + +static int tascodec_spi_init(struct tasdevice_priv *tas_priv, + void *codec, struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret; + + /* + * Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + guard(mutex)(&tas_priv->codec_lock); + + ret = scnprintf(tas_priv->rca_binaryname, + sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->index); + if (ret <= 0) { + dev_err(tas_priv->dev, "rca name err:0x%08x\n", ret); + return ret; + } + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + return ret; +} + +static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) +{ + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + tas_priv->cur_book = -1; + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + /* Store default registers address for calibration data. */ + tas_priv->cali_reg_array[0] = TASDEVICE_REG(0, 0x17, 0x74); + tas_priv->cali_reg_array[1] = TASDEVICE_REG(0, 0x18, 0x0c); + tas_priv->cali_reg_array[2] = TASDEVICE_REG(0, 0x18, 0x14); + tas_priv->cali_reg_array[3] = TASDEVICE_REG(0, 0x13, 0x70); + tas_priv->cali_reg_array[4] = TASDEVICE_REG(0, 0x18, 0x7c); + + mutex_init(&tas_priv->codec_lock); +} + +static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int val, ret; + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_update_bits(tas_priv, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret, val; + + /* Read the primary device */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + return ret; + } + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int val, ret; + + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "set digital vol err in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, + struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + return ret; + } + + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2781_read_acpi(struct tas2781_hda *tas_hda, + const char *hid, + int id) +{ + struct tasdevice_priv *p = tas_hda->priv; + struct acpi_device *adev; + struct device *physdev; + u32 values[HDA_MAX_COMPONENTS]; + const char *property; + size_t nval; + int ret, i; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, "Failed to find ACPI device: %s\n", hid); + return -ENODEV; + } + + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + tas_hda->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL */ + p->acpi_subsystem_id = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + + property = "ti,dev-index"; + ret = device_property_count_u32(physdev, property); + if (ret <= 0 || ret > ARRAY_SIZE(values)) { + ret = -EINVAL; + goto err; + } + nval = ret; + + ret = device_property_read_u32_array(physdev, property, values, nval); + if (ret) + goto err; + + p->index = U8_MAX; + for (i = 0; i < nval; i++) { + if (values[i] == id) { + p->index = i; + break; + } + } + if (p->index == U8_MAX) { + dev_dbg(p->dev, "No index found in %s\n", property); + ret = -ENODEV; + goto err; + } + + if (p->index == 0) { + /* All of amps share same RESET pin. */ + p->reset = devm_gpiod_get_index_optional(physdev, "reset", + p->index, GPIOD_OUT_LOW); + if (IS_ERR(p->reset)) { + dev_err_probe(p->dev, ret, "Failed on reset GPIO\n"); + goto err; + } + } + put_device(physdev); + + return 0; +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + acpi_dev_put(adev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + if (action == HDA_GEN_PCM_ACT_OPEN) { + pm_runtime_get_sync(dev); + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } else if (action == HDA_GEN_PCM_ACT_CLOSE) { + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->rcabin.ncfgs - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + if (tas_priv->rcabin.profile_cfg_id != val) { + tas_priv->rcabin.profile_cfg_id = val; + return 1; + } + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_priv->fmw->nr_programs - 1; + int val; + + val = clamp(nr_program, 0, max); + + if (tas_priv->cur_prog != val) { + tas_priv->cur_prog = val; + return 1; + } + + return 0; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->fmw->nr_configurations - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + + if (tas_priv->cur_conf != val) { + tas_priv->cur_conf = val; + return 1; + } + + return 0; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + * + * Return 0 if succeeded. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) { + change = false; + } else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 0", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 0", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 0", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 1", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 1", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 1", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl[] = { +{ + .name = "Speaker Profile Id - 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +{ + .name = "Speaker Profile Id - 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +}; +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl[] = { +{ + .name = "Speaker Program Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +{ + .name = "Speaker Program Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { +{ + .name = "Speaker Config Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +{ + .name = "Speaker Config Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + int j, rc; + + /* + * If no calibration data exist in tasdevice_priv *tas_priv, + * calibration apply will be ignored, and use default values + * in firmware binary, which was loaded during firmware download. + */ + if (tas_priv->cali_data[0] == 0) + return; + /* + * Calibration data was saved in tasdevice_priv *tas_priv as: + * unsigned int cali_data[CALIB_MAX]; + * and every data (in 4 bytes) will be saved in register which in + * book 0, and page number in page_array[], offset was saved in + * rgno_array[]. + */ + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + tas_priv->cali_reg_array[j], + (unsigned char *)&tas_priv->cali_data[j], 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + tas_priv->index, j, rc); + } +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE], *buf; + unsigned int attr, crc, offset, *tmp_val; + struct tm *tm = &tas_priv->tm; + unsigned long total_sz = 0; + efi_status_t status; + + tas_priv->cali_data[0] = 0; + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, data); + if (status == EFI_BUFFER_TOO_SMALL) { + if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &total_sz, data); + } + if (status != EFI_SUCCESS) + return -EINVAL; + + tmp_val = (unsigned int *)data; + if (tmp_val[0] == 2781) { + /* + * New features were added in calibrated Data V3: + * 1. Added calibration registers address define in + * a node, marked as Device id == 0x80. + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and + * uniqie_id for multi-link solutions + * 2. Support flexible number of devices instead of + * fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2781, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } // if Device number is 0x80, mean it's + * calibration registers address. + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ + crc = crc32(~0, data, (3 + tmp_val[1] * 6) * 4) ^ ~0; + + if (crc != tmp_val[3 + tmp_val[1] * 6]) + return 0; + + time64_to_tm(tmp_val[2], 0, tm); + for (int j = 0; j < tmp_val[1]; j++) { + offset = j * 6 + 3; + if (tmp_val[offset] == tas_priv->index) { + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[offset + i + 1]; + } else if (tmp_val[offset] == + TASDEVICE_CALIBRATION_REG_ADDRESS) { + for (int i = 0; i < CALIB_MAX; i++) { + buf = &data[(offset + i + 1) * 4]; + tas_priv->cali_reg_array[i] = + TASDEVICE_REG(buf[1], buf[2], + buf[3]); + } + } + tas_priv->apply_calibration(tas_priv); + } + } else { + /* + * Calibration data is in V1 format. + * struct cali_data { + * char cali_data[20]; + * } + * + * struct { + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) + * } ueft; + */ + crc = crc32(~0, data, 84) ^ ~0; + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->cali_data[i] = + tmp_val[tas_priv->index * 5 + i]; + tas_priv->apply_calibration(tas_priv); + } + } + + return 0; +} + +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct hda_codec *codec = tas_priv->codec; + int i, j, ret; + + pm_runtime_get_sync(tas_priv->dev); + guard(mutex)(&tas_priv->codec_lock); + + ret = tasdevice_spi_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + /* Add control one time only. */ + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl[tas_priv->index].name, ret); + goto out; + } + j = tas_priv->index * ARRAY_SIZE(tas2781_snd_controls) / 2; + for (i = 0; i < 3; i++) { + tas_hda->snd_ctls[i] = snd_ctl_new1(&tas2781_snd_controls[i+j], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i+tas_priv->index*3].name, + ret); + goto out; + } + } + + tasdevice_spi_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", + codec->core.subsystem_id, tas_priv->index); + ret = tasdevice_spi_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + /* Add control one time only. */ + tas_hda->dsp_prog_ctl = + snd_ctl_new1(&tas2781_dsp_prog_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_hda->dsp_conf_ctl = + snd_ctl_new1(&tas2781_dsp_conf_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_spi_prmg_load(tas_priv, 0); + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; + + /* + * If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas_priv->save_calibration(tas_priv); + +out: + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_hda->priv->dev); + pm_runtime_put_autosuspend(tas_hda->priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + struct hda_codec *codec; + int ret; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (!comp) + return -EINVAL; + + if (comp->dev) + return -EBUSY; + + codec = parent->codec; + + pm_runtime_get_sync(dev); + + comp->dev = dev; + + strscpy(comp->name, dev_name(dev), sizeof(comp->name)); + + ret = tascodec_spi_init(tas_hda->priv, codec, THIS_MODULE, + tasdev_fw_ready); + if (!ret) + comp->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component_parent *parent = master_data; + struct hda_component *comp; + + comp = hda_component_from_index(parent, tas_hda->priv->index); + if (comp && (comp->dev == dev)) { + comp->dev = NULL; + memset(comp->name, 0, sizeof(comp->name)); + comp->playback_hook = NULL; + } + + tas2781_hda_remove_controls(tas_hda); + + tasdevice_spi_config_info_remove(tas_hda->priv); + tasdevice_spi_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + component_del(tas_hda->priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_get_sync(tas_hda->priv->dev); + pm_runtime_disable(tas_hda->priv->dev); + + pm_runtime_put_noidle(tas_hda->priv->dev); + + mutex_destroy(&tas_hda->priv->codec_lock); +} + +static int tas2781_hda_spi_probe(struct spi_device *spi) +{ + struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; + const char *device_name; + int ret = 0; + + tas_hda = devm_kzalloc(&spi->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) + return -ENOMEM; + + spi->max_speed_hz = TAS2781_SPI_MAX_FREQ; + + tas_priv = devm_kzalloc(&spi->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + goto err; + tas_priv->dev = &spi->dev; + tas_hda->priv = tas_priv; + tas_priv->regmap = devm_regmap_init_spi(spi, &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate regmap: %d\n", + ret); + goto err; + } + if (strstr(dev_name(&spi->dev), "TXNW2781")) { + device_name = "TXNW2781"; + tas_priv->save_calibration = tas2781_save_calibration; + tas_priv->apply_calibration = tas2781_apply_calib; + } else { + goto err; + } + + tas_priv->irq = spi->irq; + dev_set_drvdata(&spi->dev, tas_hda); + ret = tas2781_read_acpi(tas_hda, device_name, + spi_get_chipselect(spi, 0)); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + tasdevice_spi_init(tas_priv); + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + tas2781_spi_reset(tas_priv); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component fail: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&spi->dev); + + return ret; +} + +static void tas2781_hda_spi_remove(struct spi_device *spi) +{ + tas2781_hda_remove(&spi->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + return 0; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret, val; + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + guard(mutex)(&tas_hda->priv->codec_lock); + + ret = tasdevice_spi_dev_read(tas_hda->priv, TAS2781_REG_CLK_CONFIG, + &val); + if (ret < 0) + return ret; + + if (val == TAS2781_REG_CLK_CONFIG_RESET) { + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + tas_hda->priv->cur_prog = -1; + + tasdevice_spi_prmg_load(tas_hda->priv, 0); + tas_hda->priv->apply_calibration(tas_hda->priv); + + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + } + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct spi_device_id tas2781_hda_spi_id[] = { + { "tas2781-hda", }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TXNW2781", }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct spi_driver tas2781_hda_spi_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_spi_id, + .probe = tas2781_hda_spi_probe, + .remove = tas2781_hda_spi_remove, +}; +module_spi_driver(tas2781_hda_spi_driver); + +MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); +MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c new file mode 100644 index 000000000000..9e58e8d981e3 --- /dev/null +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -0,0 +1,2114 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <asm/unaligned.h> +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> + +#include "tas2781-spi.h" + +#define OFFSET_ERROR_BIT BIT(31) + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/* should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +struct blktyp_devidx_map { + unsigned char blktyp; + unsigned char dev_idx; +}; + +/* fixed m68k compiling issue: mapping table can save code field */ +static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0x81 }, + { PRE_DEVICE_A_1X, 0x81 }, + { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0x82 }, + { PRE_DEVICE_B_1X, 0x82 }, + { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0x83 }, + { PRE_DEVICE_C_1X, 0x83 }, + { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0x84 }, + { PRE_DEVICE_D_1X, 0x84 }, + { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, + { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, +}; + +static const struct blktyp_devidx_map ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, +}; + +static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES, 0x80 }, + { MAIN_DEVICE_A, 0x81 }, + { COEFF_DEVICE_A, 0xC1 }, + { PRE_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B, 0x82 }, + { COEFF_DEVICE_B, 0xC2 }, + { PRE_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C, 0x83 }, + { COEFF_DEVICE_C, 0xC3 }, + { PRE_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D, 0x84 }, + { COEFF_DEVICE_D, 0xC4 }, + { PRE_DEVICE_D, 0xC4 }, +}; +/* + * Device support different configurations for different scene, + * like voice, music, calibration, was write in regbin file. + * Will be stored into tas_priv after regbin was loaded. + */ +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* + * In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * DSP firmware. + */ + cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + return NULL; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if ((config_offset + 64) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto config_err; + } + config_offset += 64; + } + + if ((config_offset + 4) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto config_err; + } + + /* + * convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + /* + * Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(*bk_da), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto config_err; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + goto block_err; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + goto block_err; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + i++; + goto block_err; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + + return cfg_info; +block_err: + for (int j = 0; j < i; j++) + kfree(bk_da[j]); + kfree(bk_da); +config_err: + kfree(cfg_info); + return NULL; +} + +/* Regbin file parser function. */ +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + rca = &tas_priv->rcabin; + fw_hdr = &rca->fw_hdr; + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + buf = (unsigned char *)fmw->data; + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset++]; + fw_hdr->dev_family = buf[offset++]; + fw_hdr->reserve = buf[offset++]; + fw_hdr->ndev = buf[offset++]; + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", + fw_hdr->img_sz, total_config_sz, (int)offset); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -ENOMEM; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return ret; + } + offset += (int)fw_hdr->config_size[i]; + } + + return ret; +} + +/* fixed m68k compiling issue: mapping table can save code field */ +static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block) +{ + struct blktyp_devidx_map *p = + (struct blktyp_devidx_map *)non_ppc3_mapping_table; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + int i, n = ARRAY_SIZE(non_ppc3_mapping_table); + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; + n = ARRAY_SIZE(ppc3_tas2781_mapping_table); + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + p = (struct blktyp_devidx_map *)ppc3_mapping_table; + n = ARRAY_SIZE(ppc3_mapping_table); + } + + for (i = 0; i < n; i++) { + if (block->type == p[i].blktyp) { + dev_idx = p[i].dev_idx; + break; + } + } + + return dev_idx; +} + +/* Block parser function. */ +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host. + */ + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset++]; + block->pchksum = data[offset++]; + block->is_ychksum_present = data[offset++]; + block->ychksum = data[offset++]; + block->blk_size = get_unaligned_be32(&data[offset]); + offset += 4; + block->nr_subblocks = get_unaligned_be32(&data[offset]); + offset += 4; + + /* + * Fixed m68k compiling issue: + * 1. mapping table can save code field. + * 2. storing the dev_idx as a member of block can reduce unnecessary + * time and system resource comsumption of dev_idx mapping every + * time the block data writing to the dsp. + */ + block->dev_idx = map_dev_idx(tas_fmw, block); + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += block->blk_size; + + return offset; +} + +/* Data of block parser function. */ +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + img_data->nr_blk = get_unaligned_be32(&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data_kernel( + tas_fmw, blk, fmw, offset); + if (offset < 0) { + kfree(img_data->dev_blks); + return -EINVAL; + } + } + + return offset; +} + +/* Data of DSP program parser function. */ +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* skip 72 unused byts */ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &program->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* Data of DSP configurations parser function. */ +static int fw_parse_configuration_data_kernel(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* skip extra 16 bytes */ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* DSP firmware file header parser function for early PPC3 firmware binary. */ +static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_config *config; + struct tasdevice_prog *program; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + return -EINVAL; + } + offset += 2; + fw_hdr->device = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 2; + + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || + tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + return -EINVAL; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(*tas_fmw->programs), GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &tas_fmw->programs[i]; + program->prog_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); + offset += 4; + + /* + * The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) { + kfree(tas_fmw->programs); + return -ENOMEM; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &tas_fmw->configs[i]; + config->cfg_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + + return offset; +} + +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform single byte write to device. + */ +static int tasdevice_single_byte_wr(void *context, int dev_idx, + unsigned char *data, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int i, subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, "process_block: Out of boundary\n"); + return 0; + } + + for (i = 0; i < len; i++) { + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + dev_err(tas_priv->dev, + "process_block: single write error\n"); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 4; + } + + return subblk_offset; +} +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform multi-write to device. + */ +static int tasdevice_burst_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (len % 4) { + dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", + __func__, len); + subblk_offset |= OFFSET_ERROR_BIT; + } + + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], + data[subblk_offset + 2]), + &data[subblk_offset + 4], len); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += (len + 4); + + return subblk_offset; +} +/* Just delay for ms.*/ +static int tasdevice_delay(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned int sleep_time, subblk_offset = 2; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, "%s: delay Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + sleep_time = get_unaligned_be16(&data[2]) * 1000; + fsleep(sleep_time); + } + subblk_offset += 2; + + return subblk_offset; +} +/* + * In sub-block data, have three type sub-block: + * 1. Single byte write. + * 2. Multi-byte write. + * 3. Delay. + * 4. Bits update. + * This function perform bits update. + */ +static int tasdevice_field_wr(void *context, int dev_idx, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int rc, subblk_offset = 2; + + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (dev_idx == (tas_priv->index + 1) || dev_idx == 0) { + rc = tasdevice_spi_dev_update_bits(tas_priv, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], + data[subblk_offset + 5]); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: update_bits error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + } + subblk_offset += 6; + + return subblk_offset; +} +/* Data block process function. */ +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int blktyp = dev_idx & 0xC0, subblk_offset; + unsigned char subblk_typ = data[1]; + + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: + subblk_offset = tasdevice_single_byte_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_BURST: + subblk_offset = tasdevice_burst_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_DELAY: + subblk_offset = tasdevice_delay(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + case TASDEVICE_CMD_FIELD_W: + subblk_offset = tasdevice_field_wr(tas_priv, + dev_idx & 0x4f, data, sublocksize); + break; + default: + subblk_offset = 2; + break; + } + if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + } else + tas_priv->cur_conf = -1; + } + subblk_offset &= ~OFFSET_ERROR_BIT; + + return subblk_offset; +} +/* + * Device support different configurations for different scene, + * this function was used for choose different config. + */ +void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = pContext; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + unsigned int j, k; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < blk_data[j]->n_subblks; k++) { + tas_priv->is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} +/* Block process function. */ +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + const unsigned int blk_size = block->blk_size; + unsigned char *data = block->data; + unsigned int i, length; + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + block->dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} +/* DSP firmware file header parser function. */ +static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, + struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + offset += len; + + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + return -EINVAL; + } + offset += 4; + + fw_hdr->device = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 4; + fw_hdr->ndev = 1; + + return offset; +} + +/* DSP firmware file header parser function for size variabled header. */ +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + + return offset; +} + +/* DSP firmware file block parser function. */ +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + return -EINVAL; + } + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + return -EINVAL; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = get_unaligned_be32(&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += n; + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i, n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + return -EINVAL; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + return -EINVAL; + } + offset += n; + img_data->nr_blk = get_unaligned_be16(&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(*img_data->dev_blks), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &img_data->dev_blks[i]; + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) + return -EINVAL; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /* Not error in calibration Data file, return directly */ + dev_dbg(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + return offset; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), + GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &tas_fmw->programs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &program->dev_data, fmw, + offset); + if (offset < 0) + return offset; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /* Not error for calibration Data file, return directly */ + return offset; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) + return -ENOMEM; + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &tas_fmw->configs[i]; + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + offset += n; + offset = fw_parse_data(tas_fmw, &config->dev_data, + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +/* yram5 page check. */ +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +/* DSP firmware yram block check. */ +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) { + in = check_inpage_yram_rg(cd, reg, len); + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) + in = check_inpage_yram_bk1(cd, page, reg, len); + else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} +/* yram4 page check. */ +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} +/* yram page check. */ +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (!in) + in = check_inblock_yram(cd, book, page, reg, len); + + return in; +} + +/* Checksum for data block. */ +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0, i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (len == 4)) { + /* DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_spi_dev_bulk_read(tasdevice, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) >= + TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) <= + (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + /* DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +/* Checksum for single register. */ +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + /* DSP swap command, pass */ + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + return 0; + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + return 0; + ret = tasdevice_spi_dev_read(tasdevice, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + return ret; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->err_code |= ERROR_YRAM_CRCCHK; + return -EAGAIN; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + + return ret; +} + +/* Block type check. */ +static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || + (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || + (type == MAIN_DEVICE_D)) + p->cur_prog = -1; + else + p->cur_conf = -1; +} +/* Checksum for data bytes. */ +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, book, page, reg, val); + + if (ret > 0) { + *crc_chksum += ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, tas_priv); + +end: + return ret; +} +/* Multi-data byte write. */ +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + + return ret; +} +/* Checksum for block. */ +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); + set_err_prg_cfg(block->type, tas_priv); + return ret; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, tas_priv); + } else { + tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; + } + + return ret; +} +/* Firmware block load function. */ +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int sleep_time, len, nr_cmds; + unsigned char offset, book, page, val; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_CHECKSUM, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /* Single byte write */ + if (offset <= 0x7F) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + break; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /* sleep command */ + if (offset == 0x81) { + /* book -- data[0] page -- data[1] */ + sleep_time = ((book << 8) + page)*1000; + fsleep(sleep_time); + continue; + } + /* Multiple bytes write */ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /* skip current blk */ + break; + } + + return ret; +} +/* Firmware block load function. */ +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int ret = 0; + + block->nr_retry = 6; + if (tas_priv->is_loading == false) + return 0; + ret = tasdev_load_blk(tas_priv, block); + if (ret < 0) + dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); + + return ret; +} + +/* + * Select firmware binary parser & load callback functions by ppc3 version + * and firmware binary version. + */ +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} +/* Calibration block load function. */ +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware binary file header parser function. */ +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &tas_fmw->fw_hdr; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &fw_hdr->fixed_hdr; + static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); + offset += 72; + +out: + return offset; +} + +/* DSP firmware binary file parser function. */ +static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + return -EINVAL; + } + + tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) + return -EINVAL; + + fw_fixed_hdr = &tas_fmw->fw_hdr.fixed_hdr; + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + case 0x503: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + return ret; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + + return ret; +} +/* DSP firmware binary file parser function. */ +int tasdevice_spi_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + return ret; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + + return ret; +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &prog->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +/* DSP firmware program block data remove function. */ +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +/* DSP firmware config block data remove function. */ +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &cfg->dev_data; + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &tas_dt->dev_blks[i]; + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +/* DSP firmware config remove function. */ +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +/* DSP firmware remove function. */ +void tasdevice_spi_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = context; + + if (!tas_dev->fmw) + return; + + if (tas_dev->fmw->programs) + tasdev_dsp_prog_remove(tas_dev->fmw->programs, + tas_dev->fmw->nr_programs); + if (tas_dev->fmw->configs) + tasdev_dsp_cfg_remove(tas_dev->fmw->configs, + tas_dev->fmw->nr_configurations); + kfree(tas_dev->fmw); + tas_dev->fmw = NULL; +} + +/* DSP firmware calibration data remove function. */ +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &tas_fmw->calibrations[i]; + if (!calibration) + continue; + + if (!calibration->dev_data.dev_blks) + continue; + + for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { + block = &calibration->dev_data.dev_blks[blks]; + if (!block) + continue; + kfree(block->data); + } + kfree(calibration->dev_data.dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +/* Calibration data from firmware remove function. */ +void tasdevice_spi_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + + if (tas_priv->cali_data_fmw) { + tas2781_clear_calfirmware(tas_priv->cali_data_fmw); + tas_priv->cali_data_fmw = NULL; + } +} + +/* Configuration remove function. */ +void tasdevice_spi_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **ci = rca->cfg_info; + unsigned int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} + +/* DSP firmware program block data load function. */ +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &dev_data->dev_blks[i]; + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +/* DSP firmware program load function. */ +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &tas_priv->rcabin; + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (rca_conf_no >= rca->ncfgs || + rca_conf_no < 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + return 0; + } + + if (prm_no >= 0 && + (tas_priv->cur_prog != prm_no || + tas_priv->force_fwload_status)) { + tas_priv->cur_conf = -1; + tas_priv->is_loading = true; + prog_status++; + } + + if (prog_status) { + program = &tas_fmw->programs[prm_no]; + tasdevice_load_data(tas_priv, &program->dev_data); + if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &cal->dev_data); + } + tas_priv->cur_prog = prm_no; + } + + } + + if (cfg_no && + (cfg_no != tas_priv->cur_conf) && + (tas_priv->is_loaderr == false)) { + status++; + tas_priv->is_loading = true; + } else { + tas_priv->is_loading = false; + } + if (status) { + conf = &tas_fmw->configs[cfg_no]; + status = 0; + tasdevice_load_data(tas_priv, &conf->dev_data); + if (tas_priv->is_loaderr == true) { + status |= 1 << 4; + } else if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + tas_priv->cur_conf = cfg_no; + } + } else { + dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", + __func__, cfg_no); + } + + return prog_status; +} + +/* DSP firmware program load interface function. */ +int tasdevice_spi_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (prm_no >= 0) { + tas_priv->cur_conf = 0; + tas_priv->is_loading = true; + prog_status++; + } + if (prog_status) { + program = &tas_fmw->programs[prm_no]; + tasdevice_load_data(tas_priv, &program->dev_data); + tas_priv->cur_prog = prm_no; + } + if (prog_status) { + conf = &tas_fmw->configs[tas_priv->cur_conf]; + tasdevice_load_data(tas_priv, &conf->dev_data); + } + + return prog_status; +} + +/* DSP firmware program switch interface function. */ +void tasdevice_spi_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = context; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else { + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } +} -- 2.40.1

1 0

[PATCH] sound: hda: tas2781: mark const variables as __maybe_unused
by Arnd Bergmann 19 Jul '24

19 Jul '24

From: Arnd Bergmann <arnd(a)arndb.de> An earlier patch changed the DECLARE_TLV_DB_SCALE declaration, but now there are additional static const variables that cause the same build warnings: In file included from sound/pci/hda/tas2781_hda_i2c.c:23: include/sound/tas2781-tlv.h:23:28: error: 'tas2563_dvc_table' defined but not used [-Werror=unused-const-variable=] 23 | static const unsigned char tas2563_dvc_table[][4] = { | ^~~~~~~~~~~~~~~~~ In file included from include/sound/tlv.h:10, from sound/pci/hda/tas2781_hda_i2c.c:22: include/sound/tas2781-tlv.h:20:35: error: 'tas2563_dvc_tlv' defined but not used [-Werror=unused-const-variable=] 20 | static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); | ^~~~~~~~~~~~~~~ Mark them all as unused as well. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- include/sound/tas2781-tlv.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..00fd4d449ff3 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -16,11 +16,11 @@ #define __TAS2781_TLV_H__ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); -static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); +static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.39.2

2 1

[PATCH] ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
by wangdicheng 19 Jul '24

19 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> Sonix HD USB Camera does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x84". This patch adds the USB ID to quirks.c and avoids those error messages. (snip) [1.789698] usb 3-3: new high-speed USB device number 2 using xhci_hcd [1.984121] usb 3-3: New USB device found, idVendor=0c45, idProduct=6340, bcdDevice= 0.00 [1.984124] usb 3-3: New USB device strings: Mfr=2, Product=1, SerialNumber=0 [1.984127] usb 3-3: Product: USB 2.0 Camera [1.984128] usb 3-3: Manufacturer: Sonix Technology Co., Ltd. [5.440957] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.130679] usb 3-3: 3:1: cannot get freq at ep 0x84 [12.175065] usb 3-3: 3:1: cannot get freq at ep 0x84 Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..c1fd690cbac0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2033,6 +2033,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x0499, 0x3108, /* Yamaha YIT-W12TX */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x0c45, 0x6340, /* Sonix HD USB Camera */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04d8, 0xfeea, /* Benchmark DAC1 Pre */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x04e8, 0xa051, /* Samsung USBC Headset (AKG) */ -- 2.25.1

1 0

[PATCH v3] ALSA: usb-audio: Fix microphone sound on HD webcam.
by wangdicheng 19 Jul '24

19 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> I own an external usb Webcam, HD webcam, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, After 3 failures. All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Testing patch provides consistent good sound recording quality and volume range. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v2 -> v3: make the mail address of Signed-off-by consistent with the From address V1 -> V2: align the space,Update code to v6.8-rc3,and make modifications based on it --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..d1bdb0b93bda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1211,6 +1211,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..f222532d4d43 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2185,6 +2185,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

1 0

Help on specifying 'chmap' in asound.conf
by Xinhui Zhou 19 Jul '24

19 Jul '24

Dear all, I am having a question regarding how to specify 'champ' for a plugin. I do not see any examples of this. I tried many ways to specify but failed. As indicated by the link below, I can specify the MAP as a string array. Can someone provide one example on how exactly this [chmap MAP] can be specified? " pcm.name { type cras [chmap MAP] # Provide channel maps; MAP is a string array } " https://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html I can something like these, but none of these work for me. chmap LFE chmap "LFE" chmap FR,FL,LFE champ "FR,FL,LFE" Thanks! Xinhui,

3 3

[PATCH alsa-utils 1/2] aplaymidi2: Add -a option to pass all UMP packets
by Takashi Iwai 19 Jul '24

19 Jul '24

So far, aplaymidi2 passes the MIDI1/MIDI2 channel voice UMP messages to the target while processing other UMP messages internally. But sometimes we'd like to pass all UMP messages as is and let the receiver processes. This patch adds a new option -a (or --passall) to pass the all UMP packets included in the given MIDI Clip file to the target as-is. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- seq/aplaymidi2/aplaymidi2.1 | 8 ++++++++ seq/aplaymidi2/aplaymidi2.c | 20 +++++++++++++++----- 2 files changed, 23 insertions(+), 5 deletions(-) diff --git a/seq/aplaymidi2/aplaymidi2.1 b/seq/aplaymidi2/aplaymidi2.1 index fc85943c0d38..681796222914 100644 --- a/seq/aplaymidi2/aplaymidi2.1 +++ b/seq/aplaymidi2/aplaymidi2.1 @@ -67,6 +67,14 @@ Default is 2 seconds. .I \-s, \-\-silent Don't show message texts. +.TP +.I \-a, \-\-passall +Pass all UMP packets as is. + +As default, \fBaplaymidi2\fP passes only MIDI1 and MIDI2 channel voice +messages and process other UMP packets internally. +With this option, it passes all UMP packets to the target. + .SH SEE ALSO pmidi(1) .br diff --git a/seq/aplaymidi2/aplaymidi2.c b/seq/aplaymidi2/aplaymidi2.c index 6a1f21e92444..f5dfdbd91f30 100644 --- a/seq/aplaymidi2/aplaymidi2.c +++ b/seq/aplaymidi2/aplaymidi2.c @@ -21,6 +21,7 @@ static snd_seq_addr_t ports[16]; static int queue; static int end_delay = 2; static int silent; +static int passall; static unsigned int _current_tempo = 50000000; /* default 120 bpm */ static unsigned int tempo_base = 10; @@ -411,6 +412,9 @@ static void play_midi(FILE *file) while ((len = read_ump_packet(file, ump)) > 0) { const snd_ump_msg_hdr_t *h = (snd_ump_msg_hdr_t *)ump; + if (passall) + send_ump(ump, len); + if (h->type == SND_UMP_MSG_TYPE_UTILITY) { const snd_ump_msg_utility_t *uh = (const snd_ump_msg_utility_t *)ump; @@ -448,9 +452,10 @@ static void play_midi(FILE *file) end_clip(); continue; } - } else if (h->type == SND_UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE || - h->type == SND_UMP_MSG_TYPE_DATA || - h->type == SND_UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE) { + } else if (!passall && + (h->type == SND_UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE || + h->type == SND_UMP_MSG_TYPE_DATA || + h->type == SND_UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE)) { send_ump(ump, len); } } @@ -496,7 +501,8 @@ static void usage(const char *argv0) "-V, --version print current version\n" "-p, --port=client:port,... set port(s) to play to\n" "-d, --delay=seconds delay after song ends\n" - "-s, --silent don't show texts\n", + "-s, --silent don't show texts\n" + "-a, --passall pass all UMP packets as-is\n", argv0); } @@ -513,13 +519,14 @@ int main(int argc, char *argv[]) {"port", 1, NULL, 'p'}, {"delay", 1, NULL, 'd'}, {"silent", 0, NULL, 's'}, + {"passall", 0, NULL, 'a'}, {0} }; int c; init_seq(); - while ((c = getopt_long(argc, argv, "hVp:d:s", + while ((c = getopt_long(argc, argv, "hVp:d:sa", long_options, NULL)) != -1) { switch (c) { case 'h': @@ -537,6 +544,9 @@ int main(int argc, char *argv[]) case 's': silent = 1; break; + case 'a': + passall = 1; + break; default: usage(argv[0]); return 1; -- 2.43.0

1 0

[PATCH] ASoC: sof: amd: fix for firmware reload failure in Vangogh platform
by Venkata Prasad Potturu 18 Jul '24

18 Jul '24

Setting ACP ACLK as clock source when ACP enters D0 state causing firmware load failure, as per design clock source should be internal clock. Remove acp_clkmux_sel field so that ACP will use internal clock source when ACP enters into D0 state. Fixes: d0dab6b76a9f ("ASoC: SOF: amd: Add sof support for vangogh platform") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/pci-vangogh.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index 16eb2994fbab..eba580840100 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -34,7 +34,6 @@ static const struct sof_amd_acp_desc vangogh_chip_info = { .dsp_intr_base = ACP5X_DSP_SW_INTR_BASE, .sram_pte_offset = ACP5X_SRAM_PTE_OFFSET, .hw_semaphore_offset = ACP5X_AXI2DAGB_SEM_0, - .acp_clkmux_sel = ACP5X_CLKMUX_SEL, .probe_reg_offset = ACP5X_FUTURE_REG_ACLK_0, }; -- 2.39.2

2 1

[PATCH v2] ALSA: usb-audio: Fix microphone sound on HD webcam.
by wangdicheng 18 Jul '24

18 Jul '24

I own an external usb Webcam, HD webcam, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, After 3 failures. All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Testing patch provides consistent good sound recording quality and volume range. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- V1 -> V2: align the space,Update code to v6.8-rc3,and make modifications based on it --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..d1bdb0b93bda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1211,6 +1211,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..f222532d4d43 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2185,6 +2185,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

1 0

[PATCH v1] ALSA: hda/tas2781: Add new quirk for Lenovo Hera2 Laptop
by Shenghao Ding 18 Jul '24

18 Jul '24

Add new vendor_id and subsystem_id in quirk for Lenovo Hera2 Laptop. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.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 766f0b1d3e9d..bac68a84fd50 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10539,6 +10539,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x231e, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), SND_PCI_QUIRK(0x17aa, 0x231f, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), + SND_PCI_QUIRK(0x17aa, 0x2326, "Hera2", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.34.1

2 1

[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800
by edmund.raile 18 Jul '24

18 Jul '24

On kernels since 5.14.0, ALSA playback to the FireWire RME Fireface 800 audio interface results in a deadlock involving snd_pcm_period_elapsed(), freezing the system. On kernels 5.0.0 to 5.13.19 the interface works just fine, as it does with the RME driver. Distributions tested: Ubuntu Manjaro Arch Fedora FireWire chipsets tested: LSI FW643 TI XIO2213B Platforms tested: Intel i5 4570 on AsRock H97 Pro4 Intel i5 12600K on MSI MS-7D25 The behavior was also observed on the RME forum: https://forum.rme-audio.de/viewtopic.php?pid=190472#p190472 Shortened traces of 6.10.0-rc7 (Arch linux-mainline): RIP: 0010:tasklet_unlock_spin_wait (./arch/x86/include/asm/bitops.h:206 ./arch/x86/include/asm/bitops.h:238 ./include/asm-generic/bitops/instrumented-non-atomic.h:142 kernel/softirq.c:851) <NMI> ? watchdog_hardlockup_check.cold (kernel/watchdog.c:200) ? __perf_event_overflow (kernel/events/core.c:9737 (discriminator 2)) ? handle_pmi_common (arch/x86/events/intel/core.c:3061 (discriminator 1)) ? intel_pmu_handle_irq (./arch/x86/include/asm/paravirt.h:192 arch/x86/events/intel/core.c:2428 arch/x86/events/intel/core.c:3127) ? perf_event_nmi_handler (arch/x86/events/core.c:1744 arch/x86/events/core.c:1730) ? nmi_handle (arch/x86/kernel/nmi.c:151) ? default_do_nmi (arch/x86/kernel/nmi.c:352 (discriminator 61)) ? exc_nmi (arch/x86/kernel/nmi.c:546) ? end_repeat_nmi (arch/x86/entry/entry_64.S:1408) ? tasklet_unlock_spin_wait (./arch/x86/include/asm/bitops.h:206 ./arch/x86/include/asm/bitops.h:238 ./include/asm-generic/bitops/instrumented-non-atomic.h:142 kernel/softirq.c:851) ? tasklet_unlock_spin_wait (./arch/x86/include/asm/bitops.h:206 ./arch/x86/include/asm/bitops.h:238 ./include/asm-generic/bitops/instrumented-non-atomic.h:142 kernel/softirq.c:851) ? tasklet_unlock_spin_wait (./arch/x86/include/asm/bitops.h:206 ./arch/x86/include/asm/bitops.h:238 ./include/asm-generic/bitops/instrumented-non-atomic.h:142 kernel/softirq.c:851) </NMI> <TASK> ohci_flush_iso_completions (./include/linux/interrupt.h:740 drivers/firewire/ohci.c:3530) firewire_ohci amdtp_domain_stream_pcm_pointer (sound/firewire/amdtp-stream.c:1858) snd_firewire_lib snd_pcm_update_hw_ptr0 (sound/core/pcm_lib.c:304) snd_pcm snd_pcm_status64 (sound/core/pcm_native.c:1034) snd_pcm snd_pcm_status_user64 (./include/linux/uaccess.h:191 sound/core/pcm_native.c:1096) snd_pcm snd_pcm_ioctl (sound/core/pcm_native.c:3401 (discriminator 1)) snd_pcm __x64_sys_ioctl (fs/ioctl.c:51 fs/ioctl.c:907 fs/ioctl.c:893 fs/ioctl.c:893) do_syscall_64 (arch/x86/entry/common.c:52 (discriminator 1) arch/x86/entry/common.c:83 (discriminator 1)) ? snd_pcm_status_user64 (sound/core/pcm_native.c:1096 (discriminator 1)) snd_pcm ? futex_wake (kernel/futex/waitwake.c:173) ? do_futex (kernel/futex/syscalls.c:107 (discriminator 1)) ? __x64_sys_futex (kernel/futex/syscalls.c:179 kernel/futex/syscalls.c:160 kernel/futex/syscalls.c:160) ? syscall_exit_to_user_mode (kernel/entry/common.c:221) ? do_syscall_64 (./arch/x86/include/asm/cpufeature.h:178 arch/x86/entry/common.c:98) ? do_futex (kernel/futex/syscalls.c:107 (discriminator 1)) ? __x64_sys_futex (kernel/futex/syscalls.c:179 kernel/futex/syscalls.c:160 kernel/futex/syscalls.c:160) ? syscall_exit_to_user_mode (kernel/entry/common.c:221) ? do_syscall_64 (./arch/x86/include/asm/cpufeature.h:178 arch/x86/entry/common.c:98) ? syscall_exit_to_user_mode (kernel/entry/common.c:221) ? do_syscall_64 (./arch/x86/include/asm/cpufeature.h:178 arch/x86/entry/common.c:98) ? do_syscall_64 (./arch/x86/include/asm/cpufeature.h:178 arch/x86/entry/common.c:98) ? __irq_exit_rcu (kernel/softirq.c:620 (discriminator 1) kernel/softirq.c:639 (discriminator 1)) entry_SYSCALL_64_after_hwframe (arch/x86/entry/entry_64.S:130) RIP: 0010:native_queued_spin_lock_slowpath (kernel/locking/qspinlock.c:380 (discriminator 3)) <NMI> ? watchdog_hardlockup_check.cold (kernel/watchdog.c:200) ? __perf_event_overflow (kernel/events/core.c:9737 (discriminator 2)) ? handle_pmi_common (arch/x86/events/intel/core.c:3061 (discriminator 1)) ? intel_pmu_handle_irq (./arch/x86/include/asm/paravirt.h:192 arch/x86/events/intel/core.c:2428 arch/x86/events/intel/core.c:3127) ? perf_event_nmi_handler (arch/x86/events/core.c:1744 arch/x86/events/core.c:1730) ? nmi_handle (arch/x86/kernel/nmi.c:151) ? default_do_nmi (arch/x86/kernel/nmi.c:352 (discriminator 61)) ? exc_nmi (arch/x86/kernel/nmi.c:546) ? end_repeat_nmi (arch/x86/entry/entry_64.S:1408) ? native_queued_spin_lock_slowpath (kernel/locking/qspinlock.c:380 (discriminator 3)) ? native_queued_spin_lock_slowpath (kernel/locking/qspinlock.c:380 (discriminator 3)) ? native_queued_spin_lock_slowpath (kernel/locking/qspinlock.c:380 (discriminator 3)) </NMI> <IRQ> _raw_spin_lock_irqsave (./arch/x86/include/asm/paravirt.h:584 ./arch/x86/include/asm/qspinlock.h:51 ./include/asm-generic/qspinlock.h:114 ./include/linux/spinlock.h:187 ./include/linux/spinlock_api_smp.h:111 kernel/locking/spinlock.c:162) snd_pcm_period_elapsed (sound/core/pcm_lib.c:1905) snd_pcm process_rx_packets (sound/firewire/amdtp-stream.c:1164) snd_firewire_lib irq_target_callback (sound/firewire/amdtp-stream.c:1549) snd_firewire_lib handle_it_packet (drivers/firewire/ohci.c:2786 drivers/firewire/ohci.c:2974) firewire_ohci context_tasklet (drivers/firewire/ohci.c:1127) firewire_ohci tasklet_action_common.isra.0 (kernel/softirq.c:789) handle_softirqs (kernel/softirq.c:554) __irq_exit_rcu (kernel/softirq.c:589 kernel/softirq.c:428 kernel/softirq.c:637) common_interrupt (arch/x86/kernel/irq.c:278 (discriminator 35)) </IRQ> <TASK> asm_common_interrupt (./arch/x86/include/asm/idtentry.h:693) It can be induced by direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence ranges from two seconds to 30 minutes. Loading the CPU appears to increase the likelihood: stress --cpu $(nproc) So does switching between applications via workspaces (only tested Xfce). The regression has been traced to these two commits: 7ba5ca32fe6e8d2e153fb5602997336517b34743 b5b519965c4c364ae65c49fe9f11d222dd70a9c2 I am currently testing a simple patch, in essence reverting both commits. Behaves well so far (stable), will likely send it in tomorrow. #regzbot introduced: 7ba5ca32fe6e

2 2

ThinkPad P14s Gen 5 AMD micmute LED is not working sometimes.
by GitHub issues - opened 18 Jul '24

18 Jul '24

alsa-project/alsa-ucm-conf issue #433 was opened from pma1: The mic and sound output works fine. The mute LED works fine. But when press Fn+F4, the mic-mute led is not working sometimes after bootup. The failrate is like 1/10. Tested with latest alsa-ucm from git, it's reproduced. /sys/class/sound/ctl-led/mic/card1# cat list 7 # amixer -c 1 controls numid=12,iface=CARD,name='Headphone Jack' numid=11,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Phantom Jack' numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=2,iface=MIXER,name='Headphone Playback Switch' numid=1,iface=MIXER,name='Headphone Playback Volume' numid=16,iface=MIXER,name='PCM Playback Volume' numid=17,iface=MIXER,name='Mic ACP LED Capture Switch' numid=8,iface=MIXER,name='Mic Boost Volume' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=5,iface=MIXER,name='Auto-Mute Mode' numid=4,iface=MIXER,name='Speaker Playback Switch' numid=3,iface=MIXER,name='Speaker Playback Volume' numid=15,iface=PCM,name='Capture Channel Map' numid=14,iface=PCM,name='Playback Channel Map' # echo 'Capture Switch' > detach # echo 'Mic ACP LED Capture Switch' > attach /sys/class/sound/ctl-led/mic/card1# cat list 17 Then the micmute LED will work. [alsa-info-amd-micmutled.txt](https://github.com/user-attachments/files/1627… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/433 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Inconsistent order of channels on C-Media 7.1 channel device
by victor＠allumeenergy.com.au 18 Jul '24

18 Jul '24

I have a USB sound device (Startech ICUSBAUDIO7D, which comes up as `ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device` through lsusb) which I'm trying to program the output of for 8 channels. The python code (outputs a 50Hz sine wave) I'm using is below (each channel gets a different amplitude so I can identify them with an oscilloscope). The sound card has 4 stereo jacks, to which I have a scope connected. The problem is that on different runs of the program, the output mappings are not consistent - each run of the program might have the order of the channels different to the previous. So running it one time might have amplitudes (0, 0.125) on the first jack, but the next run it might have those (0.75, 0.25) on the first jack (it doesn't appear to be 'paired' per jack) I initially discovered this when using portaudio, and decided to try using alsa-only and the issue persists. Is this likely to be a problem with the driver, alsa-lib, or is there something else going on here? ``` import numpy as np import alsaaudio def sine_wave(amplitude=1, freq=50, sample_rate=48000): wave_len = sample_rate/freq samples = (amplitude * 32767 * np.sin(2 * np.pi * np.arange(wave_len) * freq / sample_rate)).astype(np.int16) return samples def eightchan(freq=50, sample_rate=48000): wave_len = sample_rate//freq frames = np.zeros((wave_len, 8), dtype=np.int16) samples = sine_wave() for c in range(8): frames[:wave_len,c] = samples[:wave_len]*(c/8) return frames.tobytes() if __name__ == '__main__': pcm = alsaaudio.PCM(device='surround71:CARD=ICUSBAUDIO7D,DEV=0', format=alsaaudio.PCM_FORMAT_S16_LE, channels=8, rate=48000, type=alsaaudio.PCM_PLAYBACK, periodsize=960, periods=1) frames = eightchan() while True: pcm.write(frames) ``` alsa-info output is attached [alsa-info.txt](https://github.com/user-attachments/files/16277691/alsa-info…

1 1

Inconsistent order of channels on C-Media 7.1 channel device
by GitHub issues - opened 18 Jul '24

18 Jul '24

alsa-project/alsa-lib issue #402 was opened from victorallume: I have a USB sound device (Startech ICUSBAUDIO7D, which comes up as `ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device` through lsusb) which I'm trying to program the output of for 8 channels. The python code (outputs a 50Hz sine wave) I'm using is below (each channel gets a different amplitude so I can identify them with an oscilloscope). The sound card has 4 stereo jacks, to which I have a scope connected. The problem is that on different runs of the program, the output mappings are not consistent - each run of the program might have the order of the channels different to the previous. So running it one time might have amplitudes (0, 0.125) on the first jack, but the next run it might have those (0.75, 0.25) on the first jack (it doesn't appear to be 'paired' per jack) I initially discovered this when using portaudio, and decided to try using alsa-only and the issue persists. Is this likely to be a problem with the driver, alsa-lib, or is there something else going on here? ``` import numpy as np import alsaaudio def sine_wave(amplitude=1, freq=50, sample_rate=48000): wave_len = sample_rate/freq samples = (amplitude * 32767 * np.sin(2 * np.pi * np.arange(wave_len) * freq / sample_rate)).astype(np.int16) return samples def eightchan(freq=50, sample_rate=48000): wave_len = sample_rate//freq frames = np.zeros((wave_len, 8), dtype=np.int16) samples = sine_wave() for c in range(8): frames[:wave_len,c] = samples[:wave_len]*(c/8) return frames.tobytes() if __name__ == '__main__': pcm = alsaaudio.PCM(device='surround71:CARD=ICUSBAUDIO7D,DEV=0', format=alsaaudio.PCM_FORMAT_S16_LE, channels=8, rate=48000, type=alsaaudio.PCM_PLAYBACK, periodsize=960, periods=1) frames = eightchan() while True: pcm.write(frames) ``` alsa-info output is attached [alsa-info.txt](https://github.com/user-attachments/files/16277691/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/402 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: Fix microphone sound on HD webcam.
by wangdicheng 17 Jul '24

17 Jul '24

I own an external usb Webcam, HD webcam, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, After 3 failures. All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Testing patch provides consistent good sound recording quality and volume range. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- v2 -> v3: Update code to v6.8-rc3,and make modifications based on it V1 -> V2: align the space --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..d1bdb0b93bda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1211,6 +1211,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606..f222532d4d43 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2185,6 +2185,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), + DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

1 0

[PATCH] ALSA: usb-audio: Fix microphone sound on HD webcam.
by wangdicheng 17 Jul '24

17 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> I own an external usb Webcam, HD webcam, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, After 3 failures. All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Testing patch provides consistent good sound recording quality and volume range. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- V1 -> V2: align the space --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..d1bdb0b93bda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1211,6 +1211,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4e64842245e1..7cd13e4fda4f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2179,6 +2179,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

1 0

[PATCH] ALSA: usb-audio: Fix microphone sound on HD webcam.
by wangdicheng 17 Jul '24

17 Jul '24

From: wangdicheng <wangdicheng(a)kylinos.cn> I own an external usb Webcam, HD webcam, which had low mic volume and inconsistent sound quality. Video works as expected. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 Set up quirk cval->res to 16 for 256 levels, Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. Confirmed that happened anyway later due to the backoff mechanism, After 3 failures. All audio stream on device interfaces share the same values, apart from wMaxPacketSize and tSamFreq : bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 0x0001 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 1 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0064 1x 100 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioStreaming Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0x0000 (snip) Testing patch provides consistent good sound recording quality and volume range. (snip) [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- V1 -> V2: align the space --- sound/usb/mixer.c | 7 +++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 9 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 409fc1164694..d1bdb0b93bda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1211,6 +1211,13 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 16; } break; + case USB_ID(0x1bcf, 0x2281): /* HD Webcam */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + usb_audio_info(chip, + "set resolution quirk: cval->res = 16\n"); + cval->res = 16; + } + break; } } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4e64842245e1..7cd13e4fda4f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2179,6 +2179,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_FIXED_RATE), DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1bcf, 0x2281, /* HD Webcam */ + QUIRK_FLAG_GET_SAMPLE_RATE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ -- 2.25.1

1 0

[PATCH] ALSA: pcm_dmaengine: Don't synchronize DMA channel when DMA is paused
by Shengjiu Wang 17 Jul '24

17 Jul '24

When suspended, the DMA channel may enter PAUSE state if dmaengine_pause() is supported by DMA. At this state, dmaengine_synchronize() should not be called, otherwise the DMA channel can't be resumed successfully. Fixes: e8343410ddf0 ("ALSA: dmaengine: Synchronize dma channel after drop()") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/core/pcm_dmaengine.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index b54336d4bffc..b134a51b3fd5 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -352,8 +352,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan); int snd_dmaengine_pcm_sync_stop(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; - dmaengine_synchronize(prtd->dma_chan); + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status != DMA_PAUSED) + dmaengine_synchronize(prtd->dma_chan); return 0; } -- 2.34.1

2 1

Re: [PATCH] ALSA: usb-audio: Fix microphone sound on HD webcam.
by Takashi Iwai 17 Jul '24

17 Jul '24

On Wed, 17 Jul 2024 09:51:49 +0200, wangdicheng wrote: > > From: wangdicheng <wangdicheng(a)kylinos.cn> > > I own an external usb Webcam, HD webcam, which had low mic volume and > inconsistent sound quality. Video works as expected. > > (snip) > [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd > [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 > [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam > [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co > [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 > [ 96.092610][ 2] [ T3680] usb 5-2.2: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. > [ 96.102436][ 2] [ T3680] usb 5-2.2: [5] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 > > Set up quirk cval->res to 16 for 256 levels, > Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate. > Confirmed that happened anyway later due to the backoff mechanism, > After 3 failures. > > All audio stream on device interfaces share the same values, > apart from wMaxPacketSize and tSamFreq : > > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 4 > bNumEndpoints 1 > bInterfaceClass 1 Audio > > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 4 > bNumEndpoints 1 > bInterfaceClass 1 Audio > bInterfaceSubClass 2 Streaming > bInterfaceProtocol 0 > iInterface 0 > AudioStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (AS_GENERAL) > bTerminalLink 3 > bDelay 1 frames > wFormatTag 0x0001 PCM > AudioStreaming Interface Descriptor: > bLength 11 > bDescriptorType 36 > bDescriptorSubtype 2 (FORMAT_TYPE) > bFormatType 1 (FORMAT_TYPE_I) > bNrChannels 1 > bSubframeSize 2 > bBitResolution 16 > bSamFreqType 1 Discrete > tSamFreq[ 0] 48000 > Endpoint Descriptor: > bLength 9 > bDescriptorType 5 > bEndpointAddress 0x86 EP 6 IN > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x0064 1x 100 bytes > bInterval 4 > bRefresh 0 > bSynchAddress 0 > AudioStreaming Endpoint Descriptor: > bLength 7 > bDescriptorType 37 > bDescriptorSubtype 1 (EP_GENERAL) > bmAttributes 0x01 > Sampling Frequency > bLockDelayUnits 0 Undefined > wLockDelay 0x0000 > (snip) > > Testing patch provides consistent good sound recording quality and volume range. > > (snip) > [ 95.473820][ 1] [ T73] usb 5-2.2: new high-speed USB device number 7 using xhci_hcd > [ 95.773974][ 1] [ T73] usb 5-2.2: New USB device found, idVendor=1bcf, idProduct=2281, bcdDevice= 0.05 > [ 95.783445][ 1] [ T73] usb 5-2.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 95.791872][ 1] [ T73] usb 5-2.2: Product: HD webcam > [ 95.797001][ 1] [ T73] usb 5-2.2: Manufacturer: Sunplus IT Co > [ 95.802996][ 1] [ T73] usb 5-2.2: SerialNumber: 20200513 > [ 96.110630][ 3] [ T3680] usbcore: registered new interface driver snd-usb-audio > [ 96.114329][ 7] [ T3677] usb 5-2.2: Found UVC 1.00 device HD webcam (1bcf:2281) > [ 96.167555][ 7] [ T3677] usbcore: registered new interface driver uvcvideo > > Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> The patch can't be applied cleanly to the latest tree, as the table has been sorted. Could you rebase and resubmit? Also a nitpicking: > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -2179,6 +2179,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { > QUIRK_FLAG_FIXED_RATE), > DEVICE_FLG(0x1bcf, 0x2283, /* NexiGo N930AF FHD Webcam */ > QUIRK_FLAG_GET_SAMPLE_RATE), > + DEVICE_FLG(0x1bcf, 0x2281, /* FHD Webcam */ > + QUIRK_FLAG_GET_SAMPLE_RATE), Please try to align the space. thanks, Takashi

1 0

[PATCH] ASoC: Intel: Fix RT5650 SSP lookup
by Pierre-Louis Bossart 16 Jul '24

16 Jul '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Commit 8efcd4864652 ("ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization") migrated the pin assignment in the context struct up to soc-acpi-intel-ssp-common.c. This uses a lookup table to see if a device has a amp/codec before assigning the pin. The issue here arises when combination parts that serve both (with 2 ports) are used. sysfs: cannot create duplicate filename '/devices/pci0000:00/0000:00:1f.3/adl_rt5682_def/SSP0-Codec' CPU: 1 PID: 2079 Comm: udevd Tainted: G U 6.6.36-03391-g744739e00023 #1 3be1a2880a0970f65545a957db7d08ef4b3e2c0d Hardware name: Google Anraggar/Anraggar, BIOS Google_Anraggar.15217.552.0 05/07/2024 Call Trace: <TASK> dump_stack_lvl+0x69/0xa0 sysfs_warn_dup+0x5b/0x70 sysfs_create_dir_ns+0xb0/0x100 kobject_add_internal+0x133/0x3c0 kobject_add+0x66/0xb0 ? device_add+0x65/0x780 device_add+0x164/0x780 snd_soc_add_pcm_runtimes+0x2fa/0x800 snd_soc_bind_card+0x35e/0xc20 devm_snd_soc_register_card+0x48/0x90 platform_probe+0x7b/0xb0 really_probe+0xf7/0x2a0 ... kobject: kobject_add_internal failed for SSP0-Codec with -EEXIST, don't try to register things with the same name in the same directory. The issue is that the ALC5650 was only defined in the codec table and not the amp table which left the pin unassigned but the dai link was still created by the machine driver. Also patch the suffix filename code for the topology to prevent double suffix names as a result of this change. Fixes: 8efcd4864652 ("ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization") Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- .../intel/common/soc-acpi-intel-ssp-common.c | 9 ++++ sound/soc/sof/intel/hda.c | 49 ++++++++++--------- 2 files changed, 34 insertions(+), 24 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 75d0b931d895..de7a3f7f47f1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -64,6 +64,15 @@ static const struct codec_map amps[] = { CODEC_MAP_ENTRY("RT1015P", "rt1015", RT1015P_ACPI_HID, CODEC_RT1015P), CODEC_MAP_ENTRY("RT1019P", "rt1019", RT1019P_ACPI_HID, CODEC_RT1019P), CODEC_MAP_ENTRY("RT1308", "rt1308", RT1308_ACPI_HID, CODEC_RT1308), + + /* + * Monolithic components + * + * Only put components that can serve as both the amp and the codec below this line. + * This will ensure that if the part is used just as a codec and there is an amp as well + * then the amp will be selected properly. + */ + CODEC_MAP_ENTRY("RT5650", "rt5650", RT5650_ACPI_HID, CODEC_RT5650), }; enum snd_soc_acpi_intel_codec diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index daf364f773dd..5a40b8fbbbd3 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1307,9 +1307,10 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) const struct sof_dev_desc *desc = sof_pdata->desc; struct hdac_bus *bus = sof_to_bus(sdev); struct snd_soc_acpi_mach *mach = NULL; - enum snd_soc_acpi_intel_codec codec_type; + enum snd_soc_acpi_intel_codec codec_type, amp_type; const char *tplg_filename; const char *tplg_suffix; + bool amp_name_valid; /* Try I2S or DMIC if it is supported */ if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC))) @@ -1413,30 +1414,30 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } } - codec_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev); - - if (tplg_fixup && - mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME && - codec_type != CODEC_NONE) { - tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(codec_type); - if (!tplg_suffix) { - dev_err(sdev->dev, "no tplg suffix found, amp %d\n", - codec_type); - return NULL; - } - - tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, - "%s-%s", - sof_pdata->tplg_filename, - tplg_suffix); - if (!tplg_filename) - return NULL; - - sof_pdata->tplg_filename = tplg_filename; - add_extension = true; - } - + amp_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev); codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev); + amp_name_valid = amp_type != CODEC_NONE && amp_type != codec_type; + + if (tplg_fixup && amp_name_valid && + mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME) { + tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(amp_type); + if (!tplg_suffix) { + dev_err(sdev->dev, "no tplg suffix found, amp %d\n", + amp_type); + return NULL; + } + + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s-%s", + sof_pdata->tplg_filename, + tplg_suffix); + if (!tplg_filename) + return NULL; + + sof_pdata->tplg_filename = tplg_filename; + add_extension = true; + } + if (tplg_fixup && mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME && -- 2.43.0

2 1

[PATCH] ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices
by Pierre-Louis Bossart 16 Jul '24

16 Jul '24

Multiple users report a regression bisected to commit d5263dbbd8af ("ASoC: SOF: Intel: don't ignore IOC interrupts for non-audio transfers"). The firmware version is the likely suspect, as these users relied on SOF 2.0 while Intel only tested with the 2.2 release. Rather than completely disable the wait_for_completion(), which can help us gather timing information on the different stages of the boot process, the simplest course of action is to just disable it for older IPC versions which are no longer under active development. Closes: https://github.com/thesofproject/linux/issues/5072 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218961 Fixes: d5263dbbd8af ("ASoC: SOF: Intel: don't ignore IOC interrupts for non-audio transfers") Tested-by: Mike Krinkin <krinkin.m.u(a)gmail.com> Tested-by: Todd Brandt <todd.e.brandt(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/hda-loader.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index b8b914eaf7e0..75f6240cf3e1 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -310,15 +310,19 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream return ret; } - /* Wait for completion of transfer */ - time_left = wait_for_completion_timeout(&hda_stream->ioc, - msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS)); + if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { + /* Wait for completion of transfer */ + time_left = wait_for_completion_timeout(&hda_stream->ioc, + msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS)); - if (!time_left) { - dev_err(sdev->dev, "Code loader DMA did not complete\n"); - return -ETIMEDOUT; + if (!time_left) { + dev_err(sdev->dev, "Code loader DMA did not complete\n"); + return -ETIMEDOUT; + } + dev_dbg(sdev->dev, "Code loader DMA done\n"); } - dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n"); + + dev_dbg(sdev->dev, "waiting for FW_ENTERED status\n"); status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->rom_status_reg, reg, -- 2.43.0

2 1

[PATCH v1 1/2] ASoc: tas2781: Add TAS2563 into the Header
by Shenghao Ding 15 Jul '24

15 Jul '24

Add TAS2563 into the Header in case of misunderstanding. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header - Add channel no into the log for error debug --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } -- 2.34.1

2 3

[GIT PULL] ASoC updates for v6.11
by Mark Brown 15 Jul '24

15 Jul '24

The following changes since commit 680e126ec0400f6daecf0510c5bb97a55779ff03: firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files (2024-07-08 15:55:11 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.11 for you to fetch changes up to c51cba4755609ad97ba97713210c16f043c73224: Fix the unbalanced pm_runtime_enable in wcd937x-sdw (2024-07-12 17:38:18 +0100) ---------------------------------------------------------------- ASoC: Updates for for v6.11 There are a lot of changes in here, though the big bulk of things is cleanups and simplifications of various kinds which are internally rather than externally visible. A good chunk of those are DT schema conversions, but there's also a lot of changes in the code. Highlights: - Syncing of features between simple-audio-card and the two audio-graph cards so there is no reason to stick with an older driver. - Support for specifying the order of operations for components within cards to allow quirking for unusual systems. - New support for Asahi Kasei AK4619, Cirrus Logic CS530x, Everest Semiconductors ES8311, NXP i.MX95 and LPC32xx, Qualcomm LPASS v2.5 and WCD937x, Realtek RT1318 and RT1320 and Texas Instruments PCM5242. ---------------------------------------------------------------- Abdulrasaq Lawani (1): dt-bindings: sound: Convert max98088 to dtschema Aleksandr Mishin (2): ASoC: qcom: Adjust issues in case of DT error in asoc_qcom_lpass_cpu_platform_probe() ASoC: amd: Adjust error handling in case of absent codec device Amadeusz Sławiński (13): ASoC: topology: Simplify code ASoC: topology: Do not do unnecessary dobj management ASoC: topology: Properly initialize soc_enum values ASoC: topology: Cleanup soc_tplg_dapm_widget_dbytes_create ASoC: topology: Cleanup soc_tplg_dapm_widget_dmixer_create ASoC: topology: Cleanup soc_tplg_dapm_widget_denum_create ASoC: topology: Correctly set shift_r in soc_tplg_denum_create() ASoC: topology: Align dynamic object initialization for controls ASoC: topology: Rename function creating widget kcontrol ASoC: topology: Reshuffle function placement ASoC: topology: Unify code for creating standalone and widget bytes control ASoC: topology: Unify code for creating standalone and widget mixer control ASoC: topology: Unify code for creating standalone and widget enum control Andrejs Cainikovs (3): ASoC: nau8822: move nau8822_set_dai_sysclk() ASoC: nau8822: set NAU8822_REFIMP_80K only once ASoC: nau8822: add MCLK support Andy Shevchenko (6): ASoC: codecs: Remove unused of_gpio.h ASoC: fsl: Remove unused of_gpio.h ASoC: rockchip: Remove unused of_gpio.h ASoC: codecs: Replace of_gpio.h by proper one ASoC: generic: Replace of_gpio.h by proper one ASoC: samsung: Replace of_gpio.h by proper one Animesh Agarwal (11): ASoC: dt-bindings: ak4554: Convert to dtschema ASoC: dt-bindings: linux,spdif: Convert spdif-reciever.txt to dtschema ASoC: dt-bindings: wlf,wm8782: Convert to dtschema ASoC: dt-bindings: wlf,wm8804: Convert to dtschema ASoC: dt-bindings: realtek,rt5631: Convert to dtschema ASoC: dt-bindings: realtek,rt5514: Convert to dtschema ASoC: dt-bindings: realtek,rt5659: Convert to dtschema ASoC: dt-bindings: realtek,rt5677: Convert to dtschema ASoC: dt-bindings: realtek,rt5645: Convert to dtschema ASoC: dt-bindings: cirrus,cs4270: Convert to dtschema ASoC: dt-bindings: cirrus,cs42xx8: Convert to dtschema Artur Weber (6): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate Bard Liao (8): ASoC: Intel: sof_sdw_cs42l42: use dai parameter ASoC: Intel: sof_sdw_rt711: use dai parameter ASoC: Intel: sof_sdw_rt5682: use dai parameter ASoC: Intel: sof_sdw_rt700: use dai parameter ASoC: Intel: sof_sdw_rt_dmic: use from dai parameter ASoC: Intel: sof_sdw_rt_sdca_jack_common: use dai parameter ASoC: Intel: sof_sdw: remove get_codec_dai_by_name ASoC: Intel: sof_sdw: select PINCTRL_CS42L43 and SPI_CS42L43 Brent Lu (8): ASoC: SOF: sof-audio: rename dai clock setting query function ASoC: SOF: sof-audio: add sof_dai_get_tdm_slots function ASoC: SOF: ipc3-topology: support tdm slot number query ASoC: SOF: ipc4-topology: support tdm slot number query ASoC: Intel: maxim-common: rewrite max_98373_hw_params function ASoC: Intel: sof_da7219: remove local max98373 ops ASoC: Intel: sof_da7219: disable max98373 speaker pins in late_probe ASoC: Intel: maxim-common: add max_98373_get_tx_mask function Chancel Liu (3): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX95 ASoC: fsl_rpmsg: Add support for i.MX95 platform ASoC: fsl_xcvr: Improve suspend/resume flow in fsl_xcvr_trigger() Charles Keepax (3): ASoC: Intel: sof_sdw: Add missing controls for cs42l43/cs35l56 ASoC: Intel: soc-acpi: Add match entries for some cs42l43 laptops ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops Chen Ni (2): ASoC: max98088: Check for clk_prepare_enable() error ASoC: Intel: sof_sdw: Convert comma to semicolon Christian Hewitt (2): ASoC: Add support for ti,pcm5242 to the pcm512x driver ASoC: dt-bindings: add ti,pcm5242 to pcm512x Christophe JAILLET (8): ASoC: topology: Constify an argument of snd_soc_tplg_component_load() ASoC: Intel: avs: Constify struct snd_soc_tplg_ops ASoC: qdsp6: audioreach: Constify struct snd_soc_tplg_ops ASoC: Intel: Skylake: Constify struct snd_soc_tplg_ops ASoC: SOF: topology: Constify struct snd_soc_tplg_ops const_structs.checkpatch: add snd_soc_tplg_ops ASoC: intel: Constify struct snd_soc_ops ASoC: SOF: mediatek: Constify struct mtk_adsp_ipc_ops Dan Carpenter (1): ASoC: samsung: midas_wm1811: Fix error code in probe() Dr. David Alan Gilbert (3): ASoC: codecs: lpass-rx-macro: remove unused struct 'rx_macro_reg_mask_val' ASoC: codecs: wm0010: remove unused struct 'wm0010_spi_msg' ASoC: codecs: cx2072x: remove unused struct 'cx2072x_eq_ctrl' Elinor Montmasson (5): ASoC: fsl-asoc-card: add support for dai links with multiple codecs ASoC: fsl-asoc-card: add second dai link component for codecs ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge Fabio Estevam (3): ASoC: fsl: fsl_aud2htx: Switch to RUNTIME_PM_OPS() ASoC: fsl: fsl_easrc: Switch to RUNTIME_PM_OPS() ASoC: fsl: fsl_xcvr: Switch to RUNTIME_PM_OPS() Hao Ge (1): ASoc: PCM6240: Return directly after a failed devm_kzalloc() in pcmdevice_i2c_probe() Herve Codina (10): ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned value ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handling ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion routines ASoC: fsl: fsl_qmc_audio: Introduce qmc_audio_pcm_{read,write}_submit() ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved() soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle list soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles() dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAI ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode. Jack Yu (1): ASoC: rt1318: Add RT1318 audio amplifier driver Javier Carrasco (9): ASoC: cs35l34: Constify struct regmap_config ASoC: cs35l35: Constify struct regmap_config ASoC: cs35l36: Constify struct regmap_config ASoC: cs53l30: Constify struct regmap_config ASoC: jz4760: Constify struct regmap_config ASoC: jz4770: Constify struct regmap_config ASoC: wsa881x: Constify struct regmap_config ASoC: wsa883x: Constify struct regmap_config ASoC: wsa884x: Constify struct regmap_config Jeff Johnson (2): ASoC: amd: add missing MODULE_DESCRIPTION() macros ASoC: fsl: imx-pcm-fiq: add missing MODULE_DESCRIPTION() macro Jerome Brunet (2): ASoC: soc-utils: allow sample rate up to 768kHz for the dummy dai ASoC: meson: tdm: add sample rate support up to 768kHz Jiaxin Yu (1): ASoC: mediatek: mt6358: Add "Dmic Mode Switch" kcontrol for switch DMIC mode. Khanh Le (1): ASoC: Add ak4619 codec support Krzysztof Kozlowski (70): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6apm-lpass-dais: Implement proper channel mapping ASoC: qcom: qdsp6: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping ASoC: codecs: wcd938x: Drop unused duplicated MIC2 bias register defines ASoC: codecs: wcd938x: Unify define used for MIC2 bias register ASoC: codecs: wcd939x: Unify define used for MIC bias registers ASoC: codecs: wcd939x: Minor white-space and define cleanup ASoC: codecs: wcd939x: Unify define used for MIC bias VOUT registers ASoC: codecs: wcd938x: Drop no-op ADC2_BCS Disable Switch ASoC: soc-dai.h: Constify DAI ops auto_selectable_formats ASoC: Constify DAI ops auto_selectable_formats ASoC: Constify of_phandle_args in snd_soc_dai_driver ASoC: Constify of_phandle_args in snd_soc_dai_link_component ASoC: Constify passed data to core function ASoC: Constify DAI passed to get_channel_map ASoC: Constify return of snd_soc_dai_get_pcm_stream() ASoC: qcom: x1e80100: Add USB DisplayPort plug support ASoC: codecs: wcd-mbhc: Constify passed MBHC reg fields ASoC: codecs: wcd9335: Drop unused state container fields ASoC: codecs: wcd9335: Constify static data ASoC: codecs: wcd9335: Handle nicer probe deferral and simplify with dev_err_probe() ASoC: codecs: wcd9335: Drop unneeded error message ASoC: codecs: wcd9335: Drop unused dmic rate handling ASoC: codecs: wcd934x: Drop unused interp path enum ASoC: codecs: wcd934x: Constify static data ASoC: codecs: wcd934x: Drop unused mic bias voltage fields ASoC: codecs: wcd934x: Handle nicer probe deferral and simplify with dev_err_probe() ASoC: codecs: wcd937x: Constify static data ASoC: codecs: wcd937x: Constify wcd937x_sdw_ch_info ASoC: codecs: wcd937x: Drop unused enums, defines and types ASoC: codecs: wcd937x: Drop unused state container fields ASoC: codecs: wcd937x: Drop unused chipid member ASoC: codecs: wcd938x: Constify static data ASoC: codecs: wcd938x: Constify wcd938x_sdw_ch_info ASoC: codecs: wcd938x: Drop unused RX/TX direction enum ASoC: codecs: wcd938x: Drop unused num_ports field ASoC: codecs: wcd939x: Constify static data ASoC: codecs: wcd939x: Constify wcd939x_sdw_ch_info ASoC: codecs: wcd939x: Drop unused RX/TX direction enum ASoC: codecs: wcd939x: Drop unused num_ports field ASoC: codecs: lpass-rx-macro: add missing handling of v2.1 codec ASoC: codecs: lpass-wsa-macro: Drop unused define ASoC: codecs: lpass-wsa-macro: Prepare to accommodate new codec versions ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version ASoC: codecs: lpass-macro: Gracefully handle unknown version ASoC: codecs: lpass-macro: Use enum for handling codec version ASoC: codecs: lpass-wsa-macro: add missing select of common code ASoC: codecs: lpass-rx-macro: Fix using NULL pointer in probe() dev_err ASoC: codecs: lpass-wsa-macro: Fix using NULL pointer in probe() dev_err ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm ASoC: codecs: lpass-rx-macro: Simplify with cleanup.h ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgets ASoC: codecs: lpass-wsa-macro: Simplify with cleanup.h ASoC: codecs: wcd939x: Fix typec mux and switch leak during device removal ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_add_dapms() with cleanup.h ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_probe() with cleanup.h ASoC: codecs: wcd9335: Simplify with cleanup.h ASoC: codecs: wcd934x: Simplify with cleanup.h ASoC: simple-card-utils: Simplify with cleanup.h ASoC: audio-graph-card: Use cleanup.h instead of devm_kfree() ASoC: audio-graph-card2: Use cleanup.h instead of devm_kfree() ASoC: simple-card: Use cleanup.h instead of devm_kfree() ASoC: ops: Simplify with cleanup.h ASoC: dapm: Simplify dapm_cnew_widget() with cleanup.h ASoC: dapm: Simplify snd_soc_dai_link_event_pre_pmu() with cleanup.h ASoC: codecs: aw88395: Simplify with cleanup.h ASoC: qcom: topology: Simplify with cleanup.h Kuninori Morimoto (22): ASoC: simple-card-utils: remove both playback/capture_only check ASoC: audio-graph-card2: add ep_to_port() / port_to_ports() ASoC: audio-graph-card2: remove ports node name check ASoC: audio-graph-card2: expand dai_link property part ASoC: audio-graph-card2: merge graph_parse_mclk_fs() into graph_link_init() ASoC: audio-graph-card: add ep_to_port() / port_to_ports() ASoC: audio-graph-card: remove ports node name check ASoC: audio-graph-card: enable playback/capture_only property ASoC: audio-graph-card: merge graph_parse_mclk_fs() into graph_link_init() ASoC: simple-audio-card: enable playback/capture_only property ASoC: simple-audio-card: merge simple_parse_mclk_fs() into simple_link_init() ASoC: audio-graph-card2: add support for aux devices ASoC: dt-bindings: audio-graph-card2: add support for aux devices ASoC: dt-bindings: ak4619: Add initial DT binding ASoC: dt-bindings: add missing vender prefix on filename ASoC: audio-graph-port: add link-trigger-order ASoC: simple-card-utils: add link-trigger-order support ASoC: simple-audio-card: add link-trigger-order support ASoC: audio-graph-card: add link-trigger-order support ASoC: audio-graph-card2: add link-trigger-order support ASoC: simple-audio-mux: enable to select MUX names ASoC: dt-bindings: simple-audio-mux: add state-labels property Linus Walleij (1): ASoC: tas5086: Convert to GPIO descriptors Lukas Bulwahn (1): MAINTAINERS: adjust file entries after adding vendor prefix in sound dtbs Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Mark Brown (42): Support Tegra I2S client format conversion ASoC: Intel: boards: updates for 6.11 ASoC: qcom: x1e80100: Correct channel mapping ASoC: fsl_mqs: Add i.MX95 platform support ASoC: codecs: add support for everest-semi es8311 ASoC: samsung: midas-audio: Add GPIO-based headset ASoC: Merge up fixes Dead structs in sound/soc/codecs ASoC: fsl_xcvr: Support i.MX95 platform ASoC: simple-card: sync support ASoC: codecs: wcd937x: add wcd937x audio codec ACPI/ALSA/soundwire: add acpi_get_local_u64_address() ASoC: Drop or replace of_gpio.h ASoC: codecs: lpass: add support for v2.5 rx macro ASoC: Merge up fixes Cirrus Logic Family of ADCs tlv320adc3xxx: Allow MICBIAS pins to be used as ASoC: dt-bindings: convert everest,es7134.txt & ASoC: codecs: wcd family: cleanups ASoC: add compatible for ti,pcm5242 ASoC: Add ak4619 codec support ASoC: Few constifications (mostly arguments) ASoC: qcom: display port changes ASoC: nau8822: add MCLK support ASoC: codecs: ES8326: Solving headphone detection and ASoC: Intel: boards: updates for 6.11 - part2 ASoC: simple-card / audio-graph: ASoC: imx-audmix: Split capture device to be a new Add audio support for LPC32XX CPUs ASoC: codecs: wsa88xx: add support for static port ASoC: cs35l56: Remove obsolete and redundant code ASoC: simple-audio-mux: add state-labels Add master clock handling for nau8824 ASoC: topology: kcontrol registration cleanup ASoC: cs35l56: Set correct upper volume limit ASoC: codecs: lpass-rx-macro: Few code cleanups Add support for non-interleaved mode in qmc_audio ASoC: Simplify code with cleanup.h ASoC: fsl-asoc-card: add S/PDIF controller support firmware: cs_dsp: Some small coding improvements ASoC: dt-bindings: convert qcom sound bindings to Fix the unbalanced pm_runtime_enable in wcd937x-sdw Matteo Martelli (2): ASoC: es8311: dt-bindings: add everest es8311 codec ASoC: codecs: es8311: add everest es8311 codec support Maxim Kochetkov (2): ASoC: dt-bindings: nau8824: Add master clock handling ASoC: codecs: nau8824: Add master clock handling Mithil Bavishi (1): ASoC: dt-bindings: omap-mcpdm: Convert to DT schema Mohammad Rafi Shaik (3): ASoC: codecs: wcd937x-sdw: Fix Unbalanced pm_runtime_enable ASoC: codecs: wcd937x: Remove the string compare in MIC BIAS widget settings ASoC: codecs: wcd937x: Remove separate handling for vdd-buck supply Mohan Kumar (2): ASoC: simple-card-utils: Split simple_fixup_sample_fmt func ASoC: tegra: I2S client convert formats handling Nathan Chancellor (2): ASoC: fsl: lpc3xxx-i2s: Avoid using ret uninitialized in lpc32xx_i2s_probe() ASoC: fsl: lpc3xxx-i2s: Include bitfield.h for FIELD_PREP Neil Armstrong (5): ASoC: dt-bindings: convert amlogic,g12a-tohdmitx to dt-schema ASoC: dt-bindings: convert tas571x.txt to dt-schema ASoC: dt-bindings: convert everest,es7241.txt to dt-schema ASoC: dt-bindings: convert everest,es7134.txt to dt-schema ASoC: dt-bindings: amlogic,gx-sound-card: drop minItems for audio-widgets Paul Handrigan (4): ASoC: dt-bindings: cirrus,cs530x: Add initial DT binding ASoC: cs530x: Support for cs530x ADCs ASoC: cs530x: Calculate proper bclk rate using TDM ASoC: cs530x: Remove bclk from private structure Peter Ujfalusi (4): ASoC: SOF: ipc4-topology: Add support for NHLT with 16-bit only DMIC blob ASoC: SOF: Intel: pci-tgl: Align ADL-N sof_dev_desc struct name to convention ASoC: SOF: ipc4-topology: Use correct queue_id for requesting input pin format ASoC: SOF: ipc4-topology: Use single token list for the copiers Pierre-Louis Bossart (6): ASoC: SOF: Intel: hda: print PCI class info only once ACPI: utils: introduce acpi_get_local_u64_address() soundwire: slave: simplify code with acpi_get_local_u64_address() ALSA: hda: intel-sdw-acpi: use acpi_get_local_u64_address() ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C Piotr Wojtaszczyk (2): ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig Rayyan Ansari (2): ASoC: dt-bindings: qcom,msm8916-wcd-digital-codec: convert to dtschema ASoC: dt-bindings: qcom,apq8096-sndcard: use dtschema Ricard Wanderlof (3): ASoC: dt-bindings: tlv320adc3xxx: Fix incorrect GPIO description ASoC: dt-bindings: tlv320adc3xxx: Add MICBIAS-as-GPO properties ASoC: tlv320adc3xxx: Add support for using MICBIAS pins as GPO Richard Fitzgerald (10): ASoC: cs35l56: Revert support for dual-ownership of ASP registers ASoC: cs35l56: Remove support for A1 silicon ASoC: cs35l56: Remove redundant clearing of clk_stop_mode1 firmware: cs_dsp: Don't allow writes to read-only controls ASoC: cs35l56: Use header defines for Speaker Volume control definition ASoC: cs35l56: Limit Speaker Volume to +12dB maximum firmware: cs_dsp: Don't allocate temporary buffer for info text firmware: cs_dsp: Make wmfw and bin filename arguments const char * firmware: cs_dsp: Clarify wmfw format version log message firmware: cs_dsp: Rename fw_ver to wmfw_ver Rob Herring (Arm) (2): ASoC: PCM6240: Use of_property_read_reg() ASoC: tas2781: Use of_property_read_reg() Shenghao Ding (8): ASoc: tas2781: Enable RCA-based playback without DSP firmware download ASoc: tas2781: Add name_prefix as the prefix name of firmwares and kcontrol to support corresponding TAS2563/TAS2781s ASoc: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 ASoc: pcm6240: Remove unnecessary name-prefix for all the controls ASoc: TAS2781: replace beXX_to_cpup with get_unaligned_beXX for potentially broken alignment ASoc: TAS2781: rename the tas2781_reset as tasdevice_reset ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume Shengjiu Wang (9): ASoC: dt-bindings: fsl,mqs: Add i.MX95 platform support ASoC: fsl_mqs: Add i.MX95 platform support ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX95 ASoC: fsl_xcvr: Add support for i.MX95 platform ASoC: fsl_sai: Add separate DAI for transmitter and receiver ASoC: fsl_audmix: Split playback and capture stream to different DAI ASoC: imx-audmix: Split capture device for audmix ASoC: dt-bindings: fsl,xcvr: Adjust the number of interrupts ASoC: ak4458: remove "reset-gpios" property handler Shuming Fan (4): ASoC: rt1320: Add RT1320 SDCA vendor-specific driver ASoC: rt711-sdca: add GE selected mode control ASoC: rt712-sdca: change the definition name of SDCA channel number ASoC: rt712-sdca: add the function for version B Simon Trimmer (2): ASoC: cs35l56: Attempt to read from cirrus,speaker-id device property first ASoC: cs35l56: Accept values greater than 0 as IRQ numbers Srinivas Kandagatla (9): ASoC: codecs: lpass-macro: add helpers to get codec version ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions ASoC: codec: lpass-rx-macro: add support for 2.5 codec version ASoC: qcom: q6dsp: parse Display port tokens ASoC: qcom: common: add Display port Jack function ASoC: qcom: sc8280xp: add Display port Jack ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information Tony Luck (1): ASoC: Intel: avs: es8336: Switch to new Intel CPU model defines Uwe Kleine-König (1): ASoC: codecs: Drop explicit initialization of struct i2c_device_id::driver_data to 0 Vijendar Mukunda (5): ASoC: amd: acp: remove acp_i2s_probe function ASoC: amd: acp: remove unused variables from acp_resource structure ASoC: amd: acp: modify conditional check for programming i2s mclk ASoC: amd: acp: move i2s clock generation sequence ASoC: amd: acp: add pcm constraints for buffer size and period size Xiaxi Shen (1): ASoC: dt-bindings: ak4104: convert to dt schema Yang Li (1): ASoC: Remove unneeded semicolon Zhang Yi (3): ASoC: codecs: ES8326: Slove headphone detection issue ASoC: codecs: ES8326: Minimize the pop noise ASoC: codecs: ES8326: regcache_sync error issue Documentation/admin-guide/LSM/tomoyo.rst | 35 +- Documentation/admin-guide/mm/transhuge.rst | 4 +- Documentation/cdrom/cdrom-standard.rst | 4 +- .../devicetree/bindings/arm/stm32/st,mlahb.yaml | 3 +- Documentation/devicetree/bindings/arm/sunxi.yaml | 6 +- .../devicetree/bindings/input/elan,ekth6915.yaml | 19 +- .../devicetree/bindings/input/ilitek,ili2901.yaml | 66 + Documentation/devicetree/bindings/sound/ak4104.txt | 25 - Documentation/devicetree/bindings/sound/ak4554.txt | 11 - .../bindings/sound/amlogic,g12a-tohdmitx.txt | 58 - .../bindings/sound/amlogic,g12a-tohdmitx.yaml | 54 + .../bindings/sound/amlogic,gx-sound-card.yaml | 1 - .../bindings/sound/asahi-kasei,ak4104.yaml | 49 + .../sound/{ak4375.yaml => asahi-kasei,ak4375.yaml} | 2 +- .../bindings/sound/asahi-kasei,ak4554.yaml | 27 + .../sound/{ak4613.yaml => asahi-kasei,ak4613.yaml} | 2 +- .../bindings/sound/asahi-kasei,ak4619.yaml | 62 + .../sound/{ak4642.yaml => asahi-kasei,ak4642.yaml} | 2 +- .../bindings/sound/audio-graph-card2.yaml | 5 + .../bindings/sound/audio-graph-port.yaml | 9 + .../devicetree/bindings/sound/cirrus,cs4270.yaml | 59 + .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 81 + .../devicetree/bindings/sound/cirrus,cs530x.yaml | 85 + Documentation/devicetree/bindings/sound/cs4270.txt | 21 - .../devicetree/bindings/sound/cs42xx8.txt | 34 - .../devicetree/bindings/sound/everest,es7134.txt | 15 - .../devicetree/bindings/sound/everest,es71x4.yaml | 62 + .../devicetree/bindings/sound/everest,es7241.txt | 28 - .../devicetree/bindings/sound/everest,es7241.yaml | 67 + .../devicetree/bindings/sound/everest,es8316.yaml | 7 +- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 - .../devicetree/bindings/sound/fsl,mqs.yaml | 2 + .../devicetree/bindings/sound/fsl,qmc-audio.yaml | 41 +- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + .../sound/{sgtl5000.yaml => fsl,sgtl5000.yaml} | 2 +- .../devicetree/bindings/sound/fsl,xcvr.yaml | 43 +- .../devicetree/bindings/sound/fsl-asoc-card.yaml | 53 +- .../{linux,spdif-dit.yaml => linux,spdif.yaml} | 8 +- .../devicetree/bindings/sound/maxim,max98088.txt | 23 - .../devicetree/bindings/sound/maxim,max98088.yaml | 47 + .../sound/{zl38060.yaml => mscc,zl38060.yaml} | 2 +- .../devicetree/bindings/sound/nuvoton,nau8824.yaml | 8 + .../devicetree/bindings/sound/nxp,lpc3220-i2s.yaml | 73 + .../devicetree/bindings/sound/omap-mcpdm.txt | 30 - .../devicetree/bindings/sound/pcm512x.txt | 2 +- .../devicetree/bindings/sound/qcom,apq8096.txt | 128 - .../sound/qcom,msm8916-wcd-digital-codec.yaml | 55 + .../bindings/sound/qcom,msm8916-wcd-digital.txt | 20 - .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + .../bindings/sound/qcom,wcd937x-sdw.yaml | 91 + .../devicetree/bindings/sound/qcom,wcd937x.yaml | 82 + .../devicetree/bindings/sound/qcom,wsa883x.yaml | 8 + .../devicetree/bindings/sound/qcom,wsa8840.yaml | 8 + .../sound/{rt1019.yaml => realtek,rt1019.yaml} | 2 +- .../devicetree/bindings/sound/realtek,rt5514.yaml | 70 + .../devicetree/bindings/sound/realtek,rt5631.yaml | 67 + .../devicetree/bindings/sound/realtek,rt5645.yaml | 131 + .../devicetree/bindings/sound/realtek,rt5659.yaml | 129 + .../devicetree/bindings/sound/realtek,rt5677.yaml | 135 + Documentation/devicetree/bindings/sound/rt5514.txt | 37 - Documentation/devicetree/bindings/sound/rt5631.txt | 48 - Documentation/devicetree/bindings/sound/rt5645.txt | 82 - Documentation/devicetree/bindings/sound/rt5659.txt | 89 - Documentation/devicetree/bindings/sound/rt5677.txt | 78 - .../bindings/sound/samsung,midas-audio.yaml | 33 + .../bindings/sound/simple-audio-mux.yaml | 6 + .../devicetree/bindings/sound/spdif-receiver.txt | 10 - .../devicetree/bindings/sound/tas571x.txt | 49 - .../devicetree/bindings/sound/ti,omap4-mcpdm.yaml | 73 + .../sound/{tas2562.yaml => ti,tas2562.yaml} | 2 +- .../sound/{tas2770.yaml => ti,tas2770.yaml} | 2 +- .../sound/{tas27xx.yaml => ti,tas27xx.yaml} | 2 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 133 + .../sound/{tas5805m.yaml => ti,tas5805m.yaml} | 2 +- .../bindings/sound/ti,tlv320adc3xxx.yaml | 32 +- .../{tlv320adcx140.yaml => ti,tlv320adcx140.yaml} | 2 +- .../sound/{wm8750.yaml => wlf,wm8750.yaml} | 2 +- .../devicetree/bindings/sound/wlf,wm8782.yaml | 47 + .../devicetree/bindings/sound/wlf,wm8804.yaml | 58 + Documentation/devicetree/bindings/sound/wm8782.txt | 24 - Documentation/devicetree/bindings/sound/wm8804.txt | 25 - Documentation/kbuild/kconfig-language.rst | 12 +- Documentation/networking/af_xdp.rst | 33 +- .../userspace-api/media/v4l/dev-subdev.rst | 2 +- MAINTAINERS | 24 +- Makefile | 2 +- arch/arm/configs/imx_v6_v7_defconfig | 1 - arch/arm64/configs/defconfig | 1 - arch/arm64/include/asm/el2_setup.h | 6 +- arch/arm64/include/asm/io.h | 36 +- arch/arm64/include/asm/kvm_arm.h | 6 + arch/arm64/include/asm/kvm_emulate.h | 71 +- arch/arm64/include/asm/kvm_host.h | 25 +- arch/arm64/include/asm/kvm_hyp.h | 4 +- arch/arm64/include/asm/kvm_pkvm.h | 9 + arch/arm64/kernel/armv8_deprecated.c | 3 + arch/arm64/kvm/arm.c | 76 + arch/arm64/kvm/emulate-nested.c | 21 +- arch/arm64/kvm/fpsimd.c | 11 +- arch/arm64/kvm/guest.c | 3 +- arch/arm64/kvm/hyp/aarch32.c | 18 +- arch/arm64/kvm/hyp/fpsimd.S | 6 + arch/arm64/kvm/hyp/include/hyp/switch.h | 36 +- arch/arm64/kvm/hyp/include/nvhe/pkvm.h | 1 - arch/arm64/kvm/hyp/nvhe/hyp-main.c | 84 +- arch/arm64/kvm/hyp/nvhe/pkvm.c | 17 +- arch/arm64/kvm/hyp/nvhe/setup.c | 25 +- arch/arm64/kvm/hyp/nvhe/switch.c | 24 +- arch/arm64/kvm/hyp/vhe/switch.c | 12 +- arch/arm64/kvm/nested.c | 6 +- arch/arm64/kvm/reset.c | 3 + arch/arm64/mm/contpte.c | 4 +- arch/loongarch/boot/dts/loongson-2k0500-ref.dts | 4 +- arch/loongarch/boot/dts/loongson-2k1000-ref.dts | 4 +- arch/loongarch/boot/dts/loongson-2k2000-ref.dts | 2 +- arch/loongarch/include/asm/numa.h | 1 + arch/loongarch/include/asm/stackframe.h | 2 +- arch/loongarch/kernel/head.S | 2 +- arch/loongarch/kernel/setup.c | 6 +- arch/loongarch/kernel/smp.c | 5 +- arch/loongarch/kernel/vmlinux.lds.S | 10 +- arch/riscv/kvm/aia_device.c | 7 +- arch/riscv/kvm/vcpu_onereg.c | 4 +- arch/riscv/mm/fault.c | 4 +- arch/riscv/mm/init.c | 21 +- arch/s390/kernel/crash_dump.c | 54 +- arch/x86/include/asm/kvm_host.h | 1 + arch/x86/include/asm/vmxfeatures.h | 2 +- arch/x86/kernel/amd_nb.c | 9 +- arch/x86/kernel/machine_kexec_64.c | 11 +- arch/x86/kvm/Kconfig | 11 +- arch/x86/kvm/lapic.c | 39 +- arch/x86/kvm/lapic.h | 2 +- arch/x86/kvm/mmu/mmu.c | 46 +- arch/x86/kvm/mmu/spte.h | 9 + arch/x86/kvm/mmu/tdp_iter.h | 2 + arch/x86/kvm/mmu/tdp_mmu.c | 2 +- arch/x86/kvm/svm/sev.c | 19 +- arch/x86/kvm/svm/svm.c | 69 +- arch/x86/kvm/svm/svm.h | 4 +- arch/x86/kvm/vmx/nested.c | 5 + arch/x86/kvm/vmx/vmx.c | 11 +- arch/x86/kvm/x86.c | 11 +- drivers/acpi/ac.c | 4 +- drivers/acpi/apei/einj-core.c | 2 +- drivers/acpi/ec.c | 9 +- drivers/acpi/sbs.c | 4 +- drivers/acpi/utils.c | 16 +- drivers/ata/pata_macio.c | 9 +- drivers/block/null_blk/main.c | 4 +- drivers/char/tpm/tpm.h | 2 +- drivers/char/tpm/tpm_tis_core.c | 3 +- drivers/char/tpm/tpm_tis_core.h | 2 +- drivers/clk/sifive/sifive-prci.c | 8 - drivers/cpufreq/amd-pstate-ut.c | 3 +- drivers/cpufreq/amd-pstate.c | 36 +- {include/linux => drivers/cpufreq}/amd-pstate.h | 33 - drivers/cpufreq/intel_pstate.c | 3 +- drivers/cxl/core/region.c | 18 +- drivers/edac/amd64_edac.c | 8 +- drivers/edac/igen6_edac.c | 4 +- drivers/firmware/cirrus/cs_dsp.c | 71 +- drivers/firmware/efi/efi-pstore.c | 8 +- drivers/firmware/efi/libstub/loongarch.c | 2 +- drivers/firmware/efi/libstub/zboot.lds | 1 + drivers/firmware/efi/runtime-wrappers.c | 13 +- drivers/gpio/Kconfig | 2 +- drivers/gpio/gpio-gw-pld.c | 1 + drivers/gpio/gpio-mc33880.c | 1 + drivers/gpio/gpio-pcf857x.c | 1 + drivers/gpio/gpio-pl061.c | 1 + drivers/gpio/gpio-tqmx86.c | 110 +- drivers/gpu/drm/amd/include/pptable.h | 91 +- .../gpu/drm/amd/pm/swsmu/smu13/smu_v13_0_4_ppt.c | 20 +- .../gpu/drm/arm/display/komeda/komeda_color_mgmt.c | 5 - drivers/gpu/drm/panel/panel-sitronix-st7789v.c | 4 +- drivers/gpu/drm/vmwgfx/vmwgfx_drv.c | 19 +- drivers/gpu/drm/vmwgfx/vmwgfx_drv.h | 3 - drivers/gpu/drm/vmwgfx/vmwgfx_gmrid_manager.c | 4 +- drivers/gpu/drm/vmwgfx/vmwgfx_kms.c | 28 +- drivers/gpu/drm/vmwgfx/vmwgfx_stdu.c | 60 +- drivers/gpu/drm/xe/xe_gt_sriov_pf_config.c | 1 + drivers/hid/hid-asus.c | 4 +- drivers/hid/hid-core.c | 1 - drivers/hid/hid-debug.c | 2 + drivers/hid/hid-ids.h | 2 + drivers/hid/hid-input.c | 13 + drivers/hid/hid-logitech-dj.c | 4 +- drivers/hid/hid-logitech-hidpp.c | 1 + drivers/hid/hid-nintendo.c | 6 +- drivers/hid/hid-nvidia-shield.c | 4 +- drivers/hid/i2c-hid/i2c-hid-of-elan.c | 59 +- drivers/hid/intel-ish-hid/ishtp/loader.c | 79 +- drivers/hid/intel-ish-hid/ishtp/loader.h | 33 +- drivers/i2c/busses/i2c-synquacer.c | 11 +- drivers/input/touchscreen/silead.c | 19 +- drivers/iommu/amd/amd_iommu.h | 3 +- drivers/iommu/amd/init.c | 9 + drivers/iommu/amd/iommu.c | 48 +- drivers/iommu/amd/ppr.c | 25 +- drivers/iommu/dma-iommu.c | 8 +- drivers/irqchip/irq-gic-v3-its.c | 44 +- drivers/irqchip/irq-riscv-intc.c | 9 +- drivers/irqchip/irq-sifive-plic.c | 34 +- drivers/media/pci/intel/ipu6/ipu6-isys-queue.c | 6 +- drivers/media/pci/intel/ipu6/ipu6-isys.c | 71 +- drivers/media/pci/intel/ipu6/ipu6.c | 5 +- drivers/media/pci/intel/ivsc/mei_csi.c | 5 +- drivers/media/pci/mgb4/mgb4_core.c | 7 +- drivers/net/ethernet/intel/ice/ice.h | 44 +- drivers/net/ethernet/intel/ice/ice_base.c | 3 + drivers/net/ethernet/intel/ice/ice_lib.c | 27 +- drivers/net/ethernet/intel/ice/ice_main.c | 118 +- drivers/net/ethernet/intel/ice/ice_nvm.c | 116 +- drivers/net/ethernet/intel/ice/ice_type.h | 14 +- drivers/net/ethernet/intel/ice/ice_xsk.c | 13 +- drivers/net/ethernet/intel/igc/igc_ethtool.c | 9 +- drivers/net/ethernet/intel/igc/igc_main.c | 4 + .../net/ethernet/marvell/octeontx2/af/rvu_npc.c | 33 +- drivers/net/ethernet/mediatek/mtk_eth_soc.c | 104 +- drivers/net/ethernet/mediatek/mtk_eth_soc.h | 9 +- drivers/net/ethernet/mellanox/mlx5/core/fw.c | 4 + drivers/net/ethernet/mellanox/mlx5/core/health.c | 8 + .../net/ethernet/mellanox/mlx5/core/lag/port_sel.c | 8 +- .../net/ethernet/mellanox/mlx5/core/lib/pci_vsc.c | 4 + drivers/net/ethernet/mellanox/mlx5/core/main.c | 3 + drivers/net/ethernet/pensando/ionic/ionic_txrx.c | 1 + drivers/net/phy/micrel.c | 104 +- drivers/net/virtio_net.c | 42 +- drivers/net/vmxnet3/vmxnet3_drv.c | 2 +- drivers/net/vxlan/vxlan_core.c | 8 +- drivers/net/wireless/ath/ath10k/Kconfig | 1 + drivers/net/wireless/ath/ath11k/core.c | 2 +- drivers/net/wireless/ath/ath11k/mac.c | 38 +- drivers/net/wireless/ath/ath11k/pcic.c | 25 +- drivers/net/wireless/intel/iwlwifi/iwl-drv.c | 2 +- drivers/net/wireless/intel/iwlwifi/mvm/d3.c | 16 +- drivers/net/wireless/intel/iwlwifi/mvm/debugfs.c | 9 + drivers/net/wireless/intel/iwlwifi/mvm/fw.c | 14 +- drivers/net/wireless/intel/iwlwifi/mvm/mac-ctxt.c | 2 +- drivers/net/wireless/intel/iwlwifi/mvm/mac80211.c | 39 +- .../net/wireless/intel/iwlwifi/mvm/mld-mac80211.c | 2 - drivers/net/wireless/intel/iwlwifi/mvm/mld-sta.c | 13 +- drivers/net/wireless/intel/iwlwifi/mvm/mvm.h | 1 + drivers/net/wireless/intel/iwlwifi/mvm/rs.h | 9 +- drivers/net/wireless/intel/iwlwifi/mvm/rxmq.c | 5 +- drivers/net/wireless/intel/iwlwifi/mvm/scan.c | 12 +- drivers/net/wireless/intel/iwlwifi/mvm/sta.c | 12 +- drivers/net/wireless/intel/iwlwifi/mvm/sta.h | 5 + drivers/net/wireless/mediatek/mt76/mt7615/main.c | 4 + drivers/net/wireless/microchip/wilc1000/cfg80211.c | 41 +- drivers/net/wireless/microchip/wilc1000/hif.c | 17 +- drivers/net/wireless/microchip/wilc1000/netdev.c | 43 +- drivers/net/wireless/microchip/wilc1000/netdev.h | 12 +- drivers/net/wireless/microchip/wilc1000/wlan.c | 5 +- drivers/net/wireless/realtek/rtlwifi/core.c | 15 - drivers/net/wwan/iosm/iosm_ipc_devlink.c | 2 +- drivers/nvme/host/fabrics.c | 6 +- drivers/nvme/host/pr.c | 2 +- drivers/of/irq.c | 125 +- drivers/of/of_private.h | 3 + drivers/of/of_test.c | 1 + drivers/of/property.c | 30 +- drivers/pci/access.c | 4 - drivers/pci/pci.c | 1 - drivers/pci/probe.c | 3 - drivers/platform/x86/Kconfig | 1 + drivers/platform/x86/amd/hsmp.c | 50 +- drivers/platform/x86/dell/dell-smbios-base.c | 103 +- drivers/platform/x86/touchscreen_dmi.c | 59 +- drivers/pnp/base.h | 1 + drivers/pnp/driver.c | 6 + drivers/ptp/ptp_chardev.c | 3 +- drivers/scsi/device_handler/scsi_dh_alua.c | 31 +- drivers/scsi/mpi3mr/mpi3mr_transport.c | 2 +- drivers/scsi/mpt3sas/mpt3sas_scsih.c | 4 +- drivers/scsi/qedf/qedf.h | 1 + drivers/scsi/qedf/qedf_main.c | 47 +- drivers/scsi/scsi.c | 7 + drivers/scsi/sr.h | 2 +- drivers/scsi/sr_ioctl.c | 5 +- drivers/soc/fsl/qe/qmc.c | 32 +- drivers/soundwire/slave.c | 13 +- drivers/thermal/thermal_core.c | 35 +- drivers/thermal/thermal_core.h | 2 + drivers/thermal/thermal_debugfs.c | 18 +- drivers/thermal/thermal_trip.c | 20 +- drivers/ufs/core/ufs-mcq.c | 17 +- fs/bcachefs/btree_locking.c | 1 + fs/bcachefs/move.c | 16 +- fs/btrfs/btrfs_inode.h | 10 + fs/btrfs/disk-io.c | 10 +- fs/btrfs/extent_io.c | 60 +- fs/btrfs/file.c | 16 + fs/btrfs/ordered-data.c | 31 + fs/btrfs/tree-log.c | 17 +- fs/nilfs2/dir.c | 2 +- fs/nilfs2/segment.c | 3 + fs/proc/base.c | 2 +- fs/smb/client/smb2pdu.c | 3 - fs/smb/client/smb2transport.c | 2 +- include/dt-bindings/sound/audio-graph.h | 26 + include/linux/acpi.h | 1 + include/linux/atomic/atomic-arch-fallback.h | 6 +- include/linux/atomic/atomic-instrumented.h | 8 +- include/linux/atomic/atomic-long.h | 4 +- include/linux/cdrom.h | 2 +- include/linux/firmware/cirrus/cs_dsp.h | 10 +- include/linux/firmware/mediatek/mtk-adsp-ipc.h | 2 +- include/linux/huge_mm.h | 10 +- include/linux/i2c.h | 1 - include/linux/iommu.h | 2 +- include/linux/ksm.h | 17 +- include/linux/lockdep.h | 5 - include/linux/mm_types.h | 2 +- include/linux/pci.h | 2 - include/linux/pnp.h | 6 +- include/net/rtnetlink.h | 1 + include/net/tcp_ao.h | 7 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net/sunrpc/auth_gss/svcauth_gss.c | 2 +- net/unix/af_unix.c | 90 +- net/unix/diag.c | 12 +- net/wireless/core.c | 2 +- net/wireless/pmsr.c | 8 +- net/wireless/rdev-ops.h | 6 +- net/wireless/scan.c | 50 +- net/wireless/sysfs.c | 4 +- net/wireless/util.c | 7 +- net/xdp/xsk.c | 5 +- scripts/atomic/kerneldoc/sub_and_test | 2 +- scripts/const_structs.checkpatch | 1 + scripts/kconfig/confdata.c | 13 - scripts/kconfig/expr.c | 29 - scripts/kconfig/expr.h | 1 - scripts/kconfig/gconf.c | 3 +- scripts/kconfig/menu.c | 2 - scripts/link-vmlinux.sh | 2 +- scripts/mod/modpost.c | 5 +- security/tomoyo/Kconfig | 2 +- security/tomoyo/common.c | 2 +- sound/hda/intel-sdw-acpi.c | 6 +- sound/pci/hda/tas2781_hda_i2c.c | 4 +- sound/soc/amd/acp-es8336.c | 4 +- sound/soc/amd/acp/acp-i2s.c | 19 +- sound/soc/amd/acp/acp-legacy-common.c | 1 + sound/soc/amd/acp/acp-pci.c | 1 + sound/soc/amd/acp/acp-pdm.c | 1 + sound/soc/amd/acp/acp-platform.c | 14 + sound/soc/amd/acp/acp-rembrandt.c | 7 +- sound/soc/amd/acp/acp-renoir.c | 2 - sound/soc/amd/acp/acp63.c | 7 +- sound/soc/amd/acp/acp70.c | 2 - sound/soc/amd/acp/amd.h | 3 +- sound/soc/amd/ps/ps-mach.c | 1 + sound/soc/amd/renoir/acp3x-rn.c | 1 + sound/soc/amd/yc/acp6x-mach.c | 1 + sound/soc/codecs/Kconfig | 56 + sound/soc/codecs/Makefile | 19 + sound/soc/codecs/adau7118.c | 6 +- sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4458.c | 12 +- sound/soc/codecs/ak4613.c | 2 +- sound/soc/codecs/ak4619.c | 912 ++++++ sound/soc/codecs/audio-iio-aux.c | 83 +- sound/soc/codecs/aw87390.c | 2 +- sound/soc/codecs/aw88261.c | 2 +- sound/soc/codecs/aw88395/aw88395.c | 4 +- sound/soc/codecs/aw88395/aw88395_lib.c | 51 +- sound/soc/codecs/aw88399.c | 4 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-lib.c | 4 +- sound/soc/codecs/cs35l41.c | 3 +- sound/soc/codecs/cs35l56-sdw.c | 75 - sound/soc/codecs/cs35l56-shared.c | 133 +- sound/soc/codecs/cs35l56.c | 211 +- sound/soc/codecs/cs35l56.h | 2 - sound/soc/codecs/cs530x-i2c.c | 72 + sound/soc/codecs/cs530x.c | 971 +++++++ sound/soc/codecs/cs530x.h | 223 ++ sound/soc/codecs/cs53l30.c | 3 +- sound/soc/codecs/cx2072x.c | 5 - sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/es8311.c | 973 +++++++ sound/soc/codecs/es8311.h | 162 ++ sound/soc/codecs/es8326.c | 58 +- sound/soc/codecs/framer-codec.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/idt821034.c | 2 +- sound/soc/codecs/jz4760.c | 2 +- sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/lpass-macro-common.c | 23 + sound/soc/codecs/lpass-macro-common.h | 41 + sound/soc/codecs/lpass-rx-macro.c | 623 ++-- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 31 +- sound/soc/codecs/lpass-wsa-macro.c | 644 ++++- sound/soc/codecs/max98088.c | 10 +- sound/soc/codecs/max98390.c | 1 - sound/soc/codecs/max98504.c | 6 +- sound/soc/codecs/mt6358.c | 38 +- sound/soc/codecs/nau8822.c | 76 +- sound/soc/codecs/nau8822.h | 1 + sound/soc/codecs/nau8824.c | 21 +- sound/soc/codecs/nau8824.h | 1 + sound/soc/codecs/pcm3168a.c | 3 +- sound/soc/codecs/pcm512x-i2c.c | 2 + sound/soc/codecs/pcm512x-spi.c | 2 + sound/soc/codecs/pcm6240.c | 67 +- sound/soc/codecs/peb2466.c | 2 +- sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/rt1318.c | 1354 +++++++++ sound/soc/codecs/rt1318.h | 342 +++ sound/soc/codecs/rt1320-sdw.c | 2260 +++++++++++++++ sound/soc/codecs/rt1320-sdw.h | 94 + sound/soc/codecs/rt711-sdca.c | 72 + sound/soc/codecs/rt711-sdca.h | 1 + sound/soc/codecs/rt712-sdca-sdw.c | 36 +- sound/soc/codecs/rt712-sdca-sdw.h | 95 +- sound/soc/codecs/rt712-sdca.c | 665 ++++- sound/soc/codecs/rt712-sdca.h | 48 +- sound/soc/codecs/simple-mux.c | 55 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 21 +- sound/soc/codecs/tas2781-fmwlib.c | 89 +- sound/soc/codecs/tas2781-i2c.c | 235 +- sound/soc/codecs/tas5086.c | 27 +- sound/soc/codecs/tlv320adc3xxx.c | 106 +- sound/soc/codecs/tlv320adcx140.c | 1 - sound/soc/codecs/tlv320aic31xx.c | 1 - sound/soc/codecs/ts3a227e.c | 1 - sound/soc/codecs/wcd-mbhc-v2.c | 4 +- sound/soc/codecs/wcd-mbhc-v2.h | 4 +- sound/soc/codecs/wcd9335.c | 128 +- sound/soc/codecs/wcd934x.c | 72 +- sound/soc/codecs/wcd937x-sdw.c | 1137 ++++++++ sound/soc/codecs/wcd937x.c | 2971 ++++++++++++++++++++ sound/soc/codecs/wcd937x.h | 624 ++++ sound/soc/codecs/wcd938x-sdw.c | 4 +- sound/soc/codecs/wcd938x.c | 36 +- sound/soc/codecs/wcd938x.h | 10 +- sound/soc/codecs/wcd939x-sdw.c | 4 +- sound/soc/codecs/wcd939x.c | 176 +- sound/soc/codecs/wcd939x.h | 18 +- sound/soc/codecs/wm0010.c | 8 - sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 11 +- sound/soc/codecs/wsa884x.c | 10 +- sound/soc/fsl/Kconfig | 18 +- sound/soc/fsl/Makefile | 4 +- sound/soc/fsl/fsl-asoc-card.c | 383 ++- sound/soc/fsl/fsl_aud2htx.c | 11 +- sound/soc/fsl/fsl_audmix.c | 16 +- sound/soc/fsl/fsl_easrc.c | 10 +- sound/soc/fsl/fsl_mqs.c | 46 +- sound/soc/fsl/fsl_qmc_audio.c | 591 ++-- sound/soc/fsl/fsl_rpmsg.c | 9 + sound/soc/fsl/fsl_sai.c | 141 +- sound/soc/fsl/fsl_sai.h | 4 +- sound/soc/fsl/fsl_xcvr.c | 181 +- sound/soc/fsl/fsl_xcvr.h | 91 + sound/soc/fsl/imx-audmix.c | 79 +- sound/soc/fsl/imx-es8328.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 1 + sound/soc/fsl/imx-rpmsg.c | 2 - sound/soc/fsl/imx-spdif.c | 103 - sound/soc/fsl/lpc3xxx-i2s.c | 375 +++ sound/soc/fsl/lpc3xxx-i2s.h | 80 + sound/soc/fsl/lpc3xxx-pcm.c | 72 + sound/soc/generic/audio-graph-card.c | 113 +- .../soc/generic/audio-graph-card2-custom-sample.c | 3 +- sound/soc/generic/audio-graph-card2.c | 191 +- sound/soc/generic/simple-card-utils.c | 129 +- sound/soc/generic/simple-card.c | 71 +- sound/soc/generic/test-component.c | 2 +- sound/soc/intel/avs/boards/es8336.c | 8 +- sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/avs/topology.c | 2 +- sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 6 +- sound/soc/intel/boards/sof_board_helpers.h | 2 - sound/soc/intel/boards/sof_da7219.c | 55 +- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 114 +- sound/soc/intel/boards/sof_nau8825.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 64 +- sound/soc/intel/boards/sof_sdw_common.h | 6 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 11 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 11 +- sound/soc/intel/boards/sof_sdw_rt700.c | 11 +- sound/soc/intel/boards/sof_sdw_rt711.c | 11 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 2 +- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 14 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 11 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 50 + sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 50 + sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/axg-frddr.c | 4 +- sound/soc/meson/axg-tdm.h | 2 +- sound/soc/meson/axg-toddr.c | 4 +- sound/soc/qcom/common.c | 35 + sound/soc/qcom/common.h | 3 + sound/soc/qcom/lpass-cpu.c | 4 + sound/soc/qcom/qdsp6/audioreach.c | 30 +- sound/soc/qcom/qdsp6/audioreach.h | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 3 + sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 21 +- sound/soc/qcom/qdsp6/topology.c | 38 +- sound/soc/qcom/sc8280xp.c | 15 + sound/soc/qcom/x1e80100.c | 38 + sound/soc/rockchip/rockchip_i2s.c | 1 - sound/soc/rockchip/rockchip_spdif.c | 1 - sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/aries_wm8994.c | 2 +- sound/soc/samsung/midas_wm1811.c | 348 ++- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-core.c | 4 +- sound/soc/soc-dai.c | 18 +- sound/soc/soc-dapm.c | 49 +- sound/soc/soc-ops.c | 26 +- sound/soc/soc-pcm.c | 26 +- sound/soc/soc-topology.c | 572 ++-- sound/soc/soc-utils.c | 19 +- sound/soc/sof/intel/hda.c | 4 +- sound/soc/sof/intel/pci-tgl.c | 4 +- sound/soc/sof/ipc3-topology.c | 14 +- sound/soc/sof/ipc4-topology.c | 71 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sof/sof-audio.c | 20 +- sound/soc/sof/sof-audio.h | 9 +- sound/soc/sof/topology.c | 4 +- sound/soc/tegra/tegra210_i2s.c | 71 +- sound/soc/tegra/tegra210_i2s.h | 2 + tools/arch/arm64/include/asm/cputype.h | 6 + tools/arch/x86/include/asm/msr-index.h | 9 +- tools/arch/x86/include/uapi/asm/kvm.h | 22 +- tools/include/uapi/asm-generic/unistd.h | 5 +- tools/include/uapi/drm/i915_drm.h | 31 +- tools/include/uapi/linux/kvm.h | 4 +- tools/include/uapi/linux/stat.h | 4 +- tools/lib/bpf/features.c | 3 +- tools/perf/Makefile.perf | 1 + .../perf/arch/mips/entry/syscalls/syscall_n64.tbl | 1 + tools/perf/arch/powerpc/entry/syscalls/syscall.tbl | 1 + tools/perf/arch/s390/entry/syscalls/syscall.tbl | 1 + tools/perf/arch/x86/entry/syscalls/syscall_64.tbl | 3 +- tools/perf/builtin-record.c | 6 +- tools/perf/builtin-trace.c | 2 +- .../beauty/arch/x86/include/asm/irq_vectors.h | 8 +- tools/perf/trace/beauty/include/linux/socket.h | 3 +- tools/perf/trace/beauty/include/uapi/linux/fcntl.h | 14 +- tools/perf/trace/beauty/include/uapi/linux/prctl.h | 22 + tools/perf/trace/beauty/include/uapi/linux/stat.h | 4 +- tools/power/cpupower/utils/helpers/amd.c | 26 +- tools/testing/cxl/test/mem.c | 1 + tools/testing/selftests/alsa/Makefile | 2 +- .../selftests/bpf/progs/test_sk_storage_tracing.c | 2 +- tools/testing/selftests/cachestat/test_cachestat.c | 1 + .../selftests/filesystems/overlayfs/dev_in_maps.c | 1 + tools/testing/selftests/ftrace/config | 26 +- .../ftrace/test.d/dynevent/test_duplicates.tc | 2 +- .../ftrace/test.d/filter/event-filter-function.tc | 20 +- .../ftrace/test.d/kprobe/kprobe_eventname.tc | 3 +- tools/testing/selftests/futex/Makefile | 2 - tools/testing/selftests/futex/functional/Makefile | 2 +- .../selftests/futex/functional/futex_requeue_pi.c | 2 +- tools/testing/selftests/kvm/Makefile | 1 + .../selftests/kvm/s390x/shared_zeropage_test.c | 111 + tools/testing/selftests/net/hsr/config | 1 + tools/testing/selftests/net/lib.sh | 18 +- tools/testing/selftests/openat2/openat2_test.c | 1 + 661 files changed, 23899 insertions(+), 5937 deletions(-) create mode 100644 Documentation/devicetree/bindings/input/ilitek,ili2901.yaml delete mode 100644 Documentation/devicetree/bindings/sound/ak4104.txt delete mode 100644 Documentation/devicetree/bindings/sound/ak4554.txt delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml create mode 100644 Documentation/devicetree/bindings/sound/asahi-kasei,ak4104.yaml rename Documentation/devicetree/bindings/sound/{ak4375.yaml => asahi-kasei,ak4375.yaml} (94%) create mode 100644 Documentation/devicetree/bindings/sound/asahi-kasei,ak4554.yaml rename Documentation/devicetree/bindings/sound/{ak4613.yaml => asahi-kasei,ak4613.yaml} (94%) create mode 100644 Documentation/devicetree/bindings/sound/asahi-kasei,ak4619.yaml rename Documentation/devicetree/bindings/sound/{ak4642.yaml => asahi-kasei,ak4642.yaml} (94%) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs530x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt delete mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt delete mode 100644 Documentation/devicetree/bindings/sound/everest,es7134.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es71x4.yaml delete mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml rename Documentation/devicetree/bindings/sound/{sgtl5000.yaml => fsl,sgtl5000.yaml} (97%) rename Documentation/devicetree/bindings/sound/{linux,spdif-dit.yaml => linux,spdif.yaml} (75%) delete mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98088.yaml rename Documentation/devicetree/bindings/sound/{zl38060.yaml => mscc,zl38060.yaml} (96%) create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/omap-mcpdm.txt delete mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml rename Documentation/devicetree/bindings/sound/{rt1019.yaml => realtek,rt1019.yaml} (90%) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5514.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5631.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5645.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5659.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5677.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5514.txt delete mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt delete mode 100644 Documentation/devicetree/bindings/sound/rt5645.txt delete mode 100644 Documentation/devicetree/bindings/sound/rt5659.txt delete mode 100644 Documentation/devicetree/bindings/sound/rt5677.txt delete mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt delete mode 100644 Documentation/devicetree/bindings/sound/tas571x.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,omap4-mcpdm.yaml rename Documentation/devicetree/bindings/sound/{tas2562.yaml => ti,tas2562.yaml} (97%) rename Documentation/devicetree/bindings/sound/{tas2770.yaml => ti,tas2770.yaml} (97%) rename Documentation/devicetree/bindings/sound/{tas27xx.yaml => ti,tas27xx.yaml} (97%) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas57xx.yaml rename Documentation/devicetree/bindings/sound/{tas5805m.yaml => ti,tas5805m.yaml} (95%) rename Documentation/devicetree/bindings/sound/{tlv320adcx140.yaml => ti,tlv320adcx140.yaml} (99%) rename Documentation/devicetree/bindings/sound/{wm8750.yaml => wlf,wm8750.yaml} (92%) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8782.yaml create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8804.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8782.txt delete mode 100644 Documentation/devicetree/bindings/sound/wm8804.txt rename {include/linux => drivers/cpufreq}/amd-pstate.h (82%) create mode 100644 include/dt-bindings/sound/audio-graph.h create mode 100644 include/sound/rt1318.h create mode 100644 sound/soc/codecs/ak4619.c create mode 100644 sound/soc/codecs/cs530x-i2c.c create mode 100644 sound/soc/codecs/cs530x.c create mode 100644 sound/soc/codecs/cs530x.h create mode 100644 sound/soc/codecs/es8311.c create mode 100644 sound/soc/codecs/es8311.h create mode 100644 sound/soc/codecs/rt1318.c create mode 100644 sound/soc/codecs/rt1318.h create mode 100644 sound/soc/codecs/rt1320-sdw.c create mode 100644 sound/soc/codecs/rt1320-sdw.h create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h delete mode 100644 sound/soc/fsl/imx-spdif.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c create mode 100644 tools/testing/selftests/kvm/s390x/shared_zeropage_test.c

2 1

[PATCH v3 0/3] soundwire: bus: suppress probe deferral errors
by Johan Hovold 15 Jul '24

15 Jul '24

The soundwire bus code is incorrectly logging probe deferrals as errors. This series addresses that, cleans up the soundwire bus logging and drops the unused soudwire driver name field. Vinod, I booted linux-next yesterday and realised that these bogus error messages are still spamming the logs when booting x1e80100 and making it harder to see the valid warnings (which there were too many of). Would be nice to have this fixed in 6.11: [ 18.815950] wsa884x-codec sdw:4:0:0217:0204:00:0: Probe of wsa884x-codec failed: -517 [ 18.825093] wsa884x-codec sdw:1:0:0217:0204:00:0: Probe of wsa884x-codec failed: -517 [ 18.832335] wsa884x-codec sdw:4:0:0217:0204:00:1: Probe of wsa884x-codec failed: -517 [ 18.840870] wsa884x-codec sdw:1:0:0217:0204:00:1: Probe of wsa884x-codec failed: -517 [ 18.848463] wsa884x-codec sdw:1:0:0217:0204:00:0: Probe of wsa884x-codec failed: -517 Johan Changes in v3 - amend the update status error message with "at probe" to make it more descriptive (it's already unique) - drop the patch dropping a probe debug message Changes in v2 - drop probe error message completely - drop soundwire driver name field - cleanup probe warning messages - drop probe debug message Johan Hovold (3): soundwire: bus: suppress probe deferral errors soundwire: bus: drop unused driver name field soundwire: bus: clean up probe warnings drivers/soundwire/bus_type.c | 19 ++++--------------- include/linux/soundwire/sdw.h | 2 -- 2 files changed, 4 insertions(+), 17 deletions(-) -- 2.44.2

2 5

[PATCH v2] ASoC: codecs: wsa884x: Implement temperature reading and hwmon
by Krzysztof Kozlowski 14 Jul '24

14 Jul '24

Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Add missing dependency on HWMON for y!=m builds (kernel test robot report: undefined reference to `devm_hwmon_device_register_with_info'). --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wsa884x.c | 197 +++++++++++++++++++++++++++++++++++++ 2 files changed, 198 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 97bb69c9848d..09a0b209bc2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2447,6 +2447,7 @@ config SND_SOC_WSA883X config SND_SOC_WSA884X tristate "WSA884X Codec" depends on SOUNDWIRE + depends on HWMON || !HWMON select REGMAP_SOUNDWIRE help This enables support for Qualcomm WSA8840/WSA8845/WSA8845H Class-D diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 7b19df9c1728..5f4acd66f942 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include <linux/bitfield.h> +#include <linux/cleanup.h> #include <linux/device.h> #include <linux/gpio/consumer.h> +#include <linux/hwmon.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1650,6 +1691,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1661,6 +1706,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1800,6 +1849,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1849,6 +2036,7 @@ static int wsa884x_probe(struct sdw_slave *pdev, { struct device *dev = &pdev->dev; struct wsa884x_priv *wsa884x; + struct device *hwmon; unsigned int i; int ret; @@ -1856,6 +2044,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1913,6 +2103,13 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); -- 2.43.0

1 2

Handling complex matrix mixers in ALSA
by Asahi Lina 13 Jul '24

13 Jul '24

Hi, I'm reverse engineering and implementing support for the RME Digiface USB, which is an ADAT interface with a non-class-compliant interface (probably similar to other RME interfaces like the MADIface, but I don't have any others to test). The basic audio streaming works fine with an entry in quirks-table.h and a format quirk to set the system sample rate in quirks.c. Now I need to figure out how to implement the mixer controls. Currently I have the snd-usb-audio driver claiming only interface #0 (streaming) and I use a Python script to control the mixer/settings directly with libusb (control transfers and interface #1). This works fine and there's some prior art for this in the firewire world (for example, snd-dice doesn't do any mixer control stuff and you have to use ffado-mixer to control everything from userspace) but I assume it's not really the best way to go? The problem is that the device has a 66x34 matrix mixer, with up to 2048 cross points enabled at once. Exposing each cross point as an ALSA mixer control (similar to how mixer_scarlett2.c does it) would mean 2244 controls just for the mixer... which seems like a bit too much. On top of that there is also VU meter feedback for all the inputs/outputs, as well as general fader controls for each output and global output configs and status. I'm not sure about the VU meters, but everything else sounds like it would map fine to normal mixer controls. Is there some recommended way to expose this kind of matrix mixer interface to userspace? I think for something like this you pretty much have to rely on device-specific tools to make the UX manageable, so maybe hwdep... but at least exposing everything as an ALSA control would have the advantage of supporting save/restore with something like alsactl. So I don't really know what's the way to go here. System settings/general status/output faders go via control transfers, while interface #1 has an interrupt IN endpoint (streaming state feedback, not very useful) and two bulk endpoints (matrix mixer control out, VU meter data in). There's another pair of bulk endpoints in interface #2 which I'm guessing are for firmware updates (I haven't looked at that part). So in principle it's not crazy to expose all the system controls/output faders as mixer controls in ALSA and leave interface #1 entirely unclaimed so a userspace program can directly configure the matrix mixer and access VU meter levels. There is a global mixer enable bit (controlled via ctl transfer), so if that is exposed as an ALSA control and disabled by default the interface will operate as a 1:1 in/out interface without needing any custom userspace to configure the mixer. There's one other quirky thing: it also needs a way to set the sample rate as a mixer control, because you need to be able to configure the rate even when the PCM device is not open (since that affects stand-alone mixer operation). I imagine the right logic here would be to have a selector control for the system sample rate, and automatically change it and lock it when the PCM is opened with a given rate? Any thoughts welcome ^^ ~~ Lina

6 8

[PATCH v3] ASoC: rt1318: Add RT1318 audio amplifier driver
by Jack Yu 12 Jul '24

12 Jul '24

This is the initial i2s-based amplifier driver for rt1318. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- include/sound/rt1318.h | 16 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1318.c | 1354 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1318.h | 342 ++++++++++ 5 files changed, 1719 insertions(+) create mode 100644 include/sound/rt1318.h create mode 100644 sound/soc/codecs/rt1318.c create mode 100644 sound/soc/codecs/rt1318.h diff --git a/include/sound/rt1318.h b/include/sound/rt1318.h new file mode 100644 index 000000000000..fe6bff06036c --- /dev/null +++ b/include/sound/rt1318.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1318.h -- Platform data for RT1318 + * + * Copyright 2024 Realtek Semiconductor Corp. + */ + +#ifndef __LINUX_SND_RT1318_H +#define __LINUX_SND_RT1318_H + +struct rt1318_platform_data { + unsigned int init_r0_l; + unsigned int init_r0_r; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7b99556f24d3..a99cb26eb280 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -221,6 +221,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT722_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW + imply SND_SOC_RT1318 imply SND_SOC_RT1318_SDW imply SND_SOC_RT1320_SDW imply SND_SOC_RT9120 @@ -1576,6 +1577,10 @@ config SND_SOC_RT1316_SDW depends on SOUNDWIRE select REGMAP_SOUNDWIRE +config SND_SOC_RT1318 + tristate + depends on I2C + config SND_SOC_RT1318_SDW tristate "Realtek RT1318 Codec - SDW" depends on SOUNDWIRE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ca69f35cc0f7..ad0f77224c74 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -222,6 +222,7 @@ snd-soc-rt1305-y := rt1305.o snd-soc-rt1308-y := rt1308.o snd-soc-rt1308-sdw-y := rt1308-sdw.o snd-soc-rt1316-sdw-y := rt1316-sdw.o +snd-soc-rt1318-y := rt1318.o snd-soc-rt1318-sdw-y := rt1318-sdw.o snd-soc-rt1320-sdw-y := rt1320-sdw.o snd-soc-rt274-y := rt274.o @@ -618,6 +619,7 @@ obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT1308_SDW) += snd-soc-rt1308-sdw.o obj-$(CONFIG_SND_SOC_RT1316_SDW) += snd-soc-rt1316-sdw.o +obj-$(CONFIG_SND_SOC_RT1318) += snd-soc-rt1318.o obj-$(CONFIG_SND_SOC_RT1318_SDW) += snd-soc-rt1318-sdw.o obj-$(CONFIG_SND_SOC_RT1320_SDW) += snd-soc-rt1320-sdw.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o diff --git a/sound/soc/codecs/rt1318.c b/sound/soc/codecs/rt1318.c new file mode 100644 index 000000000000..4e47db4fc7fb --- /dev/null +++ b/sound/soc/codecs/rt1318.c @@ -0,0 +1,1354 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1318.c -- RT1318 ALSA SoC audio amplifier driver +// Author: Jack Yu <jack.yu(a)realtek.com> +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/acpi.h> +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt1318.h> + +#include "rt1318.h" + +static struct reg_sequence init_list[] = { + { 0x0000C000, 0x01}, + { 0x0000F20D, 0x00}, + { 0x0000F212, 0x3E}, + { 0x0000C001, 0x02}, + { 0x0000C003, 0x22}, + { 0x0000C004, 0x44}, + { 0x0000C005, 0x44}, + { 0x0000C007, 0x64}, + { 0x0000C00E, 0xE7}, + { 0x0000F223, 0x7F}, + { 0x0000F224, 0xDB}, + { 0x0000F225, 0xEE}, + { 0x0000F226, 0x3F}, + { 0x0000F227, 0x0F}, + { 0x0000F21A, 0x78}, + { 0x0000F242, 0x3C}, + { 0x0000C120, 0x40}, + { 0x0000C125, 0x03}, + { 0x0000C321, 0x0A}, + { 0x0000C200, 0xD8}, + { 0x0000C201, 0x27}, + { 0x0000C202, 0x0F}, + { 0x0000C400, 0x0E}, + { 0x0000C401, 0x43}, + { 0x0000C402, 0xE0}, + { 0x0000C403, 0x00}, + { 0x0000C404, 0x4C}, + { 0x0000C406, 0x40}, + { 0x0000C407, 0x02}, + { 0x0000C408, 0x3F}, + { 0x0000C300, 0x01}, + { 0x0000C125, 0x03}, + { 0x0000DF00, 0x10}, + { 0x0000F20B, 0x2A}, + { 0x0000DF5F, 0x01}, + { 0x0000DF60, 0xA7}, + { 0x0000C203, 0x84}, + { 0x0000C206, 0x78}, + { 0x0000F10A, 0x09}, + { 0x0000F10B, 0x4C}, + { 0x0000F104, 0xF4}, + { 0x0000F105, 0x03}, + { 0x0000F109, 0xE0}, + { 0x0000F10B, 0x5C}, + { 0x0000F104, 0xF4}, + { 0x0000F105, 0x04}, + { 0x0000F109, 0x65}, + { 0x0000F10B, 0x5C}, + { 0x0000F104, 0xF4}, + { 0x0000F105, 0x02}, + { 0x0000F109, 0x30}, + { 0x0000F10B, 0x5C}, + { 0x0000E706, 0x0F}, + { 0x0000E707, 0x30}, + { 0x0000E806, 0x0F}, + { 0x0000E807, 0x30}, + { 0x0000CE04, 0x03}, + { 0x0000CE05, 0x5F}, + { 0x0000CE06, 0xA2}, + { 0x0000CE07, 0x6B}, + { 0x0000CF04, 0x03}, + { 0x0000CF05, 0x5F}, + { 0x0000CF06, 0xA2}, + { 0x0000CF07, 0x6B}, + { 0x0000CE60, 0xE3}, + { 0x0000C130, 0x51}, + { 0x0000E000, 0xA8}, + { 0x0000F102, 0x00}, + { 0x0000F103, 0x00}, + { 0x0000F104, 0xF5}, + { 0x0000F105, 0x23}, + { 0x0000F109, 0x04}, + { 0x0000F10A, 0x0B}, + { 0x0000F10B, 0x4C}, + { 0x0000F10B, 0x5C}, + { 0x41001888, 0x00}, + { 0x0000C121, 0x0B}, + { 0x0000F102, 0x00}, + { 0x0000F103, 0x00}, + { 0x0000F104, 0xF5}, + { 0x0000F105, 0x23}, + { 0x0000F109, 0x00}, + { 0x0000F10A, 0x0B}, + { 0x0000F10B, 0x4C}, + { 0x0000F10B, 0x5C}, + { 0x0000F800, 0x20}, + { 0x0000CA00, 0x80}, + { 0x0000CA10, 0x00}, + { 0x0000CA02, 0x78}, + { 0x0000CA12, 0x78}, + { 0x0000ED00, 0x90}, + { 0x0000E604, 0x00}, + { 0x0000DB00, 0x0C}, + { 0x0000DD00, 0x0C}, + { 0x0000DC19, 0x00}, + { 0x0000DC1A, 0x6A}, + { 0x0000DC1B, 0xAA}, + { 0x0000DC1C, 0xAB}, + { 0x0000DC1D, 0x00}, + { 0x0000DC1E, 0x16}, + { 0x0000DC1F, 0xDB}, + { 0x0000DC20, 0x6D}, + { 0x0000DE19, 0x00}, + { 0x0000DE1A, 0x6A}, + { 0x0000DE1B, 0xAA}, + { 0x0000DE1C, 0xAB}, + { 0x0000DE1D, 0x00}, + { 0x0000DE1E, 0x16}, + { 0x0000DE1F, 0xDB}, + { 0x0000DE20, 0x6D}, + { 0x0000DB32, 0x00}, + { 0x0000DD32, 0x00}, + { 0x0000DB33, 0x0A}, + { 0x0000DD33, 0x0A}, + { 0x0000DB34, 0x1A}, + { 0x0000DD34, 0x1A}, + { 0x0000DB15, 0xEF}, + { 0x0000DD15, 0xEF}, + { 0x0000DB17, 0xEF}, + { 0x0000DD17, 0xEF}, + { 0x0000DB94, 0x70}, + { 0x0000DD94, 0x70}, + { 0x0000DB19, 0x40}, + { 0x0000DD19, 0x40}, + { 0x0000DB12, 0xC0}, + { 0x0000DD12, 0xC0}, + { 0x0000DB00, 0x4C}, + { 0x0000DB04, 0x05}, + { 0x0000DB05, 0x03}, + { 0x0000DD04, 0x05}, + { 0x0000DD05, 0x03}, + { 0x0000DBBB, 0x09}, + { 0x0000DBBC, 0x30}, + { 0x0000DBBD, 0xF0}, + { 0x0000DBBE, 0xF1}, + { 0x0000DDBB, 0x09}, + { 0x0000DDBC, 0x30}, + { 0x0000DDBD, 0xF0}, + { 0x0000DDBE, 0xF1}, + { 0x0000DB01, 0x79}, + { 0x0000DD01, 0x79}, + { 0x0000DB08, 0x40}, + { 0x0000DD08, 0x40}, + { 0x0000DC52, 0xEF}, + { 0x0000DE52, 0xEF}, + { 0x0000DB00, 0xCC}, + { 0x0000CC2C, 0x00}, + { 0x0000CC2D, 0x2A}, + { 0x0000CC2E, 0x83}, + { 0x0000CC2F, 0xA8}, + { 0x0000CD2C, 0x00}, + { 0x0000CD2D, 0x2A}, + { 0x0000CD2E, 0x83}, + { 0x0000CD2F, 0xA8}, + { 0x0000CC24, 0x00}, + { 0x0000CC25, 0x51}, + { 0x0000CC26, 0xEB}, + { 0x0000CC27, 0x85}, + { 0x0000CD24, 0x00}, + { 0x0000CD25, 0x51}, + { 0x0000CD26, 0xEB}, + { 0x0000CD27, 0x85}, + { 0x0000CC20, 0x00}, + { 0x0000CC21, 0x00}, + { 0x0000CC22, 0x43}, + { 0x0000CD20, 0x00}, + { 0x0000CD21, 0x00}, + { 0x0000CD22, 0x43}, + { 0x0000CC16, 0x0F}, + { 0x0000CC17, 0x00}, + { 0x0000CD16, 0x0F}, + { 0x0000CD17, 0x00}, + { 0x0000CC29, 0x5D}, + { 0x0000CC2A, 0xC0}, + { 0x0000CD29, 0x5D}, + { 0x0000CD2A, 0xC0}, + { 0x0000CC31, 0x20}, + { 0x0000CC32, 0x00}, + { 0x0000CC33, 0x00}, + { 0x0000CC34, 0x00}, + { 0x0000CD31, 0x20}, + { 0x0000CD32, 0x00}, + { 0x0000CD33, 0x00}, + { 0x0000CD34, 0x00}, + { 0x0000CC36, 0x79}, + { 0x0000CC37, 0x99}, + { 0x0000CC38, 0x99}, + { 0x0000CC39, 0x99}, + { 0x0000CD36, 0x79}, + { 0x0000CD37, 0x99}, + { 0x0000CD38, 0x99}, + { 0x0000CD39, 0x99}, + { 0x0000CC09, 0x00}, + { 0x0000CC0A, 0x07}, + { 0x0000CC0B, 0x5F}, + { 0x0000CC0C, 0x6F}, + { 0x0000CD09, 0x00}, + { 0x0000CD0A, 0x07}, + { 0x0000CD0B, 0x5F}, + { 0x0000CD0C, 0x6F}, + { 0x0000CC0E, 0x00}, + { 0x0000CC0F, 0x03}, + { 0x0000CC10, 0xAF}, + { 0x0000CC11, 0xB7}, + { 0x0000CD0E, 0x00}, + { 0x0000CD0F, 0x03}, + { 0x0000CD10, 0xAF}, + { 0x0000CD11, 0xB7}, + { 0x0000CCD6, 0x00}, + { 0x0000CCD7, 0x03}, + { 0x0000CDD6, 0x00}, + { 0x0000CDD7, 0x03}, + { 0x0000CCD8, 0x00}, + { 0x0000CCD9, 0x03}, + { 0x0000CDD8, 0x00}, + { 0x0000CDD9, 0x03}, + { 0x0000CCDA, 0x00}, + { 0x0000CCDB, 0x03}, + { 0x0000CDDA, 0x00}, + { 0x0000CDDB, 0x03}, + { 0x0000C320, 0x20}, + { 0x0000C203, 0x9C}, +}; +#define rt1318_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt1318_reg[] = { + { 0xc000, 0x00 }, + { 0xc001, 0x43 }, + { 0xc003, 0x22 }, + { 0xc004, 0x44 }, + { 0xc005, 0x44 }, + { 0xc006, 0x33 }, + { 0xc007, 0x64 }, + { 0xc008, 0x05 }, + { 0xc00a, 0xfc }, + { 0xc00b, 0x0f }, + { 0xc00c, 0x0e }, + { 0xc00d, 0xef }, + { 0xc00e, 0xe5 }, + { 0xc00f, 0xff }, + { 0xc120, 0xc0 }, + { 0xc121, 0x00 }, + { 0xc122, 0x00 }, + { 0xc123, 0x14 }, + { 0xc125, 0x00 }, + { 0xc130, 0x59 }, + { 0xc200, 0x00 }, + { 0xc201, 0x00 }, + { 0xc202, 0x00 }, + { 0xc203, 0x04 }, + { 0xc204, 0x00 }, + { 0xc205, 0x00 }, + { 0xc206, 0x68 }, + { 0xc207, 0x70 }, + { 0xc208, 0x00 }, + { 0xc20a, 0x00 }, + { 0xc20b, 0x01 }, + { 0xc20c, 0x7f }, + { 0xc20d, 0x01 }, + { 0xc20e, 0x7f }, + { 0xc300, 0x00 }, + { 0xc301, 0x00 }, + { 0xc303, 0x80 }, + { 0xc320, 0x00 }, + { 0xc321, 0x09 }, + { 0xc322, 0x02 }, + { 0xc400, 0x00 }, + { 0xc401, 0x00 }, + { 0xc402, 0x00 }, + { 0xc403, 0x00 }, + { 0xc404, 0x00 }, + { 0xc405, 0x00 }, + { 0xc406, 0x00 }, + { 0xc407, 0x00 }, + { 0xc408, 0x00 }, + { 0xc410, 0x04 }, + { 0xc430, 0x00 }, + { 0xc431, 0x00 }, + { 0xca00, 0x10 }, + { 0xca01, 0x00 }, + { 0xca02, 0x0b }, + { 0xca10, 0x10 }, + { 0xca11, 0x00 }, + { 0xca12, 0x0b }, + { 0xce04, 0x08 }, + { 0xce05, 0x00 }, + { 0xce06, 0x00 }, + { 0xce07, 0x00 }, + { 0xce60, 0x63 }, + { 0xcf04, 0x08 }, + { 0xcf05, 0x00 }, + { 0xcf06, 0x00 }, + { 0xcf07, 0x00 }, + { 0xdb00, 0x00 }, + { 0xdb08, 0x40 }, + { 0xdb12, 0x00 }, + { 0xdb35, 0x00 }, + { 0xdbb5, 0x00 }, + { 0xdbb6, 0x40 }, + { 0xdbb7, 0x00 }, + { 0xdbb8, 0x00 }, + { 0xdbc5, 0x00 }, + { 0xdbc6, 0x00 }, + { 0xdbc7, 0x00 }, + { 0xdbc8, 0x00 }, + { 0xdd08, 0x40 }, + { 0xdd12, 0x00 }, + { 0xdd35, 0x00 }, + { 0xddb5, 0x00 }, + { 0xddb6, 0x40 }, + { 0xddb7, 0x00 }, + { 0xddb8, 0x00 }, + { 0xddc5, 0x00 }, + { 0xddc6, 0x00 }, + { 0xddc7, 0x00 }, + { 0xddc8, 0x00 }, + { 0xdd93, 0x00 }, + { 0xdd94, 0x64 }, + { 0xdf00, 0x00 }, + { 0xdf5f, 0x00 }, + { 0xdf60, 0x00 }, + { 0xe000, 0x08 }, + { 0xe300, 0xa0 }, + { 0xe400, 0x22 }, + { 0xe706, 0x2f }, + { 0xe707, 0x2f }, + { 0xe806, 0x2f }, + { 0xe807, 0x2f }, + { 0xea00, 0x43 }, + { 0xed00, 0x80 }, + { 0xed01, 0x0f }, + { 0xed02, 0xff }, + { 0xed03, 0x00 }, + { 0xed04, 0x00 }, + { 0xed05, 0x0f }, + { 0xed06, 0xff }, + { 0xf010, 0x10 }, + { 0xf011, 0xec }, + { 0xf012, 0x68 }, + { 0xf013, 0x21 }, + { 0xf102, 0x00 }, + { 0xf103, 0x00 }, + { 0xf104, 0x00 }, + { 0xf105, 0x00 }, + { 0xf106, 0x00 }, + { 0xf107, 0x00 }, + { 0xf108, 0x00 }, + { 0xf109, 0x00 }, + { 0xf10a, 0x03 }, + { 0xf10b, 0x40 }, + { 0xf20b, 0x28 }, + { 0xf20d, 0x00 }, + { 0xf212, 0x00 }, + { 0xf21a, 0x00 }, + { 0xf223, 0x40 }, + { 0xf224, 0x00 }, + { 0xf225, 0x00 }, + { 0xf226, 0x00 }, + { 0xf227, 0x00 }, + { 0xf242, 0x0c }, + { 0xf800, 0x00 }, + { 0xf801, 0x12 }, + { 0xf802, 0xe0 }, + { 0xf803, 0x2f }, + { 0xf804, 0x00 }, + { 0xf805, 0x00 }, + { 0xf806, 0x07 }, + { 0xf807, 0xff }, +}; + +static bool rt1318_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0xc000: + case 0xc301: + case 0xc410: + case 0xc430 ... 0xc431: + case 0xdb06: + case 0xdb12: + case 0xdb1d ... 0xdb1f: + case 0xdb35: + case 0xdb37: + case 0xdb8a ... 0xdb92: + case 0xdbc5 ... 0xdbc8: + case 0xdc2b ... 0xdc49: + case 0xdd0b: + case 0xdd12: + case 0xdd1d ... 0xdd1f: + case 0xdd35: + case 0xdd8a ... 0xdd92: + case 0xddc5 ... 0xddc8: + case 0xde2b ... 0xde44: + case 0xdf4a ... 0xdf55: + case 0xe224 ... 0xe23b: + case 0xea01: + case 0xebc5: + case 0xebc8: + case 0xebcb ... 0xebcc: + case 0xed03 ... 0xed06: + case 0xf010 ... 0xf014: + return true; + + default: + return false; + } +} + +static bool rt1318_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0xc000 ... 0xc00f: + case 0xc120 ... 0xc130: + case 0xc200 ... 0xc20e: + case 0xc300 ... 0xc303: + case 0xc320 ... 0xc322: + case 0xc400 ... 0xc408: + case 0xc430 ... 0xc431: + case 0xca00 ... 0xca02: + case 0xca10 ... 0xca12: + case 0xcb00 ... 0xcb0b: + case 0xcc00 ... 0xcce5: + case 0xcd00 ... 0xcde5: + case 0xce00 ... 0xce6a: + case 0xcf00 ... 0xcf53: + case 0xd000 ... 0xd0cc: + case 0xd100 ... 0xd1b9: + case 0xdb00 ... 0xdc53: + case 0xdd00 ... 0xde53: + case 0xdf00 ... 0xdf6b: + case 0xe000: + case 0xe300: + case 0xe400: + case 0xe706 ... 0xe707: + case 0xe806 ... 0xe807: + case 0xea00: + case 0xeb00 ... 0xebcc: + case 0xec00 ... 0xecb9: + case 0xed00 ... 0xed06: + case 0xf010 ... 0xf014: + case 0xf102 ... 0xf10b: + case 0xf20b: + case 0xf20d ... 0xf242: + case 0xf800 ... 0xf807: + return true; + default: + return false; + } +} + +static int rt1318_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(rt1318->regmap, RT1318_PWR_STA1, + RT1318_PDB_CTRL_MASK, RT1318_PDB_CTRL_HIGH); + break; + + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(rt1318->regmap, RT1318_PWR_STA1, + RT1318_PDB_CTRL_MASK, RT1318_PDB_CTRL_LOW); + break; + + default: + break; + } + return 0; +} + +static int rt1318_dvol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + rt1318->rt1318_dvol = ucontrol->value.integer.value[0]; + + if (rt1318->rt1318_dvol <= RT1318_DVOL_STEP && rt1318->rt1318_dvol >= 0) { + regmap_write(rt1318->regmap, RT1318_DA_VOL_L_8, + rt1318->rt1318_dvol >> 8); + regmap_write(rt1318->regmap, RT1318_DA_VOL_L_1_7, + rt1318->rt1318_dvol & 0xff); + regmap_write(rt1318->regmap, RT1318_DA_VOL_R_8, + rt1318->rt1318_dvol >> 8); + regmap_write(rt1318->regmap, RT1318_DA_VOL_R_1_7, + rt1318->rt1318_dvol & 0xff); + return 1; + } + + return 0; +} + +static int rt1318_dvol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt1318->rt1318_dvol; + + return 0; +} + +static const struct snd_kcontrol_new rt1318_snd_controls[] = { + SOC_SINGLE_EXT("Amp Playback Volume", SND_SOC_NOPM, 0, 383, 0, + rt1318_dvol_get, rt1318_dvol_put), +}; + +static const struct snd_soc_dapm_widget rt1318_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, + rt1318_dac_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("Amp"), +}; + +static const struct snd_soc_dapm_route rt1318_dapm_routes[] = { + {"DAC", NULL, "AIF1RX"}, + {"Amp", NULL, "DAC"}, +}; + +static int rt1318_get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 4, 8, 16, 24}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt1318_clk_ip_info(struct snd_soc_component *component, int lrclk) +{ + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + switch (lrclk) { + case RT1318_LRCLK_48000: + case RT1318_LRCLK_44100: + case RT1318_LRCLK_16000: + regmap_update_bits(rt1318->regmap, RT1318_SRC_TCON, + RT1318_SRCIN_F12288_MASK | RT1318_SRCIN_DACLK_MASK, + RT1318_SRCIN_TCON4 | RT1318_DACLK_TCON4); + break; + case RT1318_LRCLK_96000: + regmap_update_bits(rt1318->regmap, RT1318_SRC_TCON, + RT1318_SRCIN_F12288_MASK | RT1318_SRCIN_DACLK_MASK, + RT1318_SRCIN_TCON4 | RT1318_DACLK_TCON2); + break; + case RT1318_LRCLK_192000: + regmap_update_bits(rt1318->regmap, RT1318_SRC_TCON, + RT1318_SRCIN_F12288_MASK | RT1318_SRCIN_DACLK_MASK, + RT1318_SRCIN_TCON4 | RT1318_DACLK_TCON1); + break; + default: + dev_err(component->dev, "Unsupported clock rate.\n"); + return -EINVAL; + } + + return 0; +} + +static int rt1318_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 rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + int data_len = 0, ch_len = 0; + int pre_div, ret; + + rt1318->lrck = params_rate(params); + pre_div = rt1318_get_clk_info(rt1318->sysclk, rt1318->lrck); + if (pre_div < 0) { + dev_err(component->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + ret = rt1318_clk_ip_info(component, rt1318->lrck); + if (ret < 0) { + dev_err(component->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + + switch (params_width(params)) { + case 16: + break; + case 20: + data_len = RT1318_I2S_DL_20; + ch_len = RT1318_I2S_DL_20; + break; + case 24: + data_len = RT1318_I2S_DL_24; + ch_len = RT1318_I2S_DL_24; + break; + case 32: + data_len = RT1318_I2S_DL_32; + ch_len = RT1318_I2S_DL_32; + break; + case 8: + data_len = RT1318_I2S_DL_8; + ch_len = RT1318_I2S_DL_8; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rt1318->regmap, RT1318_CLK2, + RT1318_DIV_AP_MASK | RT1318_DIV_DAMOD_MASK, + pre_div << RT1318_DIV_AP_SFT | + pre_div << RT1318_DIV_DAMOD_SFT); + regmap_update_bits(rt1318->regmap, RT1318_CLK3, + RT1318_AD_STO1_MASK | RT1318_AD_STO2_MASK, + pre_div << RT1318_AD_STO1_SFT | + pre_div << RT1318_AD_STO2_SFT); + regmap_update_bits(rt1318->regmap, RT1318_CLK4, + RT1318_AD_ANA_STO1_MASK | RT1318_AD_ANA_STO2_MASK, + pre_div << RT1318_AD_ANA_STO1_SFT | + pre_div << RT1318_AD_ANA_STO2_SFT); + regmap_update_bits(rt1318->regmap, RT1318_CLK5, + RT1318_DIV_FIFO_IN_MASK | RT1318_DIV_FIFO_OUT_MASK, + pre_div << RT1318_DIV_FIFO_IN_SFT | + pre_div << RT1318_DIV_FIFO_OUT_SFT); + regmap_update_bits(rt1318->regmap, RT1318_CLK6, + RT1318_DIV_NLMS_MASK | RT1318_DIV_AD_MONO_MASK | + RT1318_DIV_POST_G_MASK, pre_div << RT1318_DIV_NLMS_SFT | + pre_div << RT1318_DIV_AD_MONO_SFT | + pre_div << RT1318_DIV_POST_G_SFT); + + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL2, + RT1318_I2S_DL_MASK, data_len << RT1318_I2S_DL_SFT); + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL3, + RT1318_I2S_TX_CHL_MASK | RT1318_I2S_RX_CHL_MASK, + ch_len << RT1318_I2S_TX_CHL_SFT | + ch_len << RT1318_I2S_RX_CHL_SFT); + + return 0; +} + +static int rt1318_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, reg_val2 = 0; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val2 |= RT1318_TDM_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT1318_FMT_LEFT_J; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT1318_FMT_PCM_A_R; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT1318_FMT_PCM_B_R; + break; + + default: + return -EINVAL; + } + + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL1, + RT1318_I2S_FMT_MASK, reg_val); + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL1, + RT1318_TDM_BCLK_MASK, reg_val2); + + return 0; +} + +static int rt1318_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + int reg_val = 0; + + if (freq == rt1318->sysclk && clk_id == rt1318->sysclk_src) + return 0; + + switch (clk_id) { + case RT1318_SCLK_S_BCLK: + reg_val |= RT1318_SYSCLK_BCLK; + break; + case RT1318_SCLK_S_SDW: + reg_val |= RT1318_SYSCLK_SDW; + break; + case RT1318_SCLK_S_PLL2F: + reg_val |= RT1318_SYSCLK_PLL2F; + break; + case RT1318_SCLK_S_PLL2B: + reg_val |= RT1318_SYSCLK_PLL2B; + break; + case RT1318_SCLK_S_MCLK: + reg_val |= RT1318_SYSCLK_MCLK; + break; + case RT1318_SCLK_S_RC0: + reg_val |= RT1318_SYSCLK_RC1; + break; + case RT1318_SCLK_S_RC1: + reg_val |= RT1318_SYSCLK_RC2; + break; + case RT1318_SCLK_S_RC2: + reg_val |= RT1318_SYSCLK_RC3; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + rt1318->sysclk = freq; + rt1318->sysclk_src = clk_id; + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_SYSCLK_SEL_MASK, reg_val); + + return 0; +} + +static const struct pll_calc_map pll_preset_table[] = { + {512000, 4096000, 22, 190, 0, true, false}, + {1024000, 4096000, 22, 94, 0, true, false}, + {1024000, 16384000, 4, 190, 0, true, false}, + {1411200, 11289600, 6, 62, 0, true, false}, + {1536000, 12288000, 6, 62, 0, true, false}, + {2822400, 11289600, 6, 62, 0, true, false}, + {2822400, 45158400, 0, 62, 0, true, false}, + {2822400, 49152000, 0, 62, 0, true, false}, + {3072000, 12288000, 6, 62, 0, true, false}, + {3072000, 24576000, 2, 62, 0, true, false}, + {3072000, 49152000, 0, 62, 0, true, false}, + {6144000, 24576000, 2, 94, 4, false, false}, + {6144000, 49152000, 0, 30, 0, true, false}, + {6144000, 98304000, 0, 94, 4, false, true}, + {12288000, 49152000, 0, 62, 6, false, false}, +}; + +static int rt1318_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt1318_pll_code *pll_code) +{ + int max_n = RT1318_PLL_N_MAX, max_m = RT1318_PLL_M_MAX; + int i, k, red, n_t, pll_out, in_t, out_t; + int n = 0, m = 0, m_t = 0; + int red_t = abs(freq_out - freq_in); + bool m_bypass = false, k_bypass = false; + + if (RT1318_PLL_INP_MAX < freq_in || RT1318_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(pll_preset_table); i++) { + if (freq_in == pll_preset_table[i].pll_in && + freq_out == pll_preset_table[i].pll_out) { + k = pll_preset_table[i].k; + m = pll_preset_table[i].m; + n = pll_preset_table[i].n; + m_bypass = pll_preset_table[i].m_bp; + k_bypass = pll_preset_table[i].k_bp; + goto code_find; + } + } + + k = 100000000 / freq_out - 2; + if (k > RT1318_PLL_K_MAX) + k = RT1318_PLL_K_MAX; + if (k < 0) { + k = 0; + k_bypass = true; + } + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = freq_in / (k_bypass ? 1 : (k + 2)); + pll_out = freq_out / (n_t + 2); + if (in_t < 0) + continue; + if (in_t == pll_out) { + m_bypass = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + m_bypass = true; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - pll_out); + if (red < red_t) { + m_bypass = false; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + + pll_code->m_bp = m_bypass; + pll_code->k_bp = k_bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt1318_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_component *component = dai->component; + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + struct rt1318_pll_code pll_code; + int ret; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + rt1318->pll_in = 0; + rt1318->pll_out = 0; + return 0; + } + + if (source == rt1318->pll_src && freq_in == rt1318->pll_in && + freq_out == rt1318->pll_out) + return 0; + + switch (source) { + case RT1318_PLL_S_BCLK0: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_BCLK0); + break; + case RT1318_PLL_S_BCLK1: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_BCLK1); + break; + case RT1318_PLL_S_RC: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_RC); + break; + case RT1318_PLL_S_MCLK: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_MCLK); + break; + case RT1318_PLL_S_SDW_IN_PLL: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_SDW1); + break; + case RT1318_PLL_S_SDW_0: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_SDW2); + break; + case RT1318_PLL_S_SDW_1: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_SDW3); + break; + case RT1318_PLL_S_SDW_2: + regmap_update_bits(rt1318->regmap, RT1318_CLK1, + RT1318_PLLIN_MASK, RT1318_PLLIN_SDW4); + break; + default: + dev_err(component->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt1318_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + regmap_update_bits(rt1318->regmap, RT1318_PLL1_K, + RT1318_K_PLL1_MASK, pll_code.k_code); + regmap_update_bits(rt1318->regmap, RT1318_PLL1_M, + RT1318_M_PLL1_MASK, (pll_code.m_bp ? 0 : pll_code.m_code)); + regmap_update_bits(rt1318->regmap, RT1318_PLL1_N_8, + RT1318_N_8_PLL1_MASK, pll_code.n_code >> 8); + regmap_update_bits(rt1318->regmap, RT1318_PLL1_N_7_0, + RT1318_N_7_0_PLL1_MASK, pll_code.n_code); + + rt1318->pll_in = freq_in; + rt1318->pll_out = freq_out; + rt1318->pll_src = source; + + return 0; +} + +static int rt1318_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + unsigned int cn = 0, cl = 0, rx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 4: + cn |= RT1318_I2S_CH_TX_4CH; + cn |= RT1318_I2S_CH_RX_4CH; + break; + case 6: + cn |= RT1318_I2S_CH_TX_6CH; + cn |= RT1318_I2S_CH_RX_6CH; + break; + case 8: + cn |= RT1318_I2S_CH_TX_8CH; + cn |= RT1318_I2S_CH_RX_8CH; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + cl |= RT1318_I2S_TX_CHL_20; + cl |= RT1318_I2S_RX_CHL_20; + break; + case 24: + cl |= RT1318_I2S_TX_CHL_24; + cl |= RT1318_I2S_RX_CHL_24; + break; + case 32: + cl |= RT1318_I2S_TX_CHL_32; + cl |= RT1318_I2S_RX_CHL_32; + break; + case 8: + cl |= RT1318_I2S_TX_CHL_8; + cl |= RT1318_I2S_RX_CHL_8; + break; + case 16: + break; + default: + return -EINVAL; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + regmap_update_bits(rt1318->regmap, + RT1318_TDM_CTRL9, + RT1318_TDM_I2S_TX_L_DAC1_1_MASK | + RT1318_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1318_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit + 1) << RT1318_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + regmap_update_bits(rt1318->regmap, + RT1318_TDM_CTRL9, + RT1318_TDM_I2S_TX_L_DAC1_1_MASK | + RT1318_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit - 1) << RT1318_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1318_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL2, + RT1318_I2S_CH_TX_MASK | RT1318_I2S_CH_RX_MASK, cn); + regmap_update_bits(rt1318->regmap, RT1318_TDM_CTRL3, + RT1318_I2S_TX_CHL_MASK | RT1318_I2S_RX_CHL_MASK, cl); + +_set_tdm_err_: + return ret; +} + +static int rt1318_probe(struct snd_soc_component *component) +{ + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + rt1318->component = component; + + schedule_work(&rt1318->cali_work); + rt1318->rt1318_dvol = RT1318_DVOL_STEP; + + return 0; +} + +static void rt1318_remove(struct snd_soc_component *component) +{ + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + cancel_work_sync(&rt1318->cali_work); +} + +#ifdef CONFIG_PM +static int rt1318_suspend(struct snd_soc_component *component) +{ + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1318->regmap, true); + regcache_mark_dirty(rt1318->regmap); + return 0; +} + +static int rt1318_resume(struct snd_soc_component *component) +{ + struct rt1318_priv *rt1318 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1318->regmap, false); + regcache_sync(rt1318->regmap); + return 0; +} +#else +#define rt1318_suspend NULL +#define rt1318_resume NULL +#endif + +#define RT1318_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT1318_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt1318_aif_dai_ops = { + .hw_params = rt1318_hw_params, + .set_fmt = rt1318_set_dai_fmt, + .set_sysclk = rt1318_set_dai_sysclk, + .set_pll = rt1318_set_dai_pll, + .set_tdm_slot = rt1318_set_tdm_slot, +}; + +static struct snd_soc_dai_driver rt1318_dai[] = { + { + .name = "rt1318-aif", + .id = 0, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT1318_STEREO_RATES, + .formats = RT1318_FORMATS, + }, + .ops = &rt1318_aif_dai_ops, + } +}; + +static const struct snd_soc_component_driver soc_component_dev_rt1318 = { + .probe = rt1318_probe, + .remove = rt1318_remove, + .suspend = rt1318_suspend, + .resume = rt1318_resume, + .controls = rt1318_snd_controls, + .num_controls = ARRAY_SIZE(rt1318_snd_controls), + .dapm_widgets = rt1318_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1318_dapm_widgets), + .dapm_routes = rt1318_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1318_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt1318_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1318_readable_register, + .volatile_reg = rt1318_volatile_register, + .max_register = 0x41001888, + .reg_defaults = rt1318_reg, + .num_reg_defaults = ARRAY_SIZE(rt1318_reg), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt1318_i2c_id[] = { + { "rt1318", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1318_i2c_id); + +static const struct of_device_id rt1318_of_match[] = { + { .compatible = "realtek,rt1318", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt1318_of_match); + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt1318_acpi_match[] = { + { "10EC1318", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt1318_acpi_match); +#endif + +static int rt1318_parse_dt(struct rt1318_priv *rt1318, struct device *dev) +{ + device_property_read_u32(dev, "realtek,r0_l", + &rt1318->pdata.init_r0_l); + device_property_read_u32(dev, "realtek,r0_r", + &rt1318->pdata.init_r0_r); + + return 0; +} + +static void rt1318_calibration_sequence(struct rt1318_priv *rt1318) +{ + regmap_write(rt1318->regmap, RT1318_CLK1, 0x22); + regmap_write(rt1318->regmap, RT1318_PLL1_N_7_0, 0x06); + regmap_write(rt1318->regmap, RT1318_STP_TEMP_L, 0xCC); + regmap_write(rt1318->regmap, RT1318_STP_SEL_L, 0x40); + regmap_write(rt1318->regmap, RT1318_STP_SEL_R, 0x40); + regmap_write(rt1318->regmap, RT1318_SINE_GEN0, 0x20); + regmap_write(rt1318->regmap, RT1318_SPK_VOL_TH, 0x00); + regmap_write(rt1318->regmap, RT1318_FEEDBACK_PATH, 0x0B); + regmap_write(rt1318->regmap, RT1318_TCON, 0x1C); + regmap_write(rt1318->regmap, RT1318_TCON_RELATE, 0x58); + regmap_write(rt1318->regmap, RT1318_TCON_RELATE, 0x78); + regmap_write(rt1318->regmap, RT1318_STP_R0_EN_L, 0xC2); +} + +static void rt1318_r0_calculate(struct rt1318_priv *rt1318) +{ + unsigned int r0_l, r0_l_byte0, r0_l_byte1, r0_l_byte2, r0_l_byte3; + unsigned int r0_r, r0_r_byte0, r0_r_byte1, r0_r_byte2, r0_r_byte3; + unsigned int r0_l_integer, r0_l_factor, r0_r_integer, r0_r_factor; + unsigned int format = 16777216; /* 2^24 */ + + regmap_read(rt1318->regmap, RT1318_R0_L_24, &r0_l_byte0); + regmap_read(rt1318->regmap, RT1318_R0_L_23_16, &r0_l_byte1); + regmap_read(rt1318->regmap, RT1318_R0_L_15_8, &r0_l_byte2); + regmap_read(rt1318->regmap, RT1318_R0_L_7_0, &r0_l_byte3); + r0_l = r0_l_byte0 << 24 | r0_l_byte1 << 16 | r0_l_byte2 << 8 | r0_l_byte3; + r0_l_integer = format / r0_l; + r0_l_factor = (format * 10) / r0_l - r0_l_integer * 10; + + regmap_read(rt1318->regmap, RT1318_R0_R_24, &r0_r_byte0); + regmap_read(rt1318->regmap, RT1318_R0_R_23_16, &r0_r_byte1); + regmap_read(rt1318->regmap, RT1318_R0_R_15_8, &r0_r_byte2); + regmap_read(rt1318->regmap, RT1318_R0_R_7_0, &r0_r_byte3); + r0_r = r0_r_byte0 << 24 | r0_r_byte1 << 16 | r0_r_byte2 << 8 | r0_r_byte3; + r0_r_integer = format / r0_r; + r0_r_factor = (format * 10) / r0_r - r0_r_integer * 10; + + dev_dbg(rt1318->component->dev, "r0_l_ch:%d.%d ohm\n", r0_l_integer, r0_l_factor); + dev_dbg(rt1318->component->dev, "r0_r_ch:%d.%d ohm\n", r0_r_integer, r0_r_factor); +} + +static void rt1318_r0_restore(struct rt1318_priv *rt1318) +{ + regmap_write(rt1318->regmap, RT1318_PRE_R0_L_24, + (rt1318->pdata.init_r0_l >> 24) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_L_23_16, + (rt1318->pdata.init_r0_l >> 16) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_L_15_8, + (rt1318->pdata.init_r0_l >> 8) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_L_7_0, + (rt1318->pdata.init_r0_l >> 0) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_R_24, + (rt1318->pdata.init_r0_r >> 24) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_R_23_16, + (rt1318->pdata.init_r0_r >> 16) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_R_15_8, + (rt1318->pdata.init_r0_r >> 8) & 0xff); + regmap_write(rt1318->regmap, RT1318_PRE_R0_R_7_0, + (rt1318->pdata.init_r0_r >> 0) & 0xff); + regmap_write(rt1318->regmap, RT1318_STP_SEL_L, 0x80); + regmap_write(rt1318->regmap, RT1318_STP_SEL_R, 0x80); + regmap_write(rt1318->regmap, RT1318_R0_CMP_L_FLAG, 0xc0); + regmap_write(rt1318->regmap, RT1318_R0_CMP_R_FLAG, 0xc0); + regmap_write(rt1318->regmap, RT1318_STP_R0_EN_L, 0xc0); + regmap_write(rt1318->regmap, RT1318_STP_R0_EN_R, 0xc0); + regmap_write(rt1318->regmap, RT1318_STP_TEMP_L, 0xcc); + regmap_write(rt1318->regmap, RT1318_TCON, 0x9c); +} + +static int rt1318_calibrate(struct rt1318_priv *rt1318) +{ + int chk_cnt = 30, count = 0; + int val, val2; + + regmap_write(rt1318->regmap, RT1318_PWR_STA1, 0x1); + usleep_range(0, 10000); + rt1318_calibration_sequence(rt1318); + + while (count < chk_cnt) { + msleep(100); + regmap_read(rt1318->regmap, RT1318_R0_CMP_L_FLAG, &val); + regmap_read(rt1318->regmap, RT1318_R0_CMP_R_FLAG, &val2); + val = (val >> 1) & 0x1; + val2 = (val2 >> 1) & 0x1; + if (val & val2) { + dev_dbg(rt1318->component->dev, "Calibration done.\n"); + break; + } + count++; + if (count == chk_cnt) { + regmap_write(rt1318->regmap, RT1318_PWR_STA1, 0x0); + return RT1318_R0_CALIB_NOT_DONE; + } + } + regmap_write(rt1318->regmap, RT1318_PWR_STA1, 0x0); + regmap_read(rt1318->regmap, RT1318_R0_CMP_L_FLAG, &val); + regmap_read(rt1318->regmap, RT1318_R0_CMP_R_FLAG, &val2); + if ((val & 0x1) & (val2 & 0x1)) + return RT1318_R0_IN_RANGE; + else + return RT1318_R0_OUT_OF_RANGE; +} + +static void rt1318_calibration_work(struct work_struct *work) +{ + struct rt1318_priv *rt1318 = + container_of(work, struct rt1318_priv, cali_work); + int ret; + + if (rt1318->pdata.init_r0_l && rt1318->pdata.init_r0_r) + rt1318_r0_restore(rt1318); + else { + ret = rt1318_calibrate(rt1318); + if (ret == RT1318_R0_IN_RANGE) + rt1318_r0_calculate(rt1318); + dev_dbg(rt1318->component->dev, "Calibrate R0 result:%d\n", ret); + } +} + +static int rt1318_i2c_probe(struct i2c_client *i2c) +{ + struct rt1318_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt1318_priv *rt1318; + int ret, val, val2, dev_id; + + rt1318 = devm_kzalloc(&i2c->dev, sizeof(struct rt1318_priv), + GFP_KERNEL); + if (!rt1318) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1318); + + if (pdata) + rt1318->pdata = *pdata; + else + rt1318_parse_dt(rt1318, &i2c->dev); + + rt1318->regmap = devm_regmap_init_i2c(i2c, &rt1318_regmap); + if (IS_ERR(rt1318->regmap)) { + ret = PTR_ERR(rt1318->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1318->regmap, RT1318_DEV_ID1, &val); + regmap_read(rt1318->regmap, RT1318_DEV_ID2, &val2); + dev_id = (val << 8) | val2; + if (dev_id != 0x6821) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt1318\n", + dev_id); + return -ENODEV; + } + + ret = regmap_register_patch(rt1318->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + INIT_WORK(&rt1318->cali_work, rt1318_calibration_work); + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1318, rt1318_dai, ARRAY_SIZE(rt1318_dai)); +} + +static struct i2c_driver rt1318_i2c_driver = { + .driver = { + .name = "rt1318", + .of_match_table = of_match_ptr(rt1318_of_match), + .acpi_match_table = ACPI_PTR(rt1318_acpi_match), + }, + .probe = rt1318_i2c_probe, + .id_table = rt1318_i2c_id, +}; +module_i2c_driver(rt1318_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1318 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt1318.h b/sound/soc/codecs/rt1318.h new file mode 100644 index 000000000000..cec40b484216 --- /dev/null +++ b/sound/soc/codecs/rt1318.h @@ -0,0 +1,342 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1318.h -- Platform data for RT1318 + * + * Copyright 2024 Realtek Semiconductor Corp. + */ +#include <sound/rt1318.h> + +#ifndef __RT1318_H__ +#define __RT1318_H__ + +struct rt1318_priv { + struct snd_soc_component *component; + struct rt1318_platform_data pdata; + struct work_struct cali_work; + struct regmap *regmap; + + unsigned int r0_l_integer; + unsigned int r0_l_factor; + unsigned int r0_r_integer; + unsigned int r0_r_factor; + int rt1318_init; + int rt1318_dvol; + int sysclk_src; + int sysclk; + int lrck; + int bclk; + int master; + int pll_src; + int pll_in; + int pll_out; +}; + +#define RT1318_PLL_INP_MAX 40000000 +#define RT1318_PLL_INP_MIN 256000 +#define RT1318_PLL_N_MAX 0x1ff +#define RT1318_PLL_K_MAX 0x1f +#define RT1318_PLL_M_MAX 0x1f + +#define RT1318_LRCLK_192000 192000 +#define RT1318_LRCLK_96000 96000 +#define RT1318_LRCLK_48000 48000 +#define RT1318_LRCLK_44100 44100 +#define RT1318_LRCLK_16000 16000 +#define RT1318_DVOL_STEP 383 + +#define RT1318_CLK1 0xc001 +#define RT1318_CLK2 0xc003 +#define RT1318_CLK3 0xc004 +#define RT1318_CLK4 0xc005 +#define RT1318_CLK5 0xc006 +#define RT1318_CLK6 0xc007 +#define RT1318_CLK7 0xc008 +#define RT1318_PWR_STA1 0xc121 +#define RT1318_SPK_VOL_TH 0xc130 +#define RT1318_TCON 0xc203 +#define RT1318_SRC_TCON 0xc204 +#define RT1318_TCON_RELATE 0xc206 +#define RT1318_DA_VOL_L_8 0xc20b +#define RT1318_DA_VOL_L_1_7 0xc20c +#define RT1318_DA_VOL_R_8 0xc20d +#define RT1318_DA_VOL_R_1_7 0xc20e +#define RT1318_FEEDBACK_PATH 0xc321 +#define RT1318_STP_TEMP_L 0xdb00 +#define RT1318_STP_SEL_L 0xdb08 +#define RT1318_STP_R0_EN_L 0xdb12 +#define RT1318_R0_CMP_L_FLAG 0xdb35 +#define RT1318_PRE_R0_L_24 0xdbb5 +#define RT1318_PRE_R0_L_23_16 0xdbb6 +#define RT1318_PRE_R0_L_15_8 0xdbb7 +#define RT1318_PRE_R0_L_7_0 0xdbb8 +#define RT1318_R0_L_24 0xdbc5 +#define RT1318_R0_L_23_16 0xdbc6 +#define RT1318_R0_L_15_8 0xdbc7 +#define RT1318_R0_L_7_0 0xdbc8 +#define RT1318_STP_SEL_R 0xdd08 +#define RT1318_STP_R0_EN_R 0xdd12 +#define RT1318_R0_CMP_R_FLAG 0xdd35 +#define RT1318_PRE_R0_R_24 0xddb5 +#define RT1318_PRE_R0_R_23_16 0xddb6 +#define RT1318_PRE_R0_R_15_8 0xddb7 +#define RT1318_PRE_R0_R_7_0 0xddb8 +#define RT1318_R0_R_24 0xddc5 +#define RT1318_R0_R_23_16 0xddc6 +#define RT1318_R0_R_15_8 0xddc7 +#define RT1318_R0_R_7_0 0xddc8 +#define RT1318_DEV_ID1 0xf012 +#define RT1318_DEV_ID2 0xf013 +#define RT1318_PLL1_K 0xf20d +#define RT1318_PLL1_M 0xf20f +#define RT1318_PLL1_N_8 0xf211 +#define RT1318_PLL1_N_7_0 0xf212 +#define RT1318_SINE_GEN0 0xf800 +#define RT1318_TDM_CTRL1 0xf900 +#define RT1318_TDM_CTRL2 0xf901 +#define RT1318_TDM_CTRL3 0xf902 +#define RT1318_TDM_CTRL9 0xf908 + + +/* Clock-1 (0xC001) */ +#define RT1318_PLLIN_MASK (0x7 << 4) +#define RT1318_PLLIN_BCLK0 (0x0 << 4) +#define RT1318_PLLIN_BCLK1 (0x1 << 4) +#define RT1318_PLLIN_RC (0x2 << 4) +#define RT1318_PLLIN_MCLK (0x3 << 4) +#define RT1318_PLLIN_SDW1 (0x4 << 4) +#define RT1318_PLLIN_SDW2 (0x5 << 4) +#define RT1318_PLLIN_SDW3 (0x6 << 4) +#define RT1318_PLLIN_SDW4 (0x7 << 4) +#define RT1318_SYSCLK_SEL_MASK (0x7 << 0) +#define RT1318_SYSCLK_BCLK (0x0 << 0) +#define RT1318_SYSCLK_SDW (0x1 << 0) +#define RT1318_SYSCLK_PLL2F (0x2 << 0) +#define RT1318_SYSCLK_PLL2B (0x3 << 0) +#define RT1318_SYSCLK_MCLK (0x4 << 0) +#define RT1318_SYSCLK_RC1 (0x5 << 0) +#define RT1318_SYSCLK_RC2 (0x6 << 0) +#define RT1318_SYSCLK_RC3 (0x7 << 0) +/* Clock-2 (0xC003) */ +#define RT1318_DIV_AP_MASK (0x3 << 4) +#define RT1318_DIV_AP_SFT 4 +#define RT1318_DIV_AP_DIV1 (0x0 << 4) +#define RT1318_DIV_AP_DIV2 (0x1 << 4) +#define RT1318_DIV_AP_DIV4 (0x2 << 4) +#define RT1318_DIV_AP_DIV8 (0x3 << 4) +#define RT1318_DIV_DAMOD_MASK (0x3 << 0) +#define RT1318_DIV_DAMOD_SFT 0 +#define RT1318_DIV_DAMOD_DIV1 (0x0 << 0) +#define RT1318_DIV_DAMOD_DIV2 (0x1 << 0) +#define RT1318_DIV_DAMOD_DIV4 (0x2 << 0) +#define RT1318_DIV_DAMOD_DIV8 (0x3 << 0) +/* Clock-3 (0xC004) */ +#define RT1318_AD_STO1_MASK (0x7 << 4) +#define RT1318_AD_STO1_SFT 4 +#define RT1318_AD_STO1_DIV1 (0x0 << 4) +#define RT1318_AD_STO1_DIV2 (0x1 << 4) +#define RT1318_AD_STO1_DIV4 (0x2 << 4) +#define RT1318_AD_STO1_DIV8 (0x3 << 4) +#define RT1318_AD_STO1_DIV16 (0x4 << 4) +#define RT1318_AD_STO2_MASK (0x7 << 0) +#define RT1318_AD_STO2_SFT 0 +#define RT1318_AD_STO2_DIV1 (0x0 << 0) +#define RT1318_AD_STO2_DIV2 (0x1 << 0) +#define RT1318_AD_STO2_DIV4 (0x2 << 0) +#define RT1318_AD_STO2_DIV8 (0x3 << 0) +#define RT1318_AD_STO2_DIV16 (0x4 << 0) +#define RT1318_AD_STO2_SFT 0 +/* Clock-4 (0xC005) */ +#define RT1318_AD_ANA_STO1_MASK (0x7 << 4) +#define RT1318_AD_ANA_STO1_SFT 4 +#define RT1318_AD_ANA_STO1_DIV1 (0x0 << 4) +#define RT1318_AD_ANA_STO1_DIV2 (0x1 << 4) +#define RT1318_AD_ANA_STO1_DIV4 (0x2 << 4) +#define RT1318_AD_ANA_STO1_DIV8 (0x3 << 4) +#define RT1318_AD_ANA_STO1_DIV16 (0x4 << 4) +#define RT1318_AD_ANA_STO2_MASK (0x7 << 0) +#define RT1318_AD_ANA_STO2_DIV1 (0x0 << 0) +#define RT1318_AD_ANA_STO2_DIV2 (0x1 << 0) +#define RT1318_AD_ANA_STO2_DIV4 (0x2 << 0) +#define RT1318_AD_ANA_STO2_DIV8 (0x3 << 0) +#define RT1318_AD_ANA_STO2_DIV16 (0x4 << 0) +#define RT1318_AD_ANA_STO2_SFT 0 +/* Clock-5 (0xC006) */ +#define RT1318_DIV_FIFO_IN_MASK (0x3 << 4) +#define RT1318_DIV_FIFO_IN_SFT 4 +#define RT1318_DIV_FIFO_IN_DIV1 (0x0 << 4) +#define RT1318_DIV_FIFO_IN_DIV2 (0x1 << 4) +#define RT1318_DIV_FIFO_IN_DIV4 (0x2 << 4) +#define RT1318_DIV_FIFO_IN_DIV8 (0x3 << 4) +#define RT1318_DIV_FIFO_OUT_MASK (0x3 << 0) +#define RT1318_DIV_FIFO_OUT_DIV1 (0x0 << 0) +#define RT1318_DIV_FIFO_OUT_DIV2 (0x1 << 0) +#define RT1318_DIV_FIFO_OUT_DIV4 (0x2 << 0) +#define RT1318_DIV_FIFO_OUT_DIV8 (0x3 << 0) +#define RT1318_DIV_FIFO_OUT_SFT 0 +/* Clock-6 (0xC007) */ +#define RT1318_DIV_NLMS_MASK (0x3 << 6) +#define RT1318_DIV_NLMS_SFT 6 +#define RT1318_DIV_NLMS_DIV1 (0x0 << 6) +#define RT1318_DIV_NLMS_DIV2 (0x1 << 6) +#define RT1318_DIV_NLMS_DIV4 (0x2 << 6) +#define RT1318_DIV_NLMS_DIV8 (0x3 << 6) +#define RT1318_DIV_AD_MONO_MASK (0x7 << 3) +#define RT1318_DIV_AD_MONO_SFT 3 +#define RT1318_DIV_AD_MONO_DIV1 (0x0 << 3) +#define RT1318_DIV_AD_MONO_DIV2 (0x1 << 3) +#define RT1318_DIV_AD_MONO_DIV4 (0x2 << 3) +#define RT1318_DIV_AD_MONO_DIV8 (0x3 << 3) +#define RT1318_DIV_AD_MONO_DIV16 (0x4 << 3) +#define RT1318_DIV_POST_G_MASK (0x7 << 0) +#define RT1318_DIV_POST_G_SFT 0 +#define RT1318_DIV_POST_G_DIV1 (0x0 << 0) +#define RT1318_DIV_POST_G_DIV2 (0x1 << 0) +#define RT1318_DIV_POST_G_DIV4 (0x2 << 0) +#define RT1318_DIV_POST_G_DIV8 (0x3 << 0) +#define RT1318_DIV_POST_G_DIV16 (0x4 << 0) +/* Power Status 1 (0xC121) */ +#define RT1318_PDB_CTRL_MASK (0x1) +#define RT1318_PDB_CTRL_LOW (0x0) +#define RT1318_PDB_CTRL_HIGH (0x1) +#define RT1318_PDB_CTRL_SFT 0 +/* SRC Tcon(0xc204) */ +#define RT1318_SRCIN_IN_SEL_MASK (0x3 << 6) +#define RT1318_SRCIN_IN_48K (0x0 << 6) +#define RT1318_SRCIN_IN_44P1 (0x1 << 6) +#define RT1318_SRCIN_IN_32K (0x2 << 6) +#define RT1318_SRCIN_IN_16K (0x3 << 6) +#define RT1318_SRCIN_F12288_MASK (0x3 << 4) +#define RT1318_SRCIN_TCON1 (0x0 << 4) +#define RT1318_SRCIN_TCON2 (0x1 << 4) +#define RT1318_SRCIN_TCON4 (0x2 << 4) +#define RT1318_SRCIN_TCON8 (0x3 << 4) +#define RT1318_SRCIN_DACLK_MASK (0x3 << 2) +#define RT1318_DACLK_TCON1 (0x0 << 2) +#define RT1318_DACLK_TCON2 (0x1 << 2) +#define RT1318_DACLK_TCON4 (0x2 << 2) +#define RT1318_DACLK_TCON8 (0x3 << 2) +/* R0 Compare Flag (0xDB35) */ +#define RT1318_R0_RANGE_MASK (0x1) +#define RT1318_R0_OUTOFRANGE (0x0) +#define RT1318_R0_INRANGE (0x1) +/* PLL internal setting (0xF20D), K value */ +#define RT1318_K_PLL1_MASK (0x1f << 0) +/* PLL internal setting (0xF20F), M value */ +#define RT1318_M_PLL1_MASK (0x1f << 0) +/* PLL internal setting (0xF211), N_8 value */ +#define RT1318_N_8_PLL1_MASK (0x1 << 0) +/* PLL internal setting (0xF212), N_7_0 value */ +#define RT1318_N_7_0_PLL1_MASK (0xff << 0) +/* TDM CTRL 1 (0xf900) */ +#define RT1318_TDM_BCLK_MASK (0x1 << 7) +#define RT1318_TDM_BCLK_NORM (0x0 << 7) +#define RT1318_TDM_BCLK_INV (0x1 << 7) +#define RT1318_I2S_FMT_MASK (0x7 << 0) +#define RT1318_FMT_I2S (0x0 << 0) +#define RT1318_FMT_LEFT_J (0x1 << 0) +#define RT1318_FMT_PCM_A_R (0x2 << 0) +#define RT1318_FMT_PCM_B_R (0x3 << 0) +#define RT1318_FMT_PCM_A_F (0x6 << 0) +#define RT1318_FMT_PCM_B_F (0x7 << 0) +#define RT1318_I2S_FMT_SFT 0 +/* TDM CTRL 2 (0xf901) */ +#define RT1318_I2S_CH_TX_MASK (0x3 << 6) +#define RT1318_I2S_CH_TX_2CH (0x0 << 6) +#define RT1318_I2S_CH_TX_4CH (0x1 << 6) +#define RT1318_I2S_CH_TX_6CH (0x2 << 6) +#define RT1318_I2S_CH_TX_8CH (0x3 << 6) +#define RT1318_I2S_CH_RX_MASK (0x3 << 4) +#define RT1318_I2S_CH_RX_2CH (0x0 << 4) +#define RT1318_I2S_CH_RX_4CH (0x1 << 4) +#define RT1318_I2S_CH_RX_6CH (0x2 << 4) +#define RT1318_I2S_CH_RX_8CH (0x3 << 4) +#define RT1318_I2S_DL_MASK 0x7 +#define RT1318_I2S_DL_SFT 0 +#define RT1318_I2S_DL_16 0x0 +#define RT1318_I2S_DL_20 0x1 +#define RT1318_I2S_DL_24 0x2 +#define RT1318_I2S_DL_32 0x3 +#define RT1318_I2S_DL_8 0x4 +/* TDM CTRL 3 (0xf902) */ +#define RT1318_I2S_TX_CHL_MASK (0x7 << 4) +#define RT1318_I2S_TX_CHL_SFT 4 +#define RT1318_I2S_TX_CHL_16 (0x0 << 4) +#define RT1318_I2S_TX_CHL_20 (0x1 << 4) +#define RT1318_I2S_TX_CHL_24 (0x2 << 4) +#define RT1318_I2S_TX_CHL_32 (0x3 << 4) +#define RT1318_I2S_TX_CHL_8 (0x4 << 4) +#define RT1318_I2S_RX_CHL_MASK (0x7 << 0) +#define RT1318_I2S_RX_CHL_SFT 0 +#define RT1318_I2S_RX_CHL_16 (0x0 << 0) +#define RT1318_I2S_RX_CHL_20 (0x1 << 0) +#define RT1318_I2S_RX_CHL_24 (0x2 << 0) +#define RT1318_I2S_RX_CHL_32 (0x3 << 0) +#define RT1318_I2S_RX_CHL_8 (0x4 << 0) +/* TDM CTRL 9 (0xf908) */ +#define RT1318_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 4) +#define RT1318_TDM_I2S_TX_R_DAC1_1_MASK 0x7 +#define RT1318_TDM_I2S_TX_L_DAC1_1_SFT 4 +#define RT1318_TDM_I2S_TX_R_DAC1_1_SFT 0 + +#define RT1318_REG_DISP_LEN 23 + +/* System Clock Source */ +enum { + RT1318_SCLK_S_BCLK, + RT1318_SCLK_S_SDW, + RT1318_SCLK_S_PLL2F, + RT1318_SCLK_S_PLL2B, + RT1318_SCLK_S_MCLK, + RT1318_SCLK_S_RC0, + RT1318_SCLK_S_RC1, + RT1318_SCLK_S_RC2, +}; + +/* PLL Source */ +enum { + RT1318_PLL_S_BCLK0, + RT1318_PLL_S_BCLK1, + RT1318_PLL_S_RC, + RT1318_PLL_S_MCLK, + RT1318_PLL_S_SDW_IN_PLL, + RT1318_PLL_S_SDW_0, + RT1318_PLL_S_SDW_1, + RT1318_PLL_S_SDW_2, +}; + +/* TDM channel */ +enum { + RT1318_2CH, + RT1318_4CH, + RT1318_6CH, + RT1318_8CH, +}; + +/* R0 calibration result */ +enum { + RT1318_R0_OUT_OF_RANGE, + RT1318_R0_IN_RANGE, + RT1318_R0_CALIB_NOT_DONE, +}; + +/* PLL pre-defined M/N/K */ + +struct pll_calc_map { + unsigned int pll_in; + unsigned int pll_out; + int k; + int n; + int m; + bool m_bp; + bool k_bp; +}; + +struct rt1318_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + bool k_bp; /* Indicates bypass k code or not. */ + int m_code; + int n_code; + int k_code; +}; + +#endif /* __RT1318_H__ */ -- 2.34.1

3 2

[PATCH v2] ASoc: tas2781: Add new Kontrol to set tas2563 digital Volume
by Shenghao Ding 12 Jul '24

12 Jul '24

Requriment from customer to add new kcontrol to set tas2563 digital Volume Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Volume controls should end in Volume - Write this as a switch statement for extension - Correct return value of tas2563_digital_gain_put - Submit tas2563 digital Volume as an independant patch --- include/sound/tas2781-tlv.h | 262 ++++++++++++++++++++++++++++++++- include/sound/tas2781.h | 1 + sound/soc/codecs/tas2781-i2c.c | 129 ++++++++++++++-- 3 files changed, 382 insertions(+), 10 deletions(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 1dc59005d241..99c41bfc7827 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -17,5 +17,265 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; #endif diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index a43ad6dcb7c7..18161d02a96f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -50,6 +50,7 @@ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) /* Volume control */ +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index d5d95ae57c73..e79d613745b4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -31,6 +31,7 @@ #include <sound/tas2781.h> #include <sound/tlv.h> #include <sound/tas2781-tlv.h> +#include <asm/unaligned.h> static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, @@ -140,6 +141,101 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tas2563_digital_gain_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + unsigned int l = 0, r = mc->max; + unsigned int target, ar_mid, mid, ar_l, ar_r; + unsigned int reg = mc->reg; + unsigned char data[4]; + int ret; + + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + goto out; + } + + target = get_unaligned_be32(&data[0]); + + while (r > 1 + l) { + mid = (l + r) / 2; + ar_mid = get_unaligned_be32(tas2563_dvc_table[mid]); + if (target < ar_mid) + r = mid; + else + l = mid; + } + + ar_l = get_unaligned_be32(tas2563_dvc_table[l]); + ar_r = get_unaligned_be32(tas2563_dvc_table[r]); + + /* find out the member same as or closer to the current volume */ + ucontrol->value.integer.value[0] = + abs(target - ar_l) <= abs(target - ar_r) ? l : r; +out: + mutex_unlock(&tas_dev->codec_lock); + return 0; +} + +static int tas2563_digital_gain_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(codec); + int vol = ucontrol->value.integer.value[0]; + int status = 0, max = mc->max, rc = 1; + int i, ret; + unsigned int reg = mc->reg; + unsigned int volrd, volwr; + unsigned char data[4]; + + vol = clamp(vol, 0, max); + mutex_lock(&tas_dev->codec_lock); + /* Read the primary device */ + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + if (ret) { + dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); + rc = -1; + goto out; + } + + volrd = get_unaligned_be32(&data[0]); + volwr = get_unaligned_be32(tas2563_dvc_table[vol]); + + if (volrd == volwr) { + rc = 0; + goto out; + } + + for (i = 0; i < tas_dev->ndev; i++) { + ret = tasdevice_dev_bulk_write(tas_dev, i, reg, + (unsigned char *)tas2563_dvc_table[vol], 4); + if (ret) { + dev_err(tas_dev->dev, + "%s, set digital vol error in dev %d\n", + __func__, i); + status |= BIT(i); + } + } + + if (status) + rc = -1; +out: + mutex_unlock(&tas_dev->codec_lock); + return rc; +} + static const struct snd_kcontrol_new tasdevice_snd_controls[] = { SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, tasdev_force_fwload_get, tasdev_force_fwload_put), @@ -154,6 +250,13 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, + 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, + tas2563_digital_gain_get, tas2563_digital_gain_put, + tas2563_dvc_tlv), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -593,17 +696,25 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct snd_kcontrol_new *p; + unsigned int size; int rc; - if (tas_priv->chip_id == TAS2781) { - rc = snd_soc_add_component_controls(codec, - tas2781_snd_controls, - ARRAY_SIZE(tas2781_snd_controls)); - if (rc < 0) { - dev_err(tas_priv->dev, "%s: Add control err rc = %d", - __func__, rc); - return rc; - } + switch (tas_priv->chip_id) { + case TAS2781: + p = (struct snd_kcontrol_new *)tas2781_snd_controls; + size = ARRAY_SIZE(tas2781_snd_controls); + break; + default: + p = (struct snd_kcontrol_new *)tas2563_snd_controls; + size = ARRAY_SIZE(tas2563_snd_controls); + } + + rc = snd_soc_add_component_controls(codec, p, size); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; } tas_priv->name_prefix = codec->name_prefix; -- 2.34.1

2 1

[PATCH] ASoC: dt-bindings: cirrus,cs42xx8: Convert to dtschema
by Animesh Agarwal 12 Jul '24

12 Jul '24

Convert the Cirrus Logic CS42448/CS42888 audio CODEC bindings to DT schema format. Set power supply properties to required only for CS42888. Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- .../bindings/sound/cirrus,cs42xx8.yaml | 81 +++++++++++++++++++ .../devicetree/bindings/sound/cs42xx8.txt | 34 -------- 2 files changed, 81 insertions(+), 34 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml new file mode 100644 index 000000000000..725b47e82062 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs42xx8.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs42xx8.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS42448/CS42888 audio CODEC + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + compatible: + enum: + - cirrus,cs42448 + - cirrus,cs42888 + + reg: + maxItems: 1 + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + const: mclk + + VA-supply: + description: Analog power supply. + + VD-supply: + description: Digital power supply. + + VLC-supply: + description: Control port power supply + + VLS-supply: + description: Serial port interface power supply. + + reset-gpios: + description: This pin is connected to the chip's RESET pin. + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + +if: + properties: + compatible: + contains: + const: cirrus,cs42888 +then: + required: + - VA-supply + - VD-supply + - VLC-supply + - VLS-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <&reg_audio>; + VD-supply = <&reg_audio>; + VLS-supply = <&reg_audio>; + VLC-supply = <&reg_audio>; + reset-gpios = <&gpio 1>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt deleted file mode 100644 index bbfe39347c20..000000000000 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ /dev/null @@ -1,34 +0,0 @@ -CS42448/CS42888 audio CODEC - -Required properties: - - - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" - - - reg : the I2C address of the device for I2C - - - clocks : a list of phandles + clock-specifiers, one for each entry in - clock-names - - - clock-names : must contain "mclk" - - - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, - as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Optional properties: - - - reset-gpios : a GPIO spec to define which pin is connected to the chip's - !RESET pin - -Example: - -cs42888: codec@48 { - compatible = "cirrus,cs42888"; - reg = <0x48>; - clocks = <&codec_mclk 0>; - clock-names = "mclk"; - VA-supply = <&reg_audio>; - VD-supply = <&reg_audio>; - VLS-supply = <&reg_audio>; - VLC-supply = <&reg_audio>; - reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; -}; -- 2.45.2

1 0

[PATCH v2 00/10] Add support for non-interleaved mode in qmc_audio
by Herve Codina 12 Jul '24

12 Jul '24

The qmc_audio driver supports only audio in interleaved mode. Non-interleaved mode can be easily supported using several QMC channel per DAI. In that case, data related to ch0 are sent to (received from) the first QMC channel, data related to ch1 use the next QMC channel and so on up to the last channel. In terms of constraints and settings, the interleaved and non-interleaved modes are slightly different. In interleaved mode: - The sample size should fit in the number of time-slots available for the QMC channel. - The number of audio channels should fit in the number of time-slots (taking into account the sample size) available for the QMC channel. In non-interleaved mode: - The number of audio channels is the number of available QMC channels. - Each QMC channel should have the same number of time-slots. - The sample size equals the number of time-slots of one QMC channel. This series add support for the non-interleaved mode in the qmc_audio driver and is composed of the following parts: - Patches 1 and 2: Fix some issues in the qmc_audio - Patches 3 to 6: Prepare qmc_audio for the non-interleaved mode - Patches 7 and 8: Extend the QMC driver API - Patches 9 and 10: The support for non-interleaved mode itself Compared to the previous iteration, this v2 series mainly improves qmc_audio_access_is_interleaved(). Best regards, Hervé Link to v1: https://lore.kernel.org/lkml/20240620084300.397853-1-herve.codina@bootlin.c… Changes v1 -> v2 - Patches 1 to 8 No changes - Patch 9 Add 'Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org>' - Patch 10 Remove unneeded ';' Modify qmc_audio_access_is_interleaved() Herve Codina (10): ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned value ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handling ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion routines ASoC: fsl: fsl_qmc_audio: Introduce qmc_audio_pcm_{read,write}_submit() ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved() soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle list soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles() dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAI ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode. .../bindings/sound/fsl,qmc-audio.yaml | 41 +- drivers/soc/fsl/qe/qmc.c | 32 +- include/soc/fsl/qe/qmc.h | 27 +- sound/soc/fsl/fsl_qmc_audio.c | 591 +++++++++++++----- 4 files changed, 506 insertions(+), 185 deletions(-) -- 2.45.0

4 14

[PATCH] ASoC: codecs: wsa884x: Implement temperature reading and hwmon
by Krzysztof Kozlowski 12 Jul '24

12 Jul '24

Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/wsa884x.c | 197 +++++++++++++++++++++++++++++++++++++ 1 file changed, 197 insertions(+) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 7b19df9c1728..5f4acd66f942 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include <linux/bitfield.h> +#include <linux/cleanup.h> #include <linux/device.h> #include <linux/gpio/consumer.h> +#include <linux/hwmon.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1650,6 +1691,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1661,6 +1706,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1800,6 +1849,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1849,6 +2036,7 @@ static int wsa884x_probe(struct sdw_slave *pdev, { struct device *dev = &pdev->dev; struct wsa884x_priv *wsa884x; + struct device *hwmon; unsigned int i; int ret; @@ -1856,6 +2044,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1913,6 +2103,13 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); -- 2.43.0

1 0

[PATCH v9] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by baojun.xu＠ti.com 12 Jul '24

12 Jul '24

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

3 2

[GIT PULL] ASoC fixes for v6.10-rc7
by Mark Brown 11 Jul '24

11 Jul '24

The following changes since commit 68f97fe330e01450ace63da0ce5cab676fc97f9a: ASoC: rt5645: fix issue of random interrupt from push-button (2024-06-26 16:34:48 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.10-rc7 for you to fetch changes up to 680e126ec0400f6daecf0510c5bb97a55779ff03: firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files (2024-07-08 15:55:11 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.10 A few fairly small fixes for ASoC, there's a relatively large set of hardening changes for the cs_dsp firmware file parsing and a couple of other small device specific fixes. ---------------------------------------------------------------- Bard Liao (1): ASoC: rt711-sdw: add missing readable registers Kai Vehmanen (1): ASoC: SOF: Intel: hda: fix null deref on system suspend entry Peter Ujfalusi (1): ASoC: SOF: Intel: hda-pcm: Limit the maximum number of periods by MAX_BDL_ENTRIES Richard Fitzgerald (5): firmware: cs_dsp: Fix overflow checking of wmfw header firmware: cs_dsp: Return error if block header overflows file firmware: cs_dsp: Validate payload length before processing block firmware: cs_dsp: Prevent buffer overrun when processing V2 alg headers firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files drivers/firmware/cirrus/cs_dsp.c | 227 ++++++++++++++++++++++++++++----------- sound/soc/codecs/rt711-sdw.c | 2 + sound/soc/sof/intel/hda-dai.c | 12 +-- sound/soc/sof/intel/hda-pcm.c | 6 ++ 4 files changed, 176 insertions(+), 71 deletions(-)

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Re: [syzbot] [mm?] INFO: rcu detected stall in sys_wait4 (4)
by syzbot 11 Jul '24

11 Jul '24

syzbot suspects this issue was fixed by commit: commit fb66df20a7201e60f2b13d7f95d031b31a8831d3 Author: Vladimir Oltean <vladimir.oltean(a)nxp.com> Date: Mon May 27 15:39:55 2024 +0000 net/sched: taprio: extend minimum interval restriction to entire cycle too bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=10593441980000 start commit: fe46a7dd189e Merge tag 'sound-6.9-rc1' of git://git.kernel.. git tree: upstream kernel config: https://syzkaller.appspot.com/x/.config?x=fe78468a74fdc3b7 dashboard link: https://syzkaller.appspot.com/bug?extid=6969434de600a6ba9f07 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1091a5f6180000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=17a22c13180000 If the result looks correct, please mark the issue as fixed by replying with: #syz fix: net/sched: taprio: extend minimum interval restriction to entire cycle too For information about bisection process see: https://goo.gl/tpsmEJ#bisection

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[PATCH v8] ALSA: hda/tas2781: Add tas2781 hda SPI driver
by Baojun Xu 11 Jul '24

11 Jul '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. The code support Realtek as the primary codec. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v8: - Move *-y position in sound/pci/hda/Makefile. - Change + to | for bit operation in macro define. - Change TASDEVICE_Checksum to TASDEVICE_CHECKSUM. v7: - Add new project Gemtree (0x103c8daa) support for tas2781 spi driver. v6: - Remove client in tasdevice_priv struct as it was not used. - Change format for tasdevice_spi_dev_write() in tas2781_hda_spi.c. - Change update bits function to read & write mode, as tas2781 use last bit to mask read & write, it cause regmap_update_bits work not as our expected. - Add element in tas2781_snd_controls, tas2781_prof_ctrl, tas2781_dsp_conf_ctrl, to support second device. Previous version add sound card for first index only. - Change calibration registers address declaire to TASDEVICE_REG() mode. - Add snd_ctl_new1() for second device. - Change format for tas2781_hda_unbind() to fit 79 chars in one line. - Remove client for spi as it was not used. - Remove unused variables. - Remove cur_prog clean in runtime_suspend(). - Remove unused variables. - Add tasdevice_spi_tuning_switch() in runtime_resume. - Remove cur_prog clean in system_resume(). - Add tuning switch if playback started in system_resume(). - Change TASDEVICE_PAGE_REG to TASDEVICE_REG_ID in tas2781_spi_fwlib.c. - Remove calibration binary load functions as data will be loaded from EFI. v5: - Combined all modification into one patch for pass the compile. - Remove old hardware id "TIAS2781" as no production on market. - Change max page size to 256 in tas2781-spi.h. - Change book reg from 127 to GENMASK(7, 1), as one bit shift is needed. - Change register address define as one bit shift needed for offset. - Change block check sum comments to Firmware block from I2C. - Change define of (page + reg) to TASDEVICE_PAGE_REG from TASDEVICE_PGRG. - Change to lower case in defined value. - Add registers default value table in tas2781_hda_spi.c - Change rang_max to 256 * 256. - Add zero_flag_mask in regmap_config. - Change max_register to 256 * 256. - Add registers default table into structure regmap_config. - Change parameter from book number to whole register address. - Remove page 0 switch as SPI driver will do it. - Add bit(0) for read function as tas2781 will check last bit of address, to determine operation mode is read (1) or write (0). - Change switch case to if check in tas2781_hda_playback_hook(), as it may cause compile problem in clang-19 building. - Change variable declaration position. - Change order in if check. - Remove old hardware id ("TIAS2781") support. - Remove debug message in runtime suspend and resume. - Remove regmap.h include in tas2781_spi_fwlib.c. - Remove MODULE_ in fwlib as this file will work with tas2781_hda_spi.c only. v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v4: - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs - Move include of sound/tas2781-dsp.h to c source file tas2781_hda_spi.c, tas2781_spi_fwlib.c, as it's not needed in header file tas2781-spi.h - Use page and book number define in other micro define - Change 4000000 to (4 * HZ_PER_MHZ) - Add define for calibration variable name and size in UEFI. - Remove structure tasdevice & tasdevice_irqinfo, and move variables in to structure tasdevice_priv - Remove some unused variables in sttructure tasdevice_priv - Remove function declare of tascodec_init(), use tascodec_spi_init() - Remove function declare of tasdevice_remove() - Add array_size.h included in tas2781_hda_spi.c - Add cleanup.h included for change mutex to guard - Add units.h included - Include tas2781-dsp.h - Remove unused variables in tas2781_hda structure - Change 0xff to GENMASK(7, 0) - Change 128 to define of max page size - Change code as all variables was moved to tasdevice_priv structure - Change comments for error in book write, or page write - Remove initial of ret, and return 0 directly - Change comments for wrong typo, add space in front and end - Add check for bulk read, max size should be a page size - Change usleep_rang() to fsleep() - Change mutex_lock to guard(mutex), and remove mutex_unlock() - Change tasdevice_spi_init() to void from int as no value return required - Change second parameter to sizeof - Remove tasdevice_clamp(), use clamp() - Add compatible with old hardware id "TIAS2781" - Remove cs_gpio and spi device in tas2781_read_acpi() as use default CS - Change dev-index check, return -EINVAL if it's <= 0 - Change 0xFF to U8_MAX - Remove GPIO get for chip-select, use default from SPI device - Perform RESET in first AMP only as all AMPs share same RESET pin - return 0 directly in some simple routine to avoid variable initial - Change comments for function which was used by system - Change "ON" "OFF" to function str_on_off(...) - Change format of GUID - Add temp buffer for first efi.get_variable() - Free data if it was allocated - Change format of debug message of calibration date time print - Remove total_sz = 0, as it's not needed - Move save_calibration to after tuning_switch to avoid calibration data overwrite - Remove dev from structure tasdevice_hda, use dev in tasdevice_priv - Remove 0 from { "tas2781-hda", 0}, & {"TXNW2781", 0}, - Add old hardware id "TIAS2781" for compatible with old production - Add 2 devices in struct smi_node tas2781_hda, to compatible with 4 AMPs v3: - Move HID up to above /* Non-conforming _HID ... */ in scan.c, for avoid misunderstanding. - Move HID up to above /* Non-conforming _HID ... */ in serial-multi-instantiate.c, for avoid misunderstanding. - Change objs to y for snd-hda-scodec-tas2781-spi- in Makefile. - Change format for some define in tas2781-spi.h. - Add define for TASDEVICE_MAX_BOOK_NUM, TASDEVICE_MAX_PAGE. - Move declare of "enum calib_data {" to header file from source file. - Remove "enum device_catlog_id {" as only one platform was supported now. - Remove "struct calidata", as we will save pure calibration data only. - Remove "struct calidata", "enum device_catlog_id" in tasdevice_priv. - Add calibration_data in tasdevice_priv to save pure calibration data. - Remove declare of tasdevice_save_calibration() and tasdevice_apply_calibration(), as it will be used within same c file. - Add below header files included in tas2781_hda_spi.c: bits.h, mutex.h, property.h, time.h, types.h. - Move two micro define to header file tas2781-spi.h. - Change format of some micro define. - Change format of some structure. - Remove irq in tas2781_hda, as have it already in tasdevice_priv. - Remove some local functions declare as not necessarily. - Return error if regmap_write() failed. - Change fixed value 2 to sizeof(data). - Remove all of EXPORT_SYMBOL_GPL, as all of function was called within same module. - Remove empty line after last return in some functions. - Remove some variable initialyzation. - Remove variable sub, store acpi_subsystem_id directly. - Remove wrong comments for device index, it's must, can't be NULL. - Remove some local variables, use elements in structure directly. - Append commas in last element of an array. - Change calibration data process, didn't save all of calibration data in EFI, get in local, and save pure calibration data in tasdevice_priv. - Call tasdevice_save_calibration() from function pointer in tasdevice_priv; - Remove subid as one platform was supported only, needn't check. - Add initialyzation for local variable. - Move regmap initialyzation to before acpi read. - Call tasdevice_apply_calibration() from function pointer in tasdevice_priv; - Remove MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB), as all functions was performed within same module. - Update format and variables declare order. - Change format of multi conditions for if. - Remove casting which is not needed. - Change variables type to avoid casting. - Remove Unneeded parentheses. - Change if check to avoid goto. - Add {} for keep same style. - Remove some local variables, use elements in structure directly. v2: - Change depends on SPI to SPI_MASTER in sound/pci/hda/Kconfig - Add select SND_HDA_SCODEC_COMPONENT in sound/pci/hda/Kconfig - Change comp_generic_fixup() 5th parameter from "-%s:00" to "-%s:00-tas2781-hda.%d" in sound/pci/hda/patch_realtek.c - Change chain_id from ALC285_FIXUP_THINKPAD_HEADSET_JACK to ALC285_FIXUP_HP_GPIO_LED in sound/pci/hda/patch_realtek.c - Remove project name "Gemtree" - Update information in file header of tas2781-spi.h. - Remove useless micro define. - Change TASDEVICE_I2CChecksum to TASDEVICE_Checksum - Remove enum control_bus as current code support SPI only. - Remove device define as current code support TAS2781 only. - Remove spi_device **spi_devs in structure tasdevice_priv. - Remove cal_binaryname in structure tasdevice_priv. - Remove ndev in structure tasdevice_priv. - Change isacpi and isspi, replace by control_bus(I2C or SPI). - Remove void tasdevice_spi_dsp_remove(void *context). - Add acpi_device and irq in structure tas2781_hda in tas2781_hda_spi.c. - Remove parameter chn in all of registers access functions. - Add tascodec_spi_init(). - Add chip select gpio set for SPI device. - Change tasdevice_tuning_switch() to tasdevice_spi_tuning_switch(). - Change device offset from tas_priv->ndev to tas_priv->index. - Change tasdevice_dsp_remove() to tasdevice_spi_dsp_remove(). - tasdevice_prmg_load() to tasdevice_spi_prmg_load(). - Change tasdevice_config_info_remove() to tasdevice_spi_config_info_remove(). - Add one dummy byte shift when read from page 2~127, or not book 0. - Change tasdevice_spi_change_chn_book() to tasdevice_spi_switch_book() - Perform put_device(physdev) before error return. - Change calibration data id to 2781 from 2783 in EFI - Change calibration data structure name in comments. - Remove file name in tas2781_spi_fwlib.c. - Remove #include <linux/i2c.h> as it not needed in SPI mode. - Change TAB to speace in micro define in tas2781_spi_fwlib.c. - Change to up-case for DSP. - Change all of multi-line comments format, empty first line. - Change all of sizeof(struct xx) to sizeof(*xx). - Remove Explicit casting for variables compare to avoid possible fault. - Return directly without goto. - Change 1 << xx to BIT(xx). - Remove deviceNumber[] as current SPI device will support one device only. - Add memory free before return error. - Change "buf[offset]; offset += 1" to buf[offset++]. - Remove some debug information print. - Change firmware binary to single device mode. - Change all bexx_to_cpup() to get_unaligned_bexx(). - Remove ndev process as current SPI device support single device only. - Remove chn and chnend for single device support only. - Change all of registers read/write function, remove parameter chn. - Create new function for single write, burst write, delay, field write in tasdevice_process_block(). --- drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 14 + sound/pci/hda/tas2781-spi.h | 148 ++ sound/pci/hda/tas2781_hda_spi.c | 1266 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2101 +++++++++++++++++ 8 files changed, 3558 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c diff --git a/drivers/acpi/scan.c b/drivers/acpi/scan.c index 503773707e01..9c08c5886370 100644 --- a/drivers/acpi/scan.c +++ b/drivers/acpi/scan.c @@ -1760,6 +1760,7 @@ static bool acpi_device_enumeration_by_parent(struct acpi_device *device) {"CSC3557", }, {"INT33FE", }, {"INT3515", }, + {"TXNW2781", }, /* Non-conforming _HID for Cirrus Logic already released */ {"CLSA0100", }, {"CLSA0101", }, diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 97b9c6392230..23d46063423d 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -368,6 +368,17 @@ static const struct smi_node cs35l57_hda = { .bus_type = SMI_AUTO_DETECT, }; +static const struct smi_node tas2781_hda = { + .instances = { + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + {} + }, + .bus_type = SMI_AUTO_DETECT, +}; + /* * Note new device-ids must also be added to ignore_serial_bus_ids in * drivers/acpi/scan.c: acpi_device_enumeration_by_parent(). @@ -380,6 +391,7 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "CSC3556", (unsigned long)&cs35l56_hda }, { "CSC3557", (unsigned long)&cs35l57_hda }, { "INT3515", (unsigned long)&int3515_data }, + { "TXNW2781", (unsigned long)&tas2781_hda }, /* Non-conforming _HID for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, { "CLSA0101", (unsigned long)&cs35l41_hda }, diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 0da625533afc..113161618788 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -204,6 +204,20 @@ config SND_HDA_SCODEC_TAS2781_I2C comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m +config SND_HDA_SCODEC_TAS2781_SPI + tristate "Build TAS2781 HD-audio side codec support for SPI Bus" + depends on SPI_MASTER + depends on ACPI + depends on EFI + depends on SND_SOC + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 SPI HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_SPI=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 058ca0a289e4..39bb30803b24 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -39,6 +39,7 @@ snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o snd-hda-scodec-component-y := hda_component.o snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o +snd-hda-scodec-tas2781-spi-y := tas2781_hda_spi.o tas2781_spi_fwlib.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -70,6 +71,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L56_SPI) += snd-hda-scodec-cs35l56-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o obj-$(CONFIG_SND_HDA_SCODEC_COMPONENT) += snd-hda-scodec-component.o obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_SPI) += snd-hda-scodec-tas2781-spi.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa76d1c88589..38f2cf148477 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6921,6 +6921,11 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); } +static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7498,6 +7503,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC245_FIXUP_TAS2781_SPI_2, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, @@ -9688,6 +9694,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC245_FIXUP_TAS2781_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_spi, + .chained = true, + .chain_id = ALC285_FIXUP_HP_GPIO_LED, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -10156,6 +10168,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b93, "HP Varcolac", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), @@ -10212,6 +10225,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8daa, "HP Gemtree", ALC245_FIXUP_TAS2781_SPI_2), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), diff --git a/sound/pci/hda/tas2781-spi.h b/sound/pci/hda/tas2781-spi.h new file mode 100644 index 000000000000..dd9d620d3bee --- /dev/null +++ b/sound/pci/hda/tas2781-spi.h @@ -0,0 +1,148 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for TAS2781 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// + +#ifndef __TAS2781_SPI_H__ +#define __TAS2781_SPI_H__ + +#define TASDEVICE_RATES \ + (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_88200) + +#define TASDEVICE_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define TASDEVICE_MAX_BOOK_NUM 256 +#define TASDEVICE_MAX_PAGE 256 + +#define TASDEVICE_MAX_SIZE (TASDEVICE_MAX_BOOK_NUM * TASDEVICE_MAX_PAGE) + +/* PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG GENMASK(7, 1) +#define TASDEVICE_BOOK_ID(reg) ((reg & GENMASK(24, 16)) >> 16) +#define TASDEVICE_PAGE_ID(reg) ((reg & GENMASK(15, 8)) >> 8) +#define TASDEVICE_REG_ID(reg) ((reg & GENMASK(7, 1)) >> 1) +#define TASDEVICE_PAGE_REG(reg) (reg & GENMASK(15, 1)) +#define TASDEVICE_REG(book, page, reg) (((book & GENMASK(7, 0)) << 16) | \ + ((page & GENMASK(7, 0)) << 8) | \ + ((reg & GENMASK(6, 0)) << 1)) + +/* Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* Firmware block Checksum */ +#define TASDEVICE_CHECKSUM TASDEVICE_REG(0x0, 0x0, 0x7e) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +#define TAS2781_SPI_MAX_FREQ (4 * HZ_PER_MHZ) + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" +#define TASDEVICE_CALIBRATION_DATA_SIZE 256 + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +struct tasdevice_priv { + struct tasdevice_fw *cali_data_fmw; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct gpio_desc *reset; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + + const char *acpi_subsystem_id; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + + unsigned int err_code; + bool is_loading; + bool is_loaderr; + int cur_book; + int cur_prog; + int cur_conf; + int fw_state; + int index; + int irq; + void *client; + void *codec; + bool force_fwload_status; + bool playback_started; + unsigned int calibration_data[CALIB_MAX]; + + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); +}; + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int *value); +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int value); +int tasdevice_spi_dev_bulk_write( + struct tasdevice_priv *tas_priv, unsigned int reg, + unsigned char *p_data, unsigned int n_length); +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tasdevice, + unsigned int reg, unsigned int mask, unsigned int value); + +void tasdevice_spi_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_spi_config_info_remove(void *context); +int tasdevice_spi_dsp_parser(void *context); +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_spi_dsp_remove(void *context); +void tasdevice_spi_calbin_remove(void *context); +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_spi_prmg_load(void *context, int prm_no); +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_spi_tuning_switch(void *context, int state); +int tas2781_spi_load_calibration(void *context, char *file_name, + unsigned short i); +#endif /* __TAS2781_SPI_H__ */ diff --git a/sound/pci/hda/tas2781_hda_spi.c b/sound/pci/hda/tas2781_hda_spi.c new file mode 100644 index 000000000000..e9b77f602c75 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_spi.c @@ -0,0 +1,1266 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <linux/acpi.h> +#include <linux/array_size.h> +#include <linux/bits.h> +#include <linux/cleanup.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/pm_runtime.h> +#include <linux/property.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/time.h> +#include <linux/types.h> +#include <linux/units.h> + +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2781-spi.h" + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +/* + * No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) { \ + .reg = xreg, .rreg = xreg, \ + .shift = xshift, .rshift = xshift,\ + .min = xmin, .max = xmax, .invert = xinvert, \ + } \ +} + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) { \ + .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, \ + .put = xhandler_put, \ + .private_value = xdata, \ +} + +struct tas2781_hda { + struct tasdevice_priv *priv; + struct acpi_device *dacpi; + struct snd_kcontrol *dsp_prog_ctl; + struct snd_kcontrol *dsp_conf_ctl; + struct snd_kcontrol *snd_ctls[3]; + struct snd_kcontrol *prof_ctl; +}; + +static const struct reg_default tas2781_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x04, 0x1a }, /* AMP inactive. */ + { 0x14, 0x3a }, /* Enable both channels. */ + { 0xb8, 0x99 }, /* Enable power up on clock. */ +}; + +static const struct regmap_range_cfg tasdevice_ranges[] = { + { + .range_min = 0, + .range_max = TASDEVICE_MAX_SIZE, + .selector_reg = TASDEVICE_PAGE_SELECT, + .selector_mask = GENMASK(7, 0), + .selector_shift = 0, + .window_start = 0, + .window_len = TASDEVICE_MAX_PAGE, + }, +}; + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 8, + .val_bits = 8, + .zero_flag_mask = true, + .cache_type = REGCACHE_NONE, + .ranges = tasdevice_ranges, + .num_ranges = ARRAY_SIZE(tasdevice_ranges), + .max_register = TASDEVICE_MAX_SIZE, + .reg_defaults = tas2781_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2781_reg_defaults), +}; + +static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { + ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, + TASDEVICE_BOOK_ID(reg)); + if (ret < 0) { + dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); + return ret; + } + tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); + } + return 0; +} + +int tasdevice_spi_dev_read(struct tasdevice_priv *tas_priv, unsigned int reg, + unsigned int *val) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char data[2]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, sizeof(data)); + *val = data[1]; + } else { + ret = regmap_read(map, TASDEVICE_PAGE_REG(reg) | 1, val); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_write(struct tasdevice_priv *tas_priv, unsigned int reg, + unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), value); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_write(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *data, unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + ret = regmap_bulk_write(map, TASDEVICE_PAGE_REG(reg), data, len); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_bulk_read(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned char *data, unsigned int len) +{ + struct regmap *map = tas_priv->regmap; + int ret; + + ret = tasdevice_spi_switch_book(tas_priv, reg); + if (ret < 0) + return ret; + + if (len > TASDEVICE_MAX_PAGE) + len = TASDEVICE_MAX_PAGE; + /* + * In our TAS2781 SPI mode, if read from other book (not book 0), + * or read from page number larger than 1 in book 0, one more byte + * read is needed, and first byte is a dummy byte, need to be ignored. + */ + if ((TASDEVICE_BOOK_ID(reg) > 0) || (TASDEVICE_PAGE_ID(reg) > 1)) { + unsigned char buf[TASDEVICE_MAX_PAGE+1]; + + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + buf, len + 1); + memcpy(data, buf + 1, len); + } else { + ret = regmap_bulk_read(map, TASDEVICE_PAGE_REG(reg) | 1, + data, len); + } + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +int tasdevice_spi_dev_update_bits(struct tasdevice_priv *tas_priv, + unsigned int reg, unsigned int mask, unsigned int value) +{ + struct regmap *map = tas_priv->regmap; + int ret, val; + + /* + * In our TAS2781 SPI mode, read/write was masked in last bit of + * address, it cause regmap_update_bits() not work as expected. + */ + ret = tasdevice_spi_dev_read(tas_priv, reg, &val); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + ret = regmap_write(map, TASDEVICE_PAGE_REG(reg), + (val & ~mask) | (mask & value)); + if (ret < 0) + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + + return ret; +} + +static void tas2781_spi_reset(struct tasdevice_priv *tas_dev) +{ + int ret; + + if (tas_dev->reset) { + gpiod_set_value_cansleep(tas_dev->reset, 0); + fsleep(800); + gpiod_set_value_cansleep(tas_dev->reset, 1); + } else { + ret = tasdevice_spi_dev_write(tas_dev, TAS2781_REG_SWRESET, + TAS2781_REG_SWRESET_RESET); + if (ret < 0) + dev_err(tas_dev->dev, "dev sw-reset fail, %d\n", ret); + } + fsleep(1000); +} + +static int tascodec_spi_init(struct tasdevice_priv *tas_priv, void *codec, + struct module *module, + void (*cont)(const struct firmware *fw, void *context)) +{ + int ret; + + /* + * Codec Lock Hold to ensure that codec_probe and firmware parsing and + * loading do not simultaneously execute. + */ + guard(mutex)(&tas_priv->codec_lock); + + ret = scnprintf(tas_priv->rca_binaryname, + sizeof(tas_priv->rca_binaryname), "%sRCA%d.bin", + tas_priv->dev_name, tas_priv->index); + if (ret <= 0) { + dev_err(tas_priv->dev, "rca name err:0x%08x\n", ret); + return ret; + } + crc8_populate_msb(tas_priv->crc8_lkp_tbl, TASDEVICE_CRC8_POLYNOMIAL); + tas_priv->codec = codec; + ret = request_firmware_nowait(module, FW_ACTION_UEVENT, + tas_priv->rca_binaryname, tas_priv->dev, GFP_KERNEL, tas_priv, + cont); + if (ret) + dev_err(tas_priv->dev, "request_firmware_nowait err:0x%08x\n", + ret); + + return ret; +} + +static void tasdevice_spi_init(struct tasdevice_priv *tas_priv) +{ + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + tas_priv->cur_book = -1; + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + + mutex_init(&tas_priv->codec_lock); +} + +static int tasdevice_spi_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask; + int max = mc->max; + int val, ret; + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_update_bits(tas_priv, + mc->reg, mask, (unsigned int)(val << mc->shift)); + if (ret) + dev_err(tas_priv->dev, "set AMP vol error in dev %d\n", + tas_priv->index); + + return ret; +} + +static int tasdevice_spi_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + unsigned char mask = 0; + int max = mc->max; + int ret, val; + + /* Read the primary device */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get AMP vol error\n", __func__); + return ret; + } + + mask = rounddown_pow_of_two(max); + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tasdevice_spi_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int val, ret; + + val = clamp(invert ? max - ucontrol->value.integer.value[0] : + ucontrol->value.integer.value[0], 0, max); + ret = tasdevice_spi_dev_write(tas_priv, mc->reg, (unsigned int)val); + if (ret) + dev_err(tas_priv->dev, "set digital vol err in dev %d\n", + tas_priv->index); + + return ret; + +} + +static int tasdevice_spi_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc) +{ + unsigned int invert = mc->invert; + int max = mc->max; + int ret, val; + + /* Read the primary device as the whole */ + ret = tasdevice_spi_dev_read(tas_priv, mc->reg, &val); + if (ret) { + dev_err(tas_priv->dev, "%s, get digital vol err\n", __func__); + return ret; + } + + val = clamp(invert ? max - val : val, 0, max); + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2781_read_acpi(struct tas2781_hda *tas_hda, const char *hid, + int id) +{ + struct tasdevice_priv *p = tas_hda->priv; + struct acpi_device *adev; + struct device *physdev; + u32 values[HDA_MAX_COMPONENTS]; + const char *property; + size_t nval; + int ret, i; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, "Failed to find ACPI device: %s\n", hid); + return -ENODEV; + } + + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + tas_hda->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL */ + p->acpi_subsystem_id = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + + property = "ti,dev-index"; + ret = device_property_count_u32(physdev, property); + if ((ret <= 0) || (ret > ARRAY_SIZE(values))) { + ret = -EINVAL; + goto err; + } + nval = ret; + + ret = device_property_read_u32_array(physdev, property, values, nval); + if (ret) + goto err; + + p->index = U8_MAX; + for (i = 0; i < nval; i++) { + if (values[i] == id) { + p->index = i; + break; + } + } + if (p->index == U8_MAX) { + dev_dbg(p->dev, "No index found in %s\n", property); + ret = -ENODEV; + goto err; + } + + if (p->index == 0) { + /* All of amps share same RESET pin. */ + p->reset = devm_gpiod_get_index_optional(physdev, "reset", + p->index, GPIOD_OUT_LOW); + if (IS_ERR(p->reset)) { + dev_err_probe(p->dev, ret, "Failed on reset GPIO\n"); + goto err; + } + } + + put_device(physdev); + + return 0; + +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + acpi_dev_put(adev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + if ((action == HDA_GEN_PCM_ACT_OPEN) || + (action == HDA_GEN_PCM_ACT_CLOSE)) { + if (action == HDA_GEN_PCM_ACT_OPEN) + pm_runtime_get_sync(dev); + guard(mutex)(&tas_hda->priv->codec_lock); + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + if (action == HDA_GEN_PCM_ACT_CLOSE) { + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + } + } else { + dev_dbg(tas_hda->priv->dev, + "Playback action not supported: %d\n", action); + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->rcabin.ncfgs - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + if (tas_priv->rcabin.profile_cfg_id != val) { + tas_priv->rcabin.profile_cfg_id = val; + return 1; + } + + return 0; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->fmw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_priv->fmw->nr_programs - 1; + int val; + + val = clamp(nr_program, 0, max); + + if (tas_priv->cur_prog != val) { + tas_priv->cur_prog = val; + return 1; + } + + return 0; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int max = tas_priv->fmw->nr_configurations - 1; + int val; + + val = clamp(ucontrol->value.integer.value[0], 0, max); + + if (tas_priv->cur_conf != val) { + tas_priv->cur_conf = val; + return 1; + } + + return 0; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + * + * Return 0 if succeeded. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_spi_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_spi_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_spi_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) { + change = false; + } else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + str_on_off(tas_priv->force_fwload_status)); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 0", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 0", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 0", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain 1", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain 1", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load 1", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl[] = { +{ + .name = "Speaker Profile Id - 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +{ + .name = "Speaker Profile Id - 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}, +}; +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl[] = { +{ + .name = "Speaker Program Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +{ + .name = "Speaker Program Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl[] = { +{ + .name = "Speaker Config Id 0", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +{ + .name = "Speaker Config Id 1", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + static const unsigned int cali_array[CALIB_MAX] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), + }; + int j, rc; + + /* + * If no calibration data exist in tasdevice_priv *tas_priv, + * calibration apply will be ignored, and use default values + * in firmware binary, which was loaded during firmware download. + */ + if (tas_priv->calibration_data[0] == 0) + return; + /* + * Calibration data was saved in tasdevice_priv *tas_priv as: + * unsigned int calibration_data[CALIB_MAX]; + * and every data (in 4 bytes) will be saved in register which in + * book 0, and page number in page_array[], offset was saved in + * rgno_array[]. + */ + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_spi_dev_bulk_write(tas_priv, cali_array[j], + (unsigned char *)&tas_priv->calibration_data[j], 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + tas_priv->index, j, rc); + } +} + +/* + * Update the calibration data, including speaker impedance, f0, etc, + * into algo. Calibrate data is done by manufacturer in the factory. + * These data are used by Algo for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * Calibration data format in EFI is V2, since 2024. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + /* + * GUID was used for data access in BIOS, it was provided by board + * manufactory, like HP: "{02f9af02-7734-4233-b43d-93fe5aa35db3}" + */ + efi_guid_t efi_guid = + EFI_GUID(0x02f9af02, 0x7734, 0x4233, + 0xb4, 0x3d, 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + unsigned char data[TASDEVICE_CALIBRATION_DATA_SIZE]; + struct tm *tm = &tas_priv->tm; + unsigned long total_sz = 0; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->calibration_data[0] = 0; + status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, + data); + if (status == EFI_BUFFER_TOO_SMALL) { + if (total_sz > TASDEVICE_CALIBRATION_DATA_SIZE) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &total_sz, data); + } + if (status != EFI_SUCCESS) + return -EINVAL; + + tmp_val = (unsigned int *)data; + if (tmp_val[0] == 2781) { + /* + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and + * uniqie_id for multi-link solutions + * 2. Support flexible number of devices instead of + * fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2781, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ + crc = crc32(~0, data, 3 + tmp_val[0] * 6) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[3 + tmp_val[0] * 6]); + + if (crc == tmp_val[3 + tmp_val[0] * 6]) { + time64_to_tm(tmp_val[2], 0, tm); + dev_dbg(tas_priv->dev, "Calibration date %pt.\n", tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->calibration_data[i] = + tmp_val[3 + tas_priv->index * 6 + i]; + tas_priv->apply_calibration(tas_priv); + } + } else { + /* + * Calibration data is in V1 format. + * struct cali_data { + * char cali_data[20]; + * } + * + * struct { + * struct cali_data cali_data[4]; + * int TimeStamp of Calibration (4 bytes) + * int CRC (4 bytes) + * } ueft; + */ + crc = crc32(~0, data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "Calibration date %pt.\n", tm); + for (int i = 0; i < CALIB_MAX; i++) + tas_priv->calibration_data[i] = + tmp_val[tas_priv->index * 5 + i]; + tas_priv->apply_calibration(tas_priv); + } + } + + return 0; +} + +static void tas2781_hda_remove_controls(struct tas2781_hda *tas_hda) +{ + struct hda_codec *codec = tas_hda->priv->codec; + + if (tas_hda->dsp_prog_ctl) + snd_ctl_remove(codec->card, tas_hda->dsp_prog_ctl); + + if (tas_hda->dsp_conf_ctl) + snd_ctl_remove(codec->card, tas_hda->dsp_conf_ctl); + + for (int i = ARRAY_SIZE(tas_hda->snd_ctls) - 1; i >= 0; i--) + if (tas_hda->snd_ctls[i]) + snd_ctl_remove(codec->card, tas_hda->snd_ctls[i]); + + if (tas_hda->prof_ctl) + snd_ctl_remove(codec->card, tas_hda->prof_ctl); +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); + struct hda_codec *codec = tas_priv->codec; + int i, j, ret; + + pm_runtime_get_sync(tas_priv->dev); + guard(mutex)(&tas_priv->codec_lock); + + ret = tasdevice_spi_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + /* Add control one time only. */ + tas_hda->prof_ctl = snd_ctl_new1(&tas2781_prof_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->prof_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl[tas_priv->index].name, ret); + goto out; + } + j = tas_priv->index * 3; + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls)/2; i++) { + tas_hda->snd_ctls[i] = + snd_ctl_new1(&tas2781_snd_controls[i + j], tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->snd_ctls[i]); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add snd control %s = %d\n", + tas2781_snd_controls[i + j].name, ret); + goto out; + } + } + + tasdevice_spi_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%08X-%01d.bin", + codec->core.subsystem_id, tas_priv->index); + ret = tasdevice_spi_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + /* Add control one time only. */ + tas_hda->dsp_prog_ctl = + snd_ctl_new1(&tas2781_dsp_prog_ctrl[tas_priv->index], + tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_prog_ctl); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_hda->dsp_conf_ctl = snd_ctl_new1( + &tas2781_dsp_conf_ctrl[tas_priv->index], tas_priv); + ret = snd_ctl_add(codec->card, tas_hda->dsp_conf_ctl); + if (ret) { + dev_err(tas_priv->dev, "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl[tas_priv->index].name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_spi_prmg_load(tas_priv, 0); + if (tas_priv->fmw->nr_programs > 0) + tas_priv->cur_prog = 0; + if (tas_priv->fmw->nr_configurations > 0) + tas_priv->cur_conf = 0; + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + /* + * If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas_priv->save_calibration(tas_priv); + +out: + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_hda->priv->dev); + pm_runtime_put_autosuspend(tas_hda->priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + struct hda_codec *codec; + int ret; + + if (tas_hda->priv->index < 0 || + tas_hda->priv->index >= HDA_MAX_COMPONENTS || + (!comps)) + return -EINVAL; + + comps = &comps[tas_hda->priv->index]; + if (comps->dev) + return -EBUSY; + + codec = comps->codec; + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_spi_init(tas_hda->priv, codec, THIS_MODULE, + tasdev_fw_ready); + if (!ret) + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return ret; +} + +static void tas2781_hda_unbind(struct device *dev, struct device *master, + void *master_data) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + comps = &comps[tas_hda->priv->index]; + + if (comps->dev == dev) { + comps->dev = NULL; + memset(comps->name, 0, sizeof(comps->name)); + comps->playback_hook = NULL; + } + + tas2781_hda_remove_controls(tas_hda); + + tasdevice_spi_config_info_remove(tas_hda->priv); + tasdevice_spi_dsp_remove(tas_hda->priv); + + tas_hda->priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_hda->priv->dev); + pm_runtime_disable(tas_hda->priv->dev); + + component_del(tas_hda->priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_hda->priv->dev); + + mutex_destroy(&tas_hda->priv->codec_lock); +} + +static int tas2781_hda_spi_probe(struct spi_device *spi) +{ + struct tasdevice_priv *tas_priv; + struct tas2781_hda *tas_hda; + const char *device_name; + int ret = 0; + + tas_hda = devm_kzalloc(&spi->dev, sizeof(*tas_hda), GFP_KERNEL); + if (!tas_hda) + return -ENOMEM; + + spi->max_speed_hz = TAS2781_SPI_MAX_FREQ; + + tas_priv = devm_kzalloc(&spi->dev, sizeof(*tas_priv), GFP_KERNEL); + if (!tas_priv) + goto err; + tas_priv->dev = &spi->dev; + tas_hda->priv = tas_priv; + tas_priv->regmap = devm_regmap_init_spi(spi, &tasdevice_regmap); + if (IS_ERR(tas_priv->regmap)) { + ret = PTR_ERR(tas_priv->regmap); + dev_err(tas_priv->dev, "Failed to allocate regmap: %d\n", + ret); + goto err; + } + if (strstr(dev_name(&spi->dev), "TXNW2781")) { + device_name = "TXNW2781"; + tas_priv->save_calibration = tas2781_save_calibration; + tas_priv->apply_calibration = tas2781_apply_calib; + } else { + goto err; + } + + tas_priv->irq = spi->irq; + dev_set_drvdata(&spi->dev, tas_hda); + ret = tas2781_read_acpi(tas_hda, device_name, + spi_get_chipselect(spi, 0)); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + tasdevice_spi_init(tas_priv); + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + tas2781_spi_reset(tas_priv); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + } + +err: + if (ret) + tas2781_hda_remove(&spi->dev); + + return ret; +} + +static void tas2781_hda_spi_remove(struct spi_device *spi) +{ + tas2781_hda_remove(&spi->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + if (tas_hda->priv->playback_started) { + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + tas_hda->priv->playback_started = false; + } + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + + guard(mutex)(&tas_hda->priv->codec_lock); + + tasdevice_spi_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); + + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + tas_hda->priv->apply_calibration(tas_hda->priv); + + return 0; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + tasdevice_spi_tuning_switch(tas_hda->priv, 1); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tas2781_hda *tas_hda = dev_get_drvdata(dev); + int ret; + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + guard(mutex)(&tas_hda->priv->codec_lock); + + tas_hda->priv->cur_book = -1; + tas_hda->priv->cur_conf = -1; + + tas2781_spi_reset(tas_hda->priv); + tasdevice_spi_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); + + if (tas_hda->priv->playback_started) + tasdevice_spi_tuning_switch(tas_hda->priv, 0); + + /* + * If calibrated data occurs error, dsp will still work with default + * calibrated data inside algo. + */ + tas_hda->priv->apply_calibration(tas_hda->priv); + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct spi_device_id tas2781_hda_spi_id[] = { + { "tas2781-hda", }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TXNW2781", }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct spi_driver tas2781_hda_spi_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_spi_id, + .probe = tas2781_hda_spi_probe, + .remove = tas2781_hda_spi_remove, +}; +module_spi_driver(tas2781_hda_spi_driver); + +MODULE_DESCRIPTION("TAS2781 HDA SPI Driver"); +MODULE_AUTHOR("Baojun, Xu, <baojun.xug(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c new file mode 100644 index 000000000000..6e6a12df56e2 --- /dev/null +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -0,0 +1,2101 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA SPI driver +// +// Copyright 2024 Texas Instruments, Inc. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> + +#include <asm/unaligned.h> +#include <linux/crc8.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/types.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tas2781-dsp.h> +#include <sound/tlv.h> + +#include "tas2781-spi.h" + +#define OFFSET_ERROR_BIT BIT(31) + +#define ERROR_PRAM_CRCCHK 0x0000000 +#define ERROR_YRAM_CRCCHK 0x0000001 +#define PPC_DRIVER_CRCCHK 0x00000200 + +#define TAS2781_SA_COEFF_SWAP_REG TASDEVICE_REG(0, 0x35, 0x2c) +#define TAS2781_YRAM_BOOK1 140 +#define TAS2781_YRAM1_PAGE 42 +#define TAS2781_YRAM1_START_REG 88 + +#define TAS2781_YRAM2_START_PAGE 43 +#define TAS2781_YRAM2_END_PAGE 49 +#define TAS2781_YRAM2_START_REG 8 +#define TAS2781_YRAM2_END_REG 127 + +#define TAS2781_YRAM3_PAGE 50 +#define TAS2781_YRAM3_START_REG 8 +#define TAS2781_YRAM3_END_REG 27 + +/* should not include B0_P53_R44-R47 */ +#define TAS2781_YRAM_BOOK2 0 +#define TAS2781_YRAM4_START_PAGE 50 +#define TAS2781_YRAM4_END_PAGE 60 + +#define TAS2781_YRAM5_PAGE 61 +#define TAS2781_YRAM5_START_REG TAS2781_YRAM3_START_REG +#define TAS2781_YRAM5_END_REG TAS2781_YRAM3_END_REG + +#define TASDEVICE_MAXPROGRAM_NUM_KERNEL 5 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL_MULTIPLE_AMPS 64 +#define TASDEVICE_MAXCONFIG_NUM_KERNEL 10 +#define MAIN_ALL_DEVICES_1X 0x01 +#define MAIN_DEVICE_A_1X 0x02 +#define MAIN_DEVICE_B_1X 0x03 +#define MAIN_DEVICE_C_1X 0x04 +#define MAIN_DEVICE_D_1X 0x05 +#define COEFF_DEVICE_A_1X 0x12 +#define COEFF_DEVICE_B_1X 0x13 +#define COEFF_DEVICE_C_1X 0x14 +#define COEFF_DEVICE_D_1X 0x15 +#define PRE_DEVICE_A_1X 0x22 +#define PRE_DEVICE_B_1X 0x23 +#define PRE_DEVICE_C_1X 0x24 +#define PRE_DEVICE_D_1X 0x25 +#define PRE_SOFTWARE_RESET_DEVICE_A 0x41 +#define PRE_SOFTWARE_RESET_DEVICE_B 0x42 +#define PRE_SOFTWARE_RESET_DEVICE_C 0x43 +#define PRE_SOFTWARE_RESET_DEVICE_D 0x44 +#define POST_SOFTWARE_RESET_DEVICE_A 0x45 +#define POST_SOFTWARE_RESET_DEVICE_B 0x46 +#define POST_SOFTWARE_RESET_DEVICE_C 0x47 +#define POST_SOFTWARE_RESET_DEVICE_D 0x48 + +struct tas_crc { + unsigned char offset; + unsigned char len; +}; + +struct blktyp_devidx_map { + unsigned char blktyp; + unsigned char dev_idx; +}; + +/* fixed m68k compiling issue: mapping table can save code field */ +static const struct blktyp_devidx_map ppc3_tas2781_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { PRE_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { POST_SOFTWARE_RESET_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { PRE_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { POST_SOFTWARE_RESET_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { PRE_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { POST_SOFTWARE_RESET_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, + { PRE_SOFTWARE_RESET_DEVICE_D, 0xC4 }, + { POST_SOFTWARE_RESET_DEVICE_D, 0xC4 }, +}; + +static const struct blktyp_devidx_map ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES_1X, 0x80 }, + { MAIN_DEVICE_A_1X, 0x81 }, + { COEFF_DEVICE_A_1X, 0xC1 }, + { PRE_DEVICE_A_1X, 0xC1 }, + { MAIN_DEVICE_B_1X, 0x82 }, + { COEFF_DEVICE_B_1X, 0xC2 }, + { PRE_DEVICE_B_1X, 0xC2 }, + { MAIN_DEVICE_C_1X, 0x83 }, + { COEFF_DEVICE_C_1X, 0xC3 }, + { PRE_DEVICE_C_1X, 0xC3 }, + { MAIN_DEVICE_D_1X, 0x84 }, + { COEFF_DEVICE_D_1X, 0xC4 }, + { PRE_DEVICE_D_1X, 0xC4 }, +}; + +static const struct blktyp_devidx_map non_ppc3_mapping_table[] = { + { MAIN_ALL_DEVICES, 0x80 }, + { MAIN_DEVICE_A, 0x81 }, + { COEFF_DEVICE_A, 0xC1 }, + { PRE_DEVICE_A, 0xC1 }, + { MAIN_DEVICE_B, 0x82 }, + { COEFF_DEVICE_B, 0xC2 }, + { PRE_DEVICE_B, 0xC2 }, + { MAIN_DEVICE_C, 0x83 }, + { COEFF_DEVICE_C, 0xC3 }, + { PRE_DEVICE_C, 0xC3 }, + { MAIN_DEVICE_D, 0x84 }, + { COEFF_DEVICE_D, 0xC4 }, + { PRE_DEVICE_D, 0xC4 }, +}; + +static struct tasdevice_config_info *tasdevice_add_config( + struct tasdevice_priv *tas_priv, unsigned char *config_data, + unsigned int config_size, int *status) +{ + struct tasdevice_config_info *cfg_info; + struct tasdev_blk_data **bk_da; + unsigned int config_offset = 0; + unsigned int i; + + /* + * In most projects are many audio cases, such as music, handfree, + * receiver, games, audio-to-haptics, PMIC record, bypass mode, + * portrait, landscape, etc. Even in multiple audios, one or + * two of the chips will work for the special case, such as + * ultrasonic application. In order to support these variable-numbers + * of audio cases, flexible configs have been introduced in the + * DSP firmware. + */ + cfg_info = kzalloc(sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + return NULL; + } + + if (tas_priv->rcabin.fw_hdr.binary_version_num >= 0x105) { + if ((config_offset + 64) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add conf: Out of boundary\n"); + goto out; + } + config_offset += 64; + } + + if ((config_offset + 4) > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, "add config: Out of boundary\n"); + goto out; + } + + /* + * convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + /* + * Several kinds of dsp/algorithm firmwares can run on tas2781, + * the number and size of blk are not fixed and different among + * these firmwares. + */ + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(*bk_da), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d nblocks = %u!\n", + __func__, i, cfg_info->nblocks); + goto out1; + } + bk_da[i] = kzalloc(sizeof(*bk_da[i]), GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + goto out1; + } + + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(tas_priv->dev, + "%s: Out of boundary: i = %d blks = %u!\n", + __func__, i, cfg_info->nblocks); + goto out1; + } + /* instead of kzalloc+memcpy */ + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + i++; + goto out1; + } + + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } + + return cfg_info; +out1: + for (int j = 0; j < i; j++) + kfree(bk_da[j]); + kfree(bk_da); +out: + kfree(cfg_info); + return NULL; +} + +int tasdevice_spi_rca_parser(void *context, const struct firmware *fmw) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_config_info **cfg_info; + struct tasdevice_rca_hdr *fw_hdr; + struct tasdevice_rca *rca; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + rca = &(tas_priv->rcabin); + fw_hdr = &(rca->fw_hdr); + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "Failed to read %s\n", + tas_priv->rca_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + buf = (unsigned char *)fmw->data; + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(tas_priv->dev, + "File size not match, %d %u", (int)fmw->size, + fw_hdr->img_sz); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(tas_priv->dev, "File version 0x%04x is too low", + fw_hdr->binary_version_num); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset++]; + fw_hdr->dev_family = buf[offset++]; + fw_hdr->reserve = buf[offset++]; + fw_hdr->ndev = buf[offset++]; + if (offset + TASDEVICE_DEVICE_SUM > fw_hdr->img_sz) { + dev_err(tas_priv->dev, "rca_ready: Out of boundary!\n"); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(tas_priv->dev, "Bin file err %d - %d != %d!\n", + fw_hdr->img_sz, total_config_sz, (int)offset); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -EINVAL; + } + + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return -ENOMEM; + } + rca->cfg_info = cfg_info; + rca->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + rca->ncfgs += 1; + cfg_info[i] = tasdevice_add_config(tas_priv, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + return ret; + } + offset += (int)fw_hdr->config_size[i]; + } + + return ret; +} + +/* fixed m68k compiling issue: mapping table can save code field */ +static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block) +{ + struct blktyp_devidx_map *p = + (struct blktyp_devidx_map *)non_ppc3_mapping_table; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + int i, n = ARRAY_SIZE(non_ppc3_mapping_table); + unsigned char dev_idx = 0; + + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; + n = ARRAY_SIZE(ppc3_tas2781_mapping_table); + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + p = (struct blktyp_devidx_map *)ppc3_mapping_table; + n = ARRAY_SIZE(ppc3_mapping_table); + } + + for (i = 0; i < n; i++) { + if (block->type == p[i].blktyp) { + dev_idx = p[i].dev_idx; + break; + } + } + + return dev_idx; +} + +static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + + if (offset + 16 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host. + */ + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + block->is_pchksum_present = data[offset++]; + block->pchksum = data[offset++]; + block->is_ychksum_present = data[offset++]; + block->ychksum = data[offset++]; + block->blk_size = get_unaligned_be32(&data[offset]); + offset += 4; + block->nr_subblocks = get_unaligned_be32(&data[offset]); + offset += 4; + + /* + * Fixed m68k compiling issue: + * 1. mapping table can save code field. + * 2. storing the dev_idx as a member of block can reduce unnecessary + * time and system resource comsumption of dev_idx mapping every + * time the block data writing to the dsp. + */ + block->dev_idx = map_dev_idx(tas_fmw, block); + + if (offset + block->blk_size > fmw->size) { + dev_err(tas_fmw->dev, "%s: nSublocks error\n", __func__); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], block->blk_size, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += block->blk_size; + + return offset; +} + +static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = fmw->data; + struct tasdev_blk *blk; + unsigned int i; + + if (offset + 4 > fmw->size) { + dev_err(tas_fmw->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + img_data->nr_blk = get_unaligned_be32(&data[offset]); + offset += 4; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(struct tasdev_blk), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data_kernel( + tas_fmw, blk, fmw, offset); + if (offset < 0) { + kfree(img_data->dev_blks); + return -EINVAL; + } + } + + return offset; +} + +static int fw_parse_program_data_kernel( + struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset) +{ + struct tasdevice_prog *program; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + if (offset + 72 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + /* skip 72 unused byts */ + offset += 72; + + offset = fw_parse_data_kernel(tas_fmw, &(program->dev_data), + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +static int fw_parse_configuration_data_kernel(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + const unsigned char *data = fmw->data; + struct tasdevice_config *config; + unsigned int i; + + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 80 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + /* skip extra 16 bytes */ + offset += 80; + + offset = fw_parse_data_kernel(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +static int fw_parse_variable_header_kernel(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_config *config; + struct tasdevice_prog *program; + const unsigned char *buf = fmw->data; + unsigned short max_confs; + unsigned int i; + + if (offset + 12 + 4 * TASDEVICE_MAXPROGRAM_NUM_KERNEL > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); + return -EINVAL; + } + offset += 2; + fw_hdr->device = get_unaligned_be16(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 2; + + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); + offset += 4; + + if (tas_fmw->nr_programs == 0 || + tas_fmw->nr_programs > TASDEVICE_MAXPROGRAM_NUM_KERNEL) { + dev_err(tas_priv->dev, "mnPrograms is invalid\n"); + return -EINVAL; + } + + tas_fmw->programs = kcalloc(tas_fmw->nr_programs, + sizeof(*tas_fmw->programs), GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + program = &(tas_fmw->programs[i]); + program->prog_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused prog_size */ + offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); + + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); + offset += 4; + + /* + * The max number of config in firmware greater than 4 pieces of + * tas2781s is different from the one lower than 4 pieces of + * tas2781s. + */ + max_confs = TASDEVICE_MAXCONFIG_NUM_KERNEL; + if (tas_fmw->nr_configurations == 0 || + tas_fmw->nr_configurations > max_confs) { + dev_err(tas_priv->dev, "%s: Conf is invalid\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + if (offset + 4 * max_confs > fmw->size) { + dev_err(tas_priv->dev, "%s: mpConfigurations err\n", __func__); + kfree(tas_fmw->programs); + return -EINVAL; + } + + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) { + kfree(tas_fmw->programs); + return -ENOMEM; + } + + for (i = 0; i < tas_fmw->nr_programs; i++) { + config = &(tas_fmw->configs[i]); + config->cfg_size = get_unaligned_be32(&buf[offset]); + offset += 4; + } + + /* Skip the unused configs */ + offset += 4 * (max_confs - tas_fmw->nr_programs); + + return offset; +} + +static int tasdevice_single_byte_wr(void *context, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int i, subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 * len > sublocksize) { + dev_err(tas_priv->dev, "process_block: Out of boundary\n"); + return 0; + } + + for (i = 0; i < len; i++) { + rc = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], + data[subblk_offset + 1], data[subblk_offset + 2]), + data[subblk_offset + 3]); + if (rc < 0) { + dev_err(tas_priv->dev, + "process_block: single write error\n"); + subblk_offset |= OFFSET_ERROR_BIT; + } + subblk_offset += 4; + } + + return subblk_offset; +} + +static int tasdevice_burst_wr(void *context, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned short len = get_unaligned_be16(&data[2]); + int subblk_offset, rc; + + subblk_offset = 4; + if (subblk_offset + 4 + len > sublocksize) { + dev_err(tas_priv->dev, "%s: BST Out of boundary\n", __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + if (len % 4) { + dev_err(tas_priv->dev, "%s:Bst-len(%u)not div by 4\n", + __func__, len); + subblk_offset |= OFFSET_ERROR_BIT; + } + + rc = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(data[subblk_offset], data[subblk_offset + 1], + data[subblk_offset + 2]), &(data[subblk_offset + 4]), len); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: bulk_write error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + subblk_offset += (len + 4); + + return subblk_offset; +} + +static int tasdevice_delay(void *context, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + unsigned int sleep_time, subblk_offset = 2; + + if (subblk_offset + 2 > sublocksize) { + dev_err(tas_priv->dev, "%s: delay Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + sleep_time = get_unaligned_be16(&data[2]) * 1000; + fsleep(sleep_time); + subblk_offset += 2; + + return subblk_offset; +} + +static int tasdevice_field_wr(void *context, unsigned char *data, + int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int rc, subblk_offset = 2; + + if (subblk_offset + 6 > sublocksize) { + dev_err(tas_priv->dev, "%s: bit write Out of boundary\n", + __func__); + subblk_offset |= OFFSET_ERROR_BIT; + } + rc = tasdevice_spi_dev_update_bits(tas_priv, + TASDEVICE_REG(data[subblk_offset + 2], + data[subblk_offset + 3], + data[subblk_offset + 4]), + data[subblk_offset + 1], data[subblk_offset + 5]); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: update_bits error = %d\n", + __func__, rc); + subblk_offset |= OFFSET_ERROR_BIT; + } + subblk_offset += 6; + + return subblk_offset; +} + +static int tasdevice_process_block(void *context, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct tasdevice_priv *tas_priv = context; + int blktyp = dev_idx & 0xC0, subblk_offset; + unsigned char subblk_typ = data[1]; + + if (tas_priv->is_loading == false) + return sublocksize; + + switch (subblk_typ) { + case TASDEVICE_CMD_SING_W: + subblk_offset = tasdevice_single_byte_wr(tas_priv, data, + sublocksize); + break; + case TASDEVICE_CMD_BURST: + subblk_offset = tasdevice_burst_wr(tas_priv, data, + sublocksize); + break; + case TASDEVICE_CMD_DELAY: + subblk_offset = tasdevice_delay(tas_priv, data, sublocksize); + break; + case TASDEVICE_CMD_FIELD_W: + subblk_offset = tasdevice_field_wr(tas_priv, data, + sublocksize); + break; + default: + subblk_offset = 2; + break; + } + if (((subblk_offset & OFFSET_ERROR_BIT) != 0) && blktyp != 0) { + if (blktyp == 0x80) { + tas_priv->cur_prog = -1; + tas_priv->cur_conf = -1; + } else + tas_priv->cur_conf = -1; + } + subblk_offset &= ~OFFSET_ERROR_BIT; + + return subblk_offset; +} + +void tasdevice_spi_select_cfg_blk(void *pContext, int conf_no, + unsigned char block_type) +{ + struct tasdevice_priv *tas_priv = pContext; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdev_blk_data **blk_data; + unsigned int j, k; + + if (conf_no >= rca->ncfgs || conf_no < 0 || !cfg_info) { + dev_err(tas_priv->dev, "conf_no should be not more than %u\n", + rca->ncfgs); + return; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, rc = 0; + + if (block_type > 5 || block_type < 2) { + dev_err(tas_priv->dev, + "block_type should be in range from 2 to 5\n"); + break; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < blk_data[j]->n_subblks; k++) { + tas_priv->is_loading = true; + + rc = tasdevice_process_block(tas_priv, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += rc; + if (blk_data[j]->block_size < length) { + dev_err(tas_priv->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(tas_priv->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } +} + +static int tasdevice_load_block_kernel( + struct tasdevice_priv *tasdevice, struct tasdev_blk *block) +{ + const unsigned int blk_size = block->blk_size; + unsigned char *data = block->data; + unsigned int i, length; + + for (i = 0, length = 0; i < block->nr_subblocks; i++) { + int rc = tasdevice_process_block(tasdevice, data + length, + block->dev_idx, blk_size - length); + if (rc < 0) { + dev_err(tasdevice->dev, + "%s: %u %u sublock write error\n", + __func__, length, blk_size); + break; + } + length += rc; + if (blk_size < length) { + dev_err(tasdevice->dev, "%s: %u %u out of boundary\n", + __func__, length, blk_size); + break; + } + } + + return 0; +} + +static int fw_parse_variable_hdr(struct tasdevice_priv *tas_priv, + struct tasdevice_dspfw_hdr *fw_hdr, + const struct firmware *fmw, int offset) +{ + const unsigned char *buf = fmw->data; + int len = strlen((char *)&buf[offset]); + + len++; + + if (offset + len + 8 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + + offset += len; + + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device_family != 0) { + dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); + return -EINVAL; + } + offset += 4; + + fw_hdr->device = get_unaligned_be32(&buf[offset]); + if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || + fw_hdr->device == 6) { + dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); + return -EINVAL; + } + offset += 4; + fw_hdr->ndev = 1; + + return offset; +} + +static int fw_parse_variable_header_git(struct tasdevice_priv + *tas_priv, const struct firmware *fmw, int offset) +{ + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + + offset = fw_parse_variable_hdr(tas_priv, fw_hdr, fmw, offset); + + return offset; +} + +static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, + struct tasdev_blk *block, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + int n; + + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Type error\n", __func__); + return -EINVAL; + } + block->type = get_unaligned_be32(&data[offset]); + offset += 4; + + if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { + if (offset + 8 > fmw->size) { + dev_err(tas_fmw->dev, "PChkSumPresent error\n"); + return -EINVAL; + } + block->is_pchksum_present = data[offset]; + offset++; + + block->pchksum = data[offset]; + offset++; + + block->is_ychksum_present = data[offset]; + offset++; + + block->ychksum = data[offset]; + offset++; + } else { + block->is_pchksum_present = 0; + block->is_ychksum_present = 0; + } + + block->nr_cmds = get_unaligned_be32(&data[offset]); + offset += 4; + + n = block->nr_cmds * 4; + if (offset + n > fmw->size) { + dev_err(tas_fmw->dev, + "%s: File Size(%lu) error offset = %d n = %d\n", + __func__, (unsigned long)fmw->size, offset, n); + return -EINVAL; + } + /* instead of kzalloc+memcpy */ + block->data = kmemdup(&data[offset], n, GFP_KERNEL); + if (!block->data) + return -ENOMEM; + + offset += n; + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_data(struct tasdevice_fw *tas_fmw, + struct tasdevice_data *img_data, const struct firmware *fmw, + int offset) +{ + const unsigned char *data = (unsigned char *)fmw->data; + struct tasdev_blk *blk; + unsigned int i, n; + + if (offset + 64 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Name error\n", __func__); + return -EINVAL; + } + memcpy(img_data->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n++; + if (offset + n + 2 > fmw->size) { + dev_err(tas_fmw->dev, "%s: Description error\n", __func__); + return -EINVAL; + } + offset += n; + img_data->nr_blk = get_unaligned_be16(&data[offset]); + offset += 2; + + img_data->dev_blks = kcalloc(img_data->nr_blk, + sizeof(*img_data->dev_blks), GFP_KERNEL); + if (!img_data->dev_blks) + return -ENOMEM; + + for (i = 0; i < img_data->nr_blk; i++) { + blk = &(img_data->dev_blks[i]); + offset = fw_parse_block_data(tas_fmw, blk, fmw, offset); + if (offset < 0) + return -EINVAL; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_program_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *buf = (unsigned char *)fmw->data; + struct tasdevice_prog *program; + int i; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); + offset += 2; + + if (tas_fmw->nr_programs == 0) { + /* Not error in calibration Data file, return directly */ + dev_info(tas_priv->dev, "%s: No Programs data, maybe calbin\n", + __func__); + return offset; + } + + tas_fmw->programs = + kcalloc(tas_fmw->nr_programs, sizeof(*tas_fmw->programs), + GFP_KERNEL); + if (!tas_fmw->programs) + return -ENOMEM; + + for (i = 0; i < tas_fmw->nr_programs; i++) { + int n = 0; + + program = &(tas_fmw->programs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "%s: mpName error\n", __func__); + return -EINVAL; + } + offset += 64; + + n = strlen((char *)&buf[offset]); + /* skip '\0' and 5 unused bytes */ + n += 6; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + + offset += n; + + offset = fw_parse_data(tas_fmw, &(program->dev_data), fmw, + offset); + if (offset < 0) + return offset; + } + + return offset; +} + +/* + * When parsing error occurs, all the memory resource will be released + * in the end of tasdevice_rca_ready. + */ +static int fw_parse_configuration_data(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + unsigned char *data = (unsigned char *)fmw->data; + struct tasdevice_config *config; + unsigned int i, n; + + if (offset + 2 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + return -EINVAL; + } + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); + offset += 2; + + if (tas_fmw->nr_configurations == 0) { + dev_err(tas_priv->dev, "%s: Conf is zero\n", __func__); + /* Not error for calibration Data file, return directly */ + return offset; + } + tas_fmw->configs = kcalloc(tas_fmw->nr_configurations, + sizeof(*tas_fmw->configs), GFP_KERNEL); + if (!tas_fmw->configs) + return -ENOMEM; + for (i = 0; i < tas_fmw->nr_configurations; i++) { + config = &(tas_fmw->configs[i]); + if (offset + 64 > fmw->size) { + dev_err(tas_priv->dev, "File Size err\n"); + return -EINVAL; + } + memcpy(config->name, &data[offset], 64); + offset += 64; + + n = strlen((char *)&data[offset]); + n += 15; + if (offset + n > fmw->size) { + dev_err(tas_priv->dev, "Description err\n"); + return -EINVAL; + } + offset += n; + offset = fw_parse_data(tas_fmw, &(config->dev_data), + fmw, offset); + if (offset < 0) + break; + } + + return offset; +} + +static bool check_inpage_yram_rg(struct tas_crc *cd, + unsigned char reg, unsigned char len) +{ + bool in = false; + + if (reg <= TAS2781_YRAM5_END_REG && + reg >= TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_END_REG) + cd->len = TAS2781_YRAM5_END_REG - reg + 1; + else + cd->len = len; + cd->offset = reg; + in = true; + } else if (reg < TAS2781_YRAM5_START_REG) { + if (reg + len > TAS2781_YRAM5_START_REG) { + cd->offset = TAS2781_YRAM5_START_REG; + cd->len = len - TAS2781_YRAM5_START_REG + reg; + in = true; + } + } + + return in; +} + +static bool check_inpage_yram_bk1(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (page == TAS2781_YRAM1_PAGE) { + if (reg >= TAS2781_YRAM1_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg + len > TAS2781_YRAM1_START_REG) { + cd->offset = TAS2781_YRAM1_START_REG; + cd->len = len - TAS2781_YRAM1_START_REG + reg; + in = true; + } + } else if (page == TAS2781_YRAM3_PAGE) { + in = check_inpage_yram_rg(cd, reg, len); + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inpage yram + * false -- the registers are NOT in the inpage yram + */ +static bool check_inpage_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1) + in = check_inpage_yram_bk1(cd, page, reg, len); + else if (book == TAS2781_YRAM_BOOK2 && page == TAS2781_YRAM5_PAGE) + in = check_inpage_yram_rg(cd, reg, len); + + return in; +} + +static bool check_inblock_yram_bk(struct tas_crc *cd, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if ((page >= TAS2781_YRAM4_START_PAGE && + page <= TAS2781_YRAM4_END_PAGE) || + (page >= TAS2781_YRAM2_START_PAGE && + page <= TAS2781_YRAM2_END_PAGE)) { + if (reg <= TAS2781_YRAM2_END_REG && + reg >= TAS2781_YRAM2_START_REG) { + cd->offset = reg; + cd->len = len; + in = true; + } else if (reg < TAS2781_YRAM2_START_REG) { + if (reg + len - 1 >= TAS2781_YRAM2_START_REG) { + cd->offset = TAS2781_YRAM2_START_REG; + cd->len = reg + len - TAS2781_YRAM2_START_REG; + in = true; + } + } + } + + return in; +} + +/* + * Return Code: + * true -- the registers are in the inblock yram + * false -- the registers are NOT in the inblock yram + */ +static bool check_inblock_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in = false; + + if (book == TAS2781_YRAM_BOOK1 || book == TAS2781_YRAM_BOOK2) + in = check_inblock_yram_bk(cd, page, reg, len); + + return in; +} + +static bool check_yram(struct tas_crc *cd, unsigned char book, + unsigned char page, unsigned char reg, unsigned char len) +{ + bool in; + + in = check_inpage_yram(cd, book, page, reg, len); + if (!in) + in = check_inblock_yram(cd, book, page, reg, len); + + return in; +} + +static int tasdev_multibytes_chksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned int len) +{ + struct tas_crc crc_data; + unsigned char crc_chksum = 0; + unsigned char nBuf1[128]; + int ret = 0, i; + bool in; + + if ((reg + len - 1) > 127) { + ret = -EINVAL; + dev_err(tasdevice->dev, "firmware error\n"); + goto end; + } + + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg == TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (len == 4)) { + /* DSP swap command, pass */ + ret = 0; + goto end; + } + + in = check_yram(&crc_data, book, page, reg, len); + if (!in) + goto end; + + if (len == 1) { + dev_err(tasdevice->dev, "firmware error\n"); + ret = -EINVAL; + goto end; + } + + ret = tasdevice_spi_dev_bulk_read(tasdevice, + TASDEVICE_REG(book, page, crc_data.offset), + nBuf1, crc_data.len); + if (ret < 0) + goto end; + + for (i = 0; i < crc_data.len; i++) { + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) >= + TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + ((i + crc_data.offset) <= + (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + /* DSP swap command, bypass */ + continue; + else + crc_chksum += crc8(tasdevice->crc8_lkp_tbl, &nBuf1[i], + 1, 0); + } + + ret = crc_chksum; + +end: + return ret; +} + +static int do_singlereg_checksum(struct tasdevice_priv *tasdevice, + unsigned char book, unsigned char page, + unsigned char reg, unsigned char val) +{ + struct tas_crc crc_data; + unsigned int nData1; + int ret = 0; + bool in; + + /* DSP swap command, pass */ + if ((book == TASDEVICE_BOOK_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (page == TASDEVICE_PAGE_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg >= TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG)) && + (reg <= (TASDEVICE_REG_ID(TAS2781_SA_COEFF_SWAP_REG) + 4))) + return 0; + + in = check_yram(&crc_data, book, page, reg, 1); + if (!in) + return 0; + ret = tasdevice_spi_dev_read(tasdevice, + TASDEVICE_REG(book, page, reg), &nData1); + if (ret < 0) + return ret; + + if (nData1 != val) { + dev_err(tasdevice->dev, + "B[0x%x]P[0x%x]R[0x%x] W[0x%x], R[0x%x]\n", + book, page, reg, val, nData1); + tasdevice->err_code |= ERROR_YRAM_CRCCHK; + return -EAGAIN; + } + + ret = crc8(tasdevice->crc8_lkp_tbl, &val, 1, 0); + + return ret; +} + +static void set_err_prg_cfg(unsigned int type, struct tasdevice_priv *p) +{ + if ((type == MAIN_ALL_DEVICES) || (type == MAIN_DEVICE_A) || + (type == MAIN_DEVICE_B) || (type == MAIN_DEVICE_C) || + (type == MAIN_DEVICE_D)) + p->cur_prog = -1; + else + p->cur_conf = -1; +} + +static int tasdev_bytes_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned int len, + unsigned char val, unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) + ret = tasdev_multibytes_chksum(tas_priv, book, page, reg, + len); + else + ret = do_singlereg_checksum(tas_priv, book, page, reg, val); + + if (ret > 0) { + *crc_chksum += ret; + goto end; + } + + if (ret != -EAGAIN) + goto end; + + block->nr_retry--; + if (block->nr_retry > 0) + goto end; + + set_err_prg_cfg(block->type, tas_priv); + +end: + return ret; +} + +static int tasdev_multibytes_wr(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block, unsigned char book, + unsigned char page, unsigned char reg, unsigned char *data, + unsigned int len, unsigned int *nr_cmds, + unsigned char *crc_chksum) +{ + int ret; + + if (len > 1) { + ret = tasdevice_spi_dev_bulk_write(tas_priv, + TASDEVICE_REG(book, page, reg), data + 3, len); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, + book, page, reg, len, 0, crc_chksum); + } else { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, reg), data[3]); + if (ret < 0) + return ret; + if (block->is_ychksum_present) + ret = tasdev_bytes_chksum(tas_priv, block, book, + page, reg, 1, data[3], crc_chksum); + } + + if (!block->is_ychksum_present || ret >= 0) { + *nr_cmds += 1; + if (len >= 2) + *nr_cmds += ((len - 2) / 4) + 1; + } + + return ret; +} + +static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int nr_value; + int ret; + + ret = tasdevice_spi_dev_read(tas_priv, TASDEVICE_CHECKSUM, &nr_value); + if (ret < 0) { + dev_err(tas_priv->dev, "%s: read error %d.\n", __func__, ret); + set_err_prg_cfg(block->type, tas_priv); + return ret; + } + + if ((nr_value & 0xff) != block->pchksum) { + dev_err(tas_priv->dev, "%s: PChkSum err %d ", __func__, ret); + dev_err(tas_priv->dev, "PChkSum = 0x%x, Reg = 0x%x\n", + block->pchksum, (nr_value & 0xff)); + tas_priv->err_code |= ERROR_PRAM_CRCCHK; + ret = -EAGAIN; + block->nr_retry--; + + if (block->nr_retry <= 0) + set_err_prg_cfg(block->type, tas_priv); + } else { + tas_priv->err_code &= ~ERROR_PRAM_CRCCHK; + } + + return ret; +} + +static int tasdev_load_blk(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + unsigned int sleep_time, len, nr_cmds; + unsigned char offset, book, page, val; + unsigned char *data = block->data; + unsigned char crc_chksum = 0; + int ret = 0; + + while (block->nr_retry > 0) { + if (block->is_pchksum_present) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_CHECKSUM, 0); + if (ret < 0) + break; + } + + if (block->is_ychksum_present) + crc_chksum = 0; + + nr_cmds = 0; + + while (nr_cmds < block->nr_cmds) { + data = block->data + nr_cmds * 4; + + book = data[0]; + page = data[1]; + offset = data[2]; + val = data[3]; + + nr_cmds++; + /* Single byte write */ + if (offset <= 0x7F) { + ret = tasdevice_spi_dev_write(tas_priv, + TASDEVICE_REG(book, page, offset), + val); + if (ret < 0) + break; + if (block->is_ychksum_present) { + ret = tasdev_bytes_chksum(tas_priv, + block, book, page, offset, + 1, val, &crc_chksum); + if (ret < 0) + break; + } + continue; + } + /* sleep command */ + if (offset == 0x81) { + /* book -- data[0] page -- data[1] */ + sleep_time = ((book << 8) + page)*1000; + fsleep(sleep_time); + continue; + } + /* Multiple bytes write */ + if (offset == 0x85) { + data += 4; + len = (book << 8) + page; + book = data[0]; + page = data[1]; + offset = data[2]; + ret = tasdev_multibytes_wr(tas_priv, + block, book, page, offset, data, + len, &nr_cmds, &crc_chksum); + if (ret < 0) + break; + } + } + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + + if (block->is_pchksum_present) { + ret = tasdev_block_chksum(tas_priv, block); + if (ret == -EAGAIN) { + if (block->nr_retry > 0) + continue; + } else if (ret < 0) { + /* err in current device, skip it */ + break; + } + } + + if (block->is_ychksum_present) { + /* TBD, open it when FW ready */ + dev_err(tas_priv->dev, + "Blk YChkSum: FW = 0x%x, YCRC = 0x%x\n", + block->ychksum, crc_chksum); + + tas_priv->err_code &= + ~ERROR_YRAM_CRCCHK; + ret = 0; + } + /* skip current blk */ + break; + } + + return ret; +} + +static int tasdevice_load_block(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block) +{ + int ret = 0; + + block->nr_retry = 6; + if (tas_priv->is_loading == false) + return 0; + ret = tasdev_load_blk(tas_priv, block); + if (ret < 0) + dev_err(tas_priv->dev, "Blk (%d) load error\n", block->type); + + return ret; +} + +static int dspfw_default_callback(struct tasdevice_priv *tas_priv, + unsigned int drv_ver, unsigned int ppcver) +{ + int rc = 0; + + if (drv_ver == 0x100) { + if (ppcver >= PPC3_VERSION) { + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + } else { + switch (ppcver) { + case 0x00: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + dev_err(tas_priv->dev, + "%s: PPCVer must be 0x0 or 0x%02x", + __func__, PPC3_VERSION); + dev_err(tas_priv->dev, " Current:0x%02x\n", + ppcver); + rc = -EINVAL; + break; + } + } + } else { + dev_err(tas_priv->dev, + "DrvVer must be 0x0, 0x230 or above 0x230 "); + dev_err(tas_priv->dev, "current is 0x%02x\n", drv_ver); + rc = -EINVAL; + } + + return rc; +} + +static int load_calib_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +static int fw_parse_header(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct tasdevice_dspfw_hdr *fw_hdr = &(tas_fmw->fw_hdr); + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr = &(fw_hdr->fixed_hdr); + static const unsigned char magic_number[] = {0x35, 0x35, 0x35, 0x32, }; + const unsigned char *buf = (unsigned char *)fmw->data; + + if (offset + 92 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -EINVAL; + goto out; + } + if (memcmp(&buf[offset], magic_number, 4)) { + dev_err(tas_priv->dev, "%s: Magic num NOT match\n", __func__); + offset = -EINVAL; + goto out; + } + offset += 4; + + /* + * Convert data[offset], data[offset + 1], data[offset + 2] and + * data[offset + 3] into host + */ + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_fixed_hdr->fwsize != fmw->size) { + dev_err(tas_priv->dev, "File size not match, %lu %u", + (unsigned long)fmw->size, fw_fixed_hdr->fwsize); + offset = -EINVAL; + goto out; + } + offset += 4; + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); + offset += 72; + +out: + return offset; +} + +static int tasdevice_dspfw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw_fixed_hdr *fw_fixed_hdr; + struct tasdevice_fw *tas_fmw; + int offset = 0, ret = 0; + + if (!fmw || !fmw->data) { + dev_err(tas_priv->dev, "%s: Failed to read firmware %s\n", + __func__, tas_priv->coef_binaryname); + return -EINVAL; + } + + tas_priv->fmw = kzalloc(sizeof(*tas_priv->fmw), GFP_KERNEL); + if (!tas_priv->fmw) + return -ENOMEM; + tas_fmw = tas_priv->fmw; + tas_fmw->dev = tas_priv->dev; + offset = fw_parse_header(tas_priv, tas_fmw, fmw, offset); + + if (offset == -EINVAL) + return -EINVAL; + + fw_fixed_hdr = &(tas_fmw->fw_hdr.fixed_hdr); + /* Support different versions of firmware */ + switch (fw_fixed_hdr->drv_ver) { + case 0x301: + case 0x302: + case 0x502: + case 0x503: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_kernel; + tas_priv->fw_parse_program_data = + fw_parse_program_data_kernel; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data_kernel; + tas_priv->tasdevice_load_block = + tasdevice_load_block_kernel; + break; + case 0x202: + case 0x400: + tas_priv->fw_parse_variable_header = + fw_parse_variable_header_git; + tas_priv->fw_parse_program_data = + fw_parse_program_data; + tas_priv->fw_parse_configuration_data = + fw_parse_configuration_data; + tas_priv->tasdevice_load_block = + tasdevice_load_block; + break; + default: + ret = dspfw_default_callback(tas_priv, + fw_fixed_hdr->drv_ver, fw_fixed_hdr->ppcver); + if (ret) + return ret; + break; + } + + offset = tas_priv->fw_parse_variable_header(tas_priv, fmw, offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_program_data(tas_priv, tas_fmw, fmw, + offset); + if (offset < 0) + return offset; + + offset = tas_priv->fw_parse_configuration_data(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + + return ret; +} + +int tasdevice_spi_dsp_parser(void *context) +{ + struct tasdevice_priv *tas_priv = context; + const struct firmware *fw_entry; + int ret; + + ret = request_firmware(&fw_entry, tas_priv->coef_binaryname, + tas_priv->dev); + if (ret) { + dev_err(tas_priv->dev, "%s: load %s error\n", __func__, + tas_priv->coef_binaryname); + return ret; + } + + ret = tasdevice_dspfw_ready(fw_entry, tas_priv); + release_firmware(fw_entry); + fw_entry = NULL; + + return ret; +} + +static void tasdev_dsp_prog_blk_remove(struct tasdevice_prog *prog) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (!prog) + return; + + tas_dt = &(prog->dev_data); + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &(tas_dt->dev_blks[i]); + kfree(blk->data); + } + kfree(tas_dt->dev_blks); +} + +static void tasdev_dsp_prog_remove(struct tasdevice_prog *prog, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_prog_blk_remove(&prog[i]); + kfree(prog); +} + +static void tasdev_dsp_cfg_blk_remove(struct tasdevice_config *cfg) +{ + struct tasdevice_data *tas_dt; + struct tasdev_blk *blk; + unsigned int i; + + if (cfg) { + tas_dt = &(cfg->dev_data); + + if (!tas_dt->dev_blks) + return; + + for (i = 0; i < tas_dt->nr_blk; i++) { + blk = &(tas_dt->dev_blks[i]); + kfree(blk->data); + } + kfree(tas_dt->dev_blks); + } +} + +static void tasdev_dsp_cfg_remove(struct tasdevice_config *config, + unsigned short nr) +{ + int i; + + for (i = 0; i < nr; i++) + tasdev_dsp_cfg_blk_remove(&config[i]); + kfree(config); +} + +void tasdevice_spi_dsp_remove(void *context) +{ + struct tasdevice_priv *tas_dev = context; + + if (!tas_dev->fmw) + return; + + if (tas_dev->fmw->programs) + tasdev_dsp_prog_remove(tas_dev->fmw->programs, + tas_dev->fmw->nr_programs); + if (tas_dev->fmw->configs) + tasdev_dsp_cfg_remove(tas_dev->fmw->configs, + tas_dev->fmw->nr_configurations); + kfree(tas_dev->fmw); + tas_dev->fmw = NULL; +} + +static void tas2781_clear_calfirmware(struct tasdevice_fw *tas_fmw) +{ + struct tasdevice_calibration *calibration; + struct tasdev_blk *block; + unsigned int blks; + int i; + + if (!tas_fmw->calibrations) + goto out; + + for (i = 0; i < tas_fmw->nr_calibrations; i++) { + calibration = &(tas_fmw->calibrations[i]); + if (!calibration) + continue; + + if (!calibration->dev_data.dev_blks) + continue; + + for (blks = 0; blks < calibration->dev_data.nr_blk; blks++) { + block = &(calibration->dev_data.dev_blks[blks]); + if (!block) + continue; + kfree(block->data); + } + kfree(calibration->dev_data.dev_blks); + } + kfree(tas_fmw->calibrations); +out: + kfree(tas_fmw); +} + +void tasdevice_spi_calbin_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + + if (tas_priv->cali_data_fmw) { + tas2781_clear_calfirmware(tas_priv->cali_data_fmw); + tas_priv->cali_data_fmw = NULL; + } +} + +void tasdevice_spi_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **ci = rca->cfg_info; + unsigned int i, j; + + if (!ci) + return; + for (i = 0; i < rca->ncfgs; i++) { + if (!ci[i]) + continue; + if (ci[i]->blk_data) { + for (j = 0; j < ci[i]->real_nblocks; j++) { + if (!ci[i]->blk_data[j]) + continue; + kfree(ci[i]->blk_data[j]->regdata); + kfree(ci[i]->blk_data[j]); + } + kfree(ci[i]->blk_data); + } + kfree(ci[i]); + } + kfree(ci); +} + +static int tasdevice_load_data(struct tasdevice_priv *tas_priv, + struct tasdevice_data *dev_data) +{ + struct tasdev_blk *block; + unsigned int i; + int ret = 0; + + for (i = 0; i < dev_data->nr_blk; i++) { + block = &(dev_data->dev_blks[i]); + ret = tas_priv->tasdevice_load_block(tas_priv, block); + if (ret < 0) + break; + } + + return ret; +} + +int tasdevice_spi_select_tuningprm_cfg(void *context, int prm_no, + int cfg_no, int rca_conf_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_rca *rca = &(tas_priv->rcabin); + struct tasdevice_config_info **cfg_info = rca->cfg_info; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + struct tasdevice_config *conf; + int prog_status = 0; + int status; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (cfg_no >= tas_fmw->nr_configurations) { + dev_err(tas_priv->dev, + "%s: cfg(%d) is not in range of conf %u\n", + __func__, cfg_no, tas_fmw->nr_configurations); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (rca_conf_no >= rca->ncfgs || + rca_conf_no < 0 || + !cfg_info) { + dev_err(tas_priv->dev, + "conf_no:%d should be in range from 0 to %u\n", + rca_conf_no, rca->ncfgs-1); + return 0; + } + + if (prm_no >= 0 && + (tas_priv->cur_prog != prm_no || + tas_priv->force_fwload_status)) { + tas_priv->cur_conf = -1; + tas_priv->is_loading = true; + prog_status++; + } + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration + *cal = cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->cur_prog = prm_no; + } + + } + + if (cfg_no && + (cfg_no != tas_priv->cur_conf) && + (tas_priv->is_loaderr == false)) { + status++; + tas_priv->is_loading = true; + } else { + tas_priv->is_loading = false; + } + + if (status) { + conf = &(tas_fmw->configs[cfg_no]); + status = 0; + tasdevice_load_data(tas_priv, &(conf->dev_data)); + if (tas_priv->is_loaderr == true) { + status |= 1 << 4; + } else if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + tas_priv->cur_conf = cfg_no; + } + } else { + dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", + __func__, cfg_no); + } + + return prog_status; +} + +int tasdevice_spi_prmg_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (prm_no >= 0 && tas_priv->cur_prog != prm_no) { + tas_priv->cur_conf = -1; + tas_priv->is_loading = true; + prog_status++; + } + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + if ((tas_priv->is_loaderr == false) && + (tas_priv->is_loading == true)) + tas_priv->cur_prog = prm_no; + } + + return prog_status; +} + +int tasdevice_spi_prmg_calibdata_load(void *context, int prm_no) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct tasdevice_prog *program; + int prog_status = 0; + + if (!tas_fmw) { + dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__); + return 0; + } + + if (prm_no >= tas_fmw->nr_programs) { + dev_err(tas_priv->dev, + "%s: prm(%d) is not in range of Programs %u\n", + __func__, prm_no, tas_fmw->nr_programs); + return 0; + } + + if (prm_no >= 0 && tas_priv->cur_prog != prm_no) { + tas_priv->cur_conf = -1; + tas_priv->is_loading = true; + prog_status++; + } + tas_priv->is_loaderr = false; + + if (prog_status) { + program = &(tas_fmw->programs[prm_no]); + tasdevice_load_data(tas_priv, &(program->dev_data)); + if (tas_priv->is_loaderr == false && + tas_priv->is_loading == true) { + struct tasdevice_fw *cal_fmw = + tas_priv->cali_data_fmw; + + if (cal_fmw) { + struct tasdevice_calibration *cal = + cal_fmw->calibrations; + + if (cal) + load_calib_data(tas_priv, + &(cal->dev_data)); + } + tas_priv->cur_prog = prm_no; + } + + } + + return prog_status; +} + +void tasdevice_spi_tuning_switch(void *context, int state) +{ + struct tasdevice_priv *tas_priv = context; + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + int profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + dev_err(tas_priv->dev, "DSP bin file not loaded\n"); + return; + } + + if (state == 0) { + if (tas_priv->cur_prog < tas_fmw->nr_programs) { + /* dsp mode or tuning mode */ + profile_cfg_id = tas_priv->rcabin.profile_cfg_id; + tasdevice_spi_select_tuningprm_cfg(tas_priv, + tas_priv->cur_prog, tas_priv->cur_conf, + profile_cfg_id); + } + + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_POWER_UP); + } else { + tasdevice_spi_select_cfg_blk(tas_priv, profile_cfg_id, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN); + } +} -- 2.40.1

6 6

[PATCH v2 0/2] ASoC: dt-bindings: convert qcom sound bindings to yaml
by Rayyan Ansari 11 Jul '24

11 Jul '24

Hi, These patches convert the remaining plain text bindings for Qualcomm sound drivers to dt schema, so device trees can be validated against them. v1: https://lore.kernel.org/all/20240709152808.155405-1-rayyan.ansari@linaro.or… Thanks, Rayyan Rayyan Ansari (2): ASoC: dt-bindings: qcom,msm8916-wcd-digital-codec: convert to dtschema ASoC: dt-bindings: qcom,apq8096-sndcard: use dtschema .../bindings/sound/qcom,apq8096.txt | 128 ------------------ .../sound/qcom,msm8916-wcd-digital-codec.yaml | 55 ++++++++ .../sound/qcom,msm8916-wcd-digital.txt | 20 --- .../bindings/sound/qcom,sm8250.yaml | 1 + 4 files changed, 56 insertions(+), 148 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt -- 2.45.2

3 4

[PATCH v2 0/4] firmware: cs_dsp: Some small coding improvements
by Richard Fitzgerald 11 Jul '24

11 Jul '24

Commit series that makes some small improvements to code and the kernel log messages. --- Changes since V1: - Replaced commit #3 with one that keeps the wmfw format version message on its own line instead of merging it into the INFO text message. - Commit #4 rebased onto the new #3 Richard Fitzgerald (4): firmware: cs_dsp: Don't allocate temporary buffer for info text firmware: cs_dsp: Make wmfw and bin filename arguments const char * firmware: cs_dsp: Clarify wmfw format version log message firmware: cs_dsp: Rename fw_ver to wmfw_ver drivers/firmware/cirrus/cs_dsp.c | 68 +++++++++----------------- include/linux/firmware/cirrus/cs_dsp.h | 10 ++-- sound/soc/codecs/wm_adsp.c | 2 +- 3 files changed, 29 insertions(+), 51 deletions(-) -- 2.39.2

3 6

[PATCH v2] ASoC: dt-bindings: cirrus,cs4270: Convert to dtschema
by Animesh Agarwal 11 Jul '24

11 Jul '24

Convert the Cirrus Logic CS4270 audio CODEC bindings to DT schema. Add missing va-supply, vd-supply and vlc-supply properties, because they are already being used in the DTS and the driver for this device. Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- Changes in v2: - Added proper description for power supply properties. - Added ref to dai-common. - Added '#sound-dai-cells'. - Dropped unused labels in example. - Replaced example node name to a generic name. --- .../bindings/sound/cirrus,cs4270.yaml | 59 +++++++++++++++++++ .../devicetree/bindings/sound/cs4270.txt | 21 ------- 2 files changed, 59 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml new file mode 100644 index 000000000000..336e11773694 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml @@ -0,0 +1,59 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4270.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS4270 audio CODEC + +maintainers: + - patches(a)opensource.cirrus.com + +description: + The CS4270 is a stereo audio codec. The driver for this device currently only + supports I2C. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: cirrus,cs4270 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + reset-gpios: + description: + This pin will be deasserted before communication to the codec starts. + maxItems: 1 + + va-supply: + description: Analog power supply. + + vd-supply: + description: Digital power supply. + + vlc-supply: + description: Serial Control Port power supply. + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt deleted file mode 100644 index c33770ec4c3c..000000000000 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ /dev/null @@ -1,21 +0,0 @@ -CS4270 audio CODEC - -The driver for this device currently only supports I2C. - -Required properties: - - - compatible : "cirrus,cs4270" - - - reg : the I2C address of the device for I2C - -Optional properties: - - - reset-gpios : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. - -Example: - -codec: cs4270@48 { - compatible = "cirrus,cs4270"; - reg = <0x48>; -}; -- 2.45.2

1 0

[PATCH 1/2] ASoC: codecs: aw88395: Simplify with cleanup.h
by Krzysztof Kozlowski 10 Jul '24

10 Jul '24

Allocate memory, which is being freed at end of the scope, with scoped/cleanup.h to reduce number of error paths and make code a bit simpler. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/aw88395/aw88395_lib.c | 51 +++++++++----------------- 1 file changed, 17 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index f25f6e0d4428..769ca32a5c8e 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -7,6 +7,7 @@ // Author: Bruce zhao <zhaolei(a)awinic.com> // +#include <linux/cleanup.h> #include <linux/crc8.h> #include <linux/i2c.h> #include "aw88395_lib.h" @@ -361,11 +362,11 @@ static int aw_dev_parse_raw_dsp_fw(unsigned char *data, unsigned int data_len, static int aw_dev_prof_parse_multi_bin(struct aw_device *aw_dev, unsigned char *data, unsigned int data_len, struct aw_prof_desc *prof_desc) { - struct aw_bin *aw_bin; int ret; int i; - aw_bin = devm_kzalloc(aw_dev->dev, data_len + sizeof(struct aw_bin), GFP_KERNEL); + struct aw_bin *aw_bin __free(kfree) = kzalloc(data_len + sizeof(struct aw_bin), + GFP_KERNEL); if (!aw_bin) return -ENOMEM; @@ -375,7 +376,7 @@ static int aw_dev_prof_parse_multi_bin(struct aw_device *aw_dev, unsigned char * ret = aw_parsing_bin_file(aw_dev, aw_bin); if (ret < 0) { dev_err(aw_dev->dev, "parse bin failed"); - goto parse_bin_failed; + return ret; } for (i = 0; i < aw_bin->all_bin_parse_num; i++) { @@ -387,10 +388,8 @@ static int aw_dev_prof_parse_multi_bin(struct aw_device *aw_dev, unsigned char * data + aw_bin->header_info[i].valid_data_addr; break; case DATA_TYPE_DSP_REG: - if (aw_bin->header_info[i].valid_data_len & 0x01) { - ret = -EINVAL; - goto parse_bin_failed; - } + if (aw_bin->header_info[i].valid_data_len & 0x01) + return -EINVAL; swab16_array((u16 *)(data + aw_bin->header_info[i].valid_data_addr), aw_bin->header_info[i].valid_data_len >> 1); @@ -402,10 +401,8 @@ static int aw_dev_prof_parse_multi_bin(struct aw_device *aw_dev, unsigned char * break; case DATA_TYPE_DSP_FW: case DATA_TYPE_SOC_APP: - if (aw_bin->header_info[i].valid_data_len & 0x01) { - ret = -EINVAL; - goto parse_bin_failed; - } + if (aw_bin->header_info[i].valid_data_len & 0x01) + return -EINVAL; swab16_array((u16 *)(data + aw_bin->header_info[i].valid_data_addr), aw_bin->header_info[i].valid_data_len >> 1); @@ -422,20 +419,17 @@ static int aw_dev_prof_parse_multi_bin(struct aw_device *aw_dev, unsigned char * } } prof_desc->prof_st = AW88395_PROFILE_OK; - ret = 0; -parse_bin_failed: - devm_kfree(aw_dev->dev, aw_bin); - return ret; + return 0; } static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, uint8_t *data, uint32_t data_len, struct aw_prof_desc *prof_desc) { - struct aw_bin *aw_bin; int ret; - aw_bin = devm_kzalloc(aw_dev->dev, data_len + sizeof(*aw_bin), GFP_KERNEL); + struct aw_bin *aw_bin __free(kfree) = kzalloc(data_len + sizeof(*aw_bin), + GFP_KERNEL); if (!aw_bin) return -ENOMEM; @@ -445,14 +439,13 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, ret = aw_parsing_bin_file(aw_dev, aw_bin); if (ret < 0) { dev_err(aw_dev->dev, "parse bin failed"); - goto parse_bin_failed; + return ret; } if ((aw_bin->all_bin_parse_num != 1) || (aw_bin->header_info[0].bin_data_type != DATA_TYPE_REGISTER)) { dev_err(aw_dev->dev, "bin num or type error"); - ret = -EINVAL; - goto parse_bin_failed; + return -EINVAL; } prof_desc->sec_desc[AW88395_DATA_TYPE_REG].data = @@ -461,15 +454,7 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, aw_bin->header_info[0].valid_data_len; prof_desc->prof_st = AW88395_PROFILE_OK; - devm_kfree(aw_dev->dev, aw_bin); - aw_bin = NULL; - return 0; - -parse_bin_failed: - devm_kfree(aw_dev->dev, aw_bin); - aw_bin = NULL; - return ret; } static int aw_dev_parse_data_by_sec_type(struct aw_device *aw_dev, struct aw_cfg_hdr *cfg_hdr, @@ -678,21 +663,21 @@ static int aw_dev_cfg_get_multiple_valid_prof(struct aw_device *aw_dev, static int aw_dev_load_cfg_by_hdr(struct aw_device *aw_dev, struct aw_cfg_hdr *prof_hdr) { - struct aw_all_prof_info *all_prof_info; int ret; - all_prof_info = devm_kzalloc(aw_dev->dev, sizeof(struct aw_all_prof_info), GFP_KERNEL); + struct aw_all_prof_info *all_prof_info __free(kfree) = kzalloc(sizeof(*all_prof_info), + GFP_KERNEL); if (!all_prof_info) return -ENOMEM; ret = aw_dev_parse_dev_type(aw_dev, prof_hdr, all_prof_info); if (ret < 0) { - goto exit; + return ret; } else if (ret == AW88395_DEV_TYPE_NONE) { dev_dbg(aw_dev->dev, "get dev type num is 0, parse default dev"); ret = aw_dev_parse_dev_default_type(aw_dev, prof_hdr, all_prof_info); if (ret < 0) - goto exit; + return ret; } switch (aw_dev->prof_data_type) { @@ -710,8 +695,6 @@ static int aw_dev_load_cfg_by_hdr(struct aw_device *aw_dev, if (!ret) aw_dev->prof_info.prof_name_list = profile_name; -exit: - devm_kfree(aw_dev->dev, all_prof_info); return ret; } -- 2.43.0

2 2

Fix labeling of BOSS Katana ports
by GitHub pull_request - edited 10 Jul '24

10 Jul '24

alsa-project/alsa-ucm-conf pull request #430 was edited from aledomu: I'd like to rename the MIDI ports as well but I don't know how. If you can point me where I can learn about it, I'll be glad to do so and mark this PR as finished. I guess this fix works the same for all Katana models from all generations, but I don't have the PID available to enable it, nor the devices for testing. Edit: Didn't post the [`alsa-info.sh` output](https://gist.github.com/aledomu/ac3b2741ca2426bc6fbc057907ad0dab). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] firmware: cs_dsp: Some small coding improvements
by Richard Fitzgerald 10 Jul '24

10 Jul '24

Commit series that makes some small improvements to code and the kernel log messages. Richard Fitzgerald (4): firmware: cs_dsp: Don't allocate temporary buffer for info text firmware: cs_dsp: Make wmfw and bin filename arguments const char * firmware: cs_dsp: Merge wmfw format log message into INFO message firmware: cs_dsp: Rename fw_ver to wmfw_ver drivers/firmware/cirrus/cs_dsp.c | 64 +++++++++----------------- include/linux/firmware/cirrus/cs_dsp.h | 10 ++-- sound/soc/codecs/wm_adsp.c | 2 +- 3 files changed, 27 insertions(+), 49 deletions(-) -- 2.39.2

2 6

[PATCH 0/2] ASoC: dt-bindings: convert qcom sound bindings to yaml
by Rayyan Ansari 10 Jul '24

10 Jul '24

Hi, These patches convert the remaining plain text bindings for Qualcomm sound drivers to dt schema, so device trees can be validated against them. Thanks, Rayyan Rayyan Ansari (2): ASoC: dt-bindings: qcom,msm8916-wcd-digital-codec: convert to dtschema ASoC: dt-bindings: qcom,apq8096-sndcard: use dtschema .../bindings/sound/qcom,apq8096.txt | 128 ------------------ .../sound/qcom,msm8916-wcd-digital-codec.yaml | 53 ++++++++ .../sound/qcom,msm8916-wcd-digital.txt | 20 --- .../bindings/sound/qcom,sm8250.yaml | 1 + 4 files changed, 54 insertions(+), 148 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-digital.txt -- 2.45.2

2 4

[PATCH 0/9] soundwire: simplify code with cleanup.h
by Krzysztof Kozlowski 10 Jul '24

10 Jul '24

Allocate the memory with scoped/cleanup.h to reduce error handling (simpler error paths) and make the code a bit smaller. Best regards, Krzysztof --- Krzysztof Kozlowski (9): soundwire: amd: simplify return path in hw_params soundwire: amd: simplify with cleanup.h soundwire: amd_init: simplify with cleanup.h soundwire: intel: simplify return path in hw_params soundwire: intel: simplify with cleanup.h soundwire: intel_ace2x: simplify return path in hw_params soundwire: intel_ace2x: simplify with cleanup.h soundwire: cadence: simplify with cleanup.h soundwire: debugfs: simplify with cleanup.h drivers/soundwire/amd_init.c | 7 +++---- drivers/soundwire/amd_manager.c | 13 +++++-------- drivers/soundwire/cadence_master.c | 5 ++--- drivers/soundwire/debugfs.c | 7 ++----- drivers/soundwire/intel.c | 25 +++++++++---------------- drivers/soundwire/intel_ace2x.c | 22 ++++++++-------------- 6 files changed, 29 insertions(+), 50 deletions(-) --- base-commit: 246cf92bafedeea46cb16964d16478d6b68f8e5a change-id: 20240703-soundwire-cleanup-h-a29941fb3dd4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 10

I have the MPG Z690 Edge Wifi DDR4 motherboard, and no sound in front panel or back panel
by GitHub issues - opened 10 Jul '24

10 Jul '24

alsa-project/alsa-ucm-conf issue #432 was opened from angelbearuk: I'm using Linux Mint 21, and am very frustrated. Not being able to hear any sound just kills the Linux experience for me. I had no issues with sound on Windows 10. Please help me. I'm a Linux newbie. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/432 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoc: TAS2781: rename the tas2781_reset as tasdevice_reset
by Shenghao Ding 09 Jul '24

09 Jul '24

Rename the tas2781_reset as tasdevice_reset in case of misunderstanding. RESET register for both tas2563 and tas2781 is same and the use of reset pin is also same. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Rename tas2781_reset as tasdevice_reset - Changed the copyright year to 2024 in tas2781.h. - Rename TAS2781_REG_SWRESET_XXX as TASDEVICE_REG_SWRESET_XXX - Add tasdevice_reset into tasdevice_i2c_probe. --- include/sound/tas2781.h | 8 ++++---- sound/pci/hda/tas2781_hda_i2c.c | 4 ++-- sound/soc/codecs/tas2781-comlib.c | 8 ++++---- sound/soc/codecs/tas2781-i2c.c | 2 ++ 4 files changed, 12 insertions(+), 10 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index cd8ce522b78e..a43ad6dcb7c7 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -43,8 +43,8 @@ (page * 128)) + reg) /*Software Reset */ -#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) -#define TAS2781_REG_SWRESET_RESET BIT(0) +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TASDEVICE_REG_SWRESET_RESET BIT(0) /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) @@ -140,7 +140,7 @@ struct tasdevice_priv { void (*apply_calibration)(struct tasdevice_priv *tas_priv); }; -void tas2781_reset(struct tasdevice_priv *tas_dev); +void tasdevice_reset(struct tasdevice_priv *tas_dev); int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, struct module *module, void (*cont)(const struct firmware *fw, void *context)); diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 75f7674c66ee..10420360b214 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -834,7 +834,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) pm_runtime_set_active(tas_hda->dev); pm_runtime_enable(tas_hda->dev); - tas2781_reset(tas_hda->priv); + tasdevice_reset(tas_hda->priv); ret = component_add(tas_hda->dev, &tas2781_hda_comp_ops); if (ret) { @@ -929,7 +929,7 @@ static int tas2781_system_resume(struct device *dev) tas_hda->priv->tasdevice[i].cur_prog = -1; tas_hda->priv->tasdevice[i].cur_conf = -1; } - tas2781_reset(tas_hda->priv); + tasdevice_reset(tas_hda->priv); tasdevice_prmg_load(tas_hda->priv, tas_hda->priv->cur_prog); /* If calibrated data occurs error, dsp will still work with default diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 6db1a260da82..1fbf4560f5cc 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -243,7 +243,7 @@ struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c) } EXPORT_SYMBOL_GPL(tasdevice_kzalloc); -void tas2781_reset(struct tasdevice_priv *tas_dev) +void tasdevice_reset(struct tasdevice_priv *tas_dev) { int ret, i; @@ -254,8 +254,8 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) } else { for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, - TAS2781_REG_SWRESET, - TAS2781_REG_SWRESET_RESET); + TASDEVICE_REG_SWRESET, + TASDEVICE_REG_SWRESET_RESET); if (ret < 0) dev_err(tas_dev->dev, "dev %d swreset fail, %d\n", @@ -264,7 +264,7 @@ void tas2781_reset(struct tasdevice_priv *tas_dev) } usleep_range(1000, 1050); } -EXPORT_SYMBOL_GPL(tas2781_reset); +EXPORT_SYMBOL_GPL(tasdevice_reset); int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, struct module *module, diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index b412f8f2e5d5..d5d95ae57c73 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -742,6 +742,8 @@ static int tasdevice_i2c_probe(struct i2c_client *i2c) if (ret) goto err; + tasdevice_reset(tas_priv); + ret = devm_snd_soc_register_component(tas_priv->dev, &soc_codec_driver_tasdevice, tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); -- 2.34.1

2 1

[PATCH] ASoC: Intel: sof_sdw: Convert comma to semicolon
by Chen Ni 09 Jul '24

09 Jul '24

Replace a comma between expression statements by a semicolon. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/boards/sof_sdw.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e94849b84a6b..e5feaef669d1 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -2136,9 +2136,9 @@ static int mc_probe(struct platform_device *pdev) card = &ctx->card; card->dev = &pdev->dev; - card->name = "soundwire", - card->owner = THIS_MODULE, - card->late_probe = sof_sdw_card_late_probe, + card->name = "soundwire"; + card->owner = THIS_MODULE; + card->late_probe = sof_sdw_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: cirrus,cs4270: Convert to dtschema
by Animesh Agarwal 09 Jul '24

09 Jul '24

Convert the Cirrus Logic CS4270 audio CODEC bindings to DT schema. Add missing va-supply, vd-supply and vlc-supply properties, because they are already being used in the DTS and the driver for this device. Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Animesh Agarwal <animeshagarwal28(a)gmail.com> --- .../bindings/sound/cirrus,cs4270.yaml | 53 +++++++++++++++++++ .../devicetree/bindings/sound/cs4270.txt | 21 -------- 2 files changed, 53 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4270.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml new file mode 100644 index 000000000000..824d799ee5ff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4270.yaml @@ -0,0 +1,53 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4270.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS4270 audio CODEC + +maintainers: + - patches(a)opensource.cirrus.com + +description: + The CS4270 is a stereo audio codec. The driver for this device currently only + supports I2C. + +properties: + compatible: + const: cirrus,cs4270 + + reg: + maxItems: 1 + + reset-gpios: + description: + This pin will be deasserted before communication to the codec starts. + maxItems: 1 + + va-supply: + description: Voltage regulator phandle for the VA supply. + + vd-supply: + description: Voltage regulator phandle for the VD supply. + + vlc-supply: + description: Voltage regulator phandle for the VLC supply. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: cs4270@48 { + compatible = "cirrus,cs4270"; + reg = <0x48>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/cs4270.txt b/Documentation/devicetree/bindings/sound/cs4270.txt deleted file mode 100644 index c33770ec4c3c..000000000000 --- a/Documentation/devicetree/bindings/sound/cs4270.txt +++ /dev/null @@ -1,21 +0,0 @@ -CS4270 audio CODEC - -The driver for this device currently only supports I2C. - -Required properties: - - - compatible : "cirrus,cs4270" - - - reg : the I2C address of the device for I2C - -Optional properties: - - - reset-gpios : a GPIO spec for the reset pin. If specified, it will be - deasserted before communication to the codec starts. - -Example: - -codec: cs4270@48 { - compatible = "cirrus,cs4270"; - reg = <0x48>; -}; -- 2.45.2

2 2

[PATCH AUTOSEL 4.19 2/7] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 19f425eb4a40f..16e2ab2903754 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -474,6 +474,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 5.4 04/11] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index def2a0ce88863..e4f661c5349ef 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -353,11 +353,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 5.4 03/11] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 0541071f454bd..9a182e4c7d572 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1419,10 +1419,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1437,6 +1438,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1536,9 +1538,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 5.4 02/11] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c740dec00f83b..057ecfe2c8b5c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -465,6 +465,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 5.10 06/14] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 3328c02f93c74..1dfe439d13417 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -353,11 +353,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 5.10 05/14] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a6b72ad53b434..61ea444f2018d 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1441,10 +1441,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1459,6 +1460,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1558,9 +1560,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 5.10 03/14] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 1d049685e7075..47b581d99da67 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -468,6 +468,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 06/17] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 3328c02f93c74..1dfe439d13417 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -353,11 +353,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 05/17] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 5b82329f44401..dbd30604816e5 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 03/17] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 434679afa7e1a..3d2a0e8cad9a5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -607,6 +607,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/27] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 0dc0475670ffe..554e7896e8053 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 08/27] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 4edf5b27e136b..c6ef8f92b25f1 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 06/27] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d6ef8e850412b..ff879e173d51d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 11/33] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index a3663ab065ac2..0a731b21e5a58 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 10/33] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index a5c2cca38d01a..8c8b2a2f6f862 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1474,10 +1474,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1492,6 +1493,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1591,9 +1593,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 08/33] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 651408c6f399d..5b8b21ade9cfe 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 18/40] ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
by Sasha Levin 09 Jul '24

09 Jul '24

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> [ Upstream commit 8af49868e51ed1ba117b74728af12abe1eda82e5 ] If the ASP1 DAI is hooked up by the machine driver the ASP TX mixer sources should be initialized to disconnected. There aren't currently any available products using the ASP so this doesn't affect any existing systems. The cs35l56 does not have any fixed default for the mixer source registers. When the cs35l56 boots, its firmware patches these registers to setup a system-specific routing; this is so that Windows can use generic SDCA drivers instead of needing knowledge of chip-specific registers. The setup varies between end-products, which each have customized firmware, and so the default register state varies between end-products. It can also change if the firmware on an end-product is upgraded - for example if a change was needed to the routing for Windows use-cases. It must be emphasized that the settings applied by the firmware are not internal magic tuning; they are statically implementing use-case setup that on Linux would be done via ALSA controls. The driver is currently syncing the mixer controls with whatever initial state the firmware wrote to the registers, so that they report the actual audio routing. But if the ASP DAI is hooked up this can create a powered-up DAPM graph without anything intentionally setting up a path. This can lead to parts of the audio system powering up unexpectedly. For example when cs35l56 is connected to cs42l43 using a codec-codec link, this can create a complete DAPM graph which then powers-up cs42l43. But the cs42l43 can only be clocked from its SoundWire bus so this causes a bunch of errors in the kernel log where cs42l43 is unexpectedly powered-up without a clock. If the host is taking ownership of the ASP (either directly or as a codec-to-codec link) there is no need to keep the mixer settings that the firmware wrote. The driver has ALSA controls for setting these using standard Linux mechanisms. So if the machine driver hooks up the ASP the ASP mixers are initialized to "None" (no input). This prevents unintended DAPM-graph power-ups, and means the initial state of the mixers is always going to be None. Since the initial state of the mixers can vary from system to system and potentially between firmware upgrades, no use-case manager can currently assume that cs35l56 has a known initial state. The firmware could just as easily default them to "None" as to any input source. So defaulting them to "None" in the driver is not increasing the entropy of the system. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20240613132527.46537-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l56-shared.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index fd02b621da52c..d29878af2a80d 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -214,6 +214,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = { REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5, 0x00020100), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1, 0x00000018), REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5, 0x00000018), + REG_SEQ0(CS35L56_ASP1TX1_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX2_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX3_INPUT, 0x00000000), + REG_SEQ0(CS35L56_ASP1TX4_INPUT, 0x00000000), }; /* -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 14/40] ASoC: ti: omap-hdmi: Fix too long driver name
by Sasha Levin 09 Jul '24

09 Jul '24

From: Primoz Fiser <primoz.fiser(a)norik.com> [ Upstream commit 524d3f126362b6033e92cbe107ae2158d7fbff94 ] Set driver name to "HDMI". This simplifies the code and gets rid of the following error messages: ASoC: driver name too long 'HDMI 58040000.encoder' -> 'HDMI_58040000_e' Signed-off-by: Primoz Fiser <primoz.fiser(a)norik.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20240610125847.773394-1-primoz.fiser@norik.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-hdmi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 4513b527ab970..ad8925b6481ca 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = devm_kasprintf(dev, GFP_KERNEL, - "HDMI %s", dev_name(ad->dssdev)); - if (!card->name) - return -ENOMEM; - + card->name = "HDMI"; card->owner = THIS_MODULE; card->dai_link = devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL); -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 13/40] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
by Sasha Levin 09 Jul '24

09 Jul '24

From: Jai Luthra <j-luthra(a)ti.com> [ Upstream commit c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 ] The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples. With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Signed-off-by: Jai Luthra <j-luthra(a)ti.com> Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 1e760c3155213..2b1ed91a736c9 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + u8 numevt = *((u8 *)rule->private); struct snd_interval frames; snd_interval_any(&frames); - frames.min = 64; + frames.min = numevt; frames.integer = 1; return snd_interval_refine(period_size, &frames); @@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots; + u8 *numevt; /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream]) @@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; } + numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &mcasp->txnumevt : + &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - davinci_mcasp_hw_rule_min_periodsize, NULL, + davinci_mcasp_hw_rule_min_periodsize, numevt, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); return 0; -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 11/40] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Sasha Levin 09 Jul '24

09 Jul '24

From: Thomas GENTY <tomlohave(a)gmail.com> [ Upstream commit e3209a1827646daaab744aa6a5767b1f57fb5385 ] When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240608170251.99936-1-tomlohave@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b41a1147f1c34..a64d1989e28a5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.43.0

1 0

[PATCH AUTOSEL 6.9 08/40] ASoC: Intel: avs: Fix route override
by Sasha Levin 09 Jul '24

09 Jul '24

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit fd660b1bd015e5aa9a558ee04088f2431010548d ] Instead of overriding existing memory strings that may be too short, just allocate needed memory and point the route at it. Reported-by: Jason Montleon <jmontleo(a)redhat.com> Link: https://github.com/thesofproject/avs-topology-xml/issues/22#issuecomment-21… Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20240603102818.36165-3-amadeuszx.slawinski@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/topology.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 42b42903ae9de..691d16ce95a0f 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1545,8 +1545,8 @@ static int avs_route_load(struct snd_soc_component *comp, int index, { struct snd_soc_acpi_mach *mach = dev_get_platdata(comp->card->dev); size_t len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN; - char buf[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ssp_port, tdm_slot; + char *buf; /* See parse_link_formatted_string() for dynamic naming when(s). */ if (!avs_mach_singular_ssp(mach)) @@ -1557,13 +1557,24 @@ static int avs_route_load(struct snd_soc_component *comp, int index, return 0; tdm_slot = avs_mach_ssp_tdm(mach, ssp_port); + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->source, ssp_port, tdm_slot); - strscpy((char *)route->source, buf, len); + route->source = buf; + + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->sink, ssp_port, tdm_slot); - strscpy((char *)route->sink, buf, len); + route->sink = buf; + if (route->control) { + buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL); + if (!buf) + return -ENOMEM; avs_ssp_sprint(buf, len, route->control, ssp_port, tdm_slot); - strscpy((char *)route->control, buf, len); + route->control = buf; } return 0; -- 2.43.0

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my headphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Enable Mute LED on HP 250 G7
by Nazar Bilinskyi 09 Jul '24

09 Jul '24

HP 250 G7 has a mute LED that can be made to work using quirk ALC269_FIXUP_HP_LINE1_MIC1_LED. Enable already existing quirk. Signed-off-by: Nazar Bilinskyi <nbilinskyi(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 811e82474..57eb67f05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10053,6 +10053,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84a6, "HP 250 G7 Notebook PC", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.45.2

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - edited 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was edited from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04 and with alsa-ucm-conf-1.2.12 applied. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Mic not working on "ROG Strix Z790-A Gaming WIFI II" with ALC4080
by GitHub issues - opened 09 Jul '24

09 Jul '24

alsa-project/alsa-ucm-conf issue #431 was opened from luluonino: I ran into this issue https://github.com/alsa-project/alsa-ucm-conf/issues/394 while searching about exactly the same problem on "ROG Strix Z790-A Gaming WIFI II". The commit included in issue 394 fixed the speaker on my microphone plugged into the case front panel. But the mic still doesn't work. It is detected as "Microphone - USB Audio" but it doesn't pick up any sound. I'm running Ubuntu 24.04. Here is the alsa-info dump. Note after applying the fix in issue 394, the audio codec is still detected as generic USB audio but there is indeed sound afterwards. [alsa-info.txt](https://github.com/user-attachments/files/16136523/alsa-info… I've tried all potential solutions I could find online but none worked. Any help is greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/431 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: (subset) [PATCH v6 0/7] ASoC: fsl-asoc-card: add S/PDIF controller support
by Mark Brown 08 Jul '24

08 Jul '24

On Thu, 27 Jun 2024 10:30:57 +0200, Elinor Montmasson wrote: > This is the v6 of the series of patches aiming to make the machine > driver `fsl-asoc-card` compatible with S/PDIF controllers on imx boards. > The main goal is to allow the use of S/PDIF controllers with ASRC > modules. > > The `imx-spdif` machine driver already has specific support for S/PDIF > controllers but doesn't support using an ASRC with it. However, the > `fsl-asoc-card` machine driver has the necessary code to create a sound > card which can use an ASRC module. > It is then possible to extend the support for S/PDIF audio cards by > merging the `imx-spdif` driver into `fsl-asoc-card`. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/7] ASoC: fsl-asoc-card: add support for dai links with multiple codecs commit: a613b63db233b6e7d46ec72f009c1cbb7db8be66 [2/7] ASoC: fsl-asoc-card: add second dai link component for codecs commit: c68fa0d9b0f8cc7c3ae7d29c02adbc97622a73f5 [3/7] ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links commit: fcc6ace84f1f6ce2211af25c3c8fb30a0fb2bb2c [4/7] ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c commit: 6d174cc4f22461ad3fe383570527e86bf1948a2e [5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge commit: 4359caadd17fbde64d656c10bd6f2dc91b675a11 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH alsa-ucm-conf v3 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

2 3

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No change since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

2 3

[PATCH] firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files
by Richard Fitzgerald 08 Jul '24

08 Jul '24

Use strnlen() instead of strlen() on the algorithm and coefficient name string arrays in V1 wmfw files. In V1 wmfw files the name is a NUL-terminated string in a fixed-size array. cs_dsp should protect against overrunning the array if the NUL terminator is missing. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs") --- drivers/firmware/cirrus/cs_dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index d2aa0980ed78..0a504a3c4375 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1166,7 +1166,7 @@ static inline void cs_dsp_coeff_parse_alg(struct cs_dsp *dsp, const u8 **data, blk->id = le32_to_cpu(raw->id); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ncoeff = le32_to_cpu(raw->ncoeff); break; default: @@ -1199,7 +1199,7 @@ static inline void cs_dsp_coeff_parse_coeff(struct cs_dsp *dsp, const u8 **data, blk->offset = le16_to_cpu(raw->hdr.offset); blk->mem_type = le16_to_cpu(raw->hdr.type); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ctl_type = le16_to_cpu(raw->ctl_type); blk->flags = le16_to_cpu(raw->flags); blk->len = le32_to_cpu(raw->len); -- 2.39.2

2 5

[PATCH v4 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 08 Jul '24

08 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Olivier Moysan (8): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 23 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 153 ++++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 298 +++++++++++++++--- drivers/iio/industrialio-backend.c | 116 ++++++- include/linux/iio/backend.h | 14 +- 7 files changed, 633 insertions(+), 65 deletions(-) base-commit: 529d2e1900642eba6df28307e26e19793e227546 -- 2.25.1

2 2

[PATCH 00/11] ASoC: Simplify code with cleanup.h
by Krzysztof Kozlowski 08 Jul '24

08 Jul '24

Allocate the memory with scoped/cleanup.h to reduce error handling (simpler error paths) and make the code a bit smaller. Best regards, Krzysztof --- Krzysztof Kozlowski (11): ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_add_dapms() with cleanup.h ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_probe() with cleanup.h ASoC: codecs: wcd9335: Simplify with cleanup.h ASoC: codecs: wcd934x: Simplify with cleanup.h ASoC: simple-card-utils: Simplify with cleanup.h ASoC: audio-graph-card: Use cleanup.h instead of devm_kfree() ASoC: audio-graph-card2: Use cleanup.h instead of devm_kfree() ASoC: simple-card: Use cleanup.h instead of devm_kfree() ASoC: ops: Simplify with cleanup.h ASoC: dapm: Simplify dapm_cnew_widget() with cleanup.h ASoC: dapm: Simplify snd_soc_dai_link_event_pre_pmu() with cleanup.h sound/soc/codecs/audio-iio-aux.c | 83 ++++++++++++----------------------- sound/soc/codecs/wcd9335.c | 22 ++++------ sound/soc/codecs/wcd934x.c | 22 ++++------ sound/soc/generic/audio-graph-card.c | 5 +-- sound/soc/generic/audio-graph-card2.c | 5 +-- sound/soc/generic/simple-card-utils.c | 13 +++--- sound/soc/generic/simple-card.c | 5 +-- sound/soc/soc-dapm.c | 47 ++++++++------------ sound/soc/soc-ops.c | 26 +++++------ 9 files changed, 87 insertions(+), 141 deletions(-) --- base-commit: cf601827c51b0496f6bb8a1865f1bf69ebd246b7 change-id: 20240703-asoc-cleanup-h-ef9cbfb67b66 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 16

[PATCH v2] firmware: cs_dsp: Use strnlen() on name fields in V1 wmfw files
by Richard Fitzgerald 08 Jul '24

08 Jul '24

Use strnlen() instead of strlen() on the algorithm and coefficient name string arrays in V1 wmfw files. In V1 wmfw files the name is a NUL-terminated string in a fixed-size array. cs_dsp should protect against overrunning the array if the NUL terminator is missing. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs") --- drivers/firmware/cirrus/cs_dsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 16484ab9b09d..8a347b938406 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -1183,4 +1183,4 @@ static int cs_dsp_coeff_parse_alg(struct cs_dsp *dsp, blk->id = le32_to_cpu(raw->id); blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ncoeff = le32_to_cpu(raw->ncoeff); @@ -1260,4 +1260,4 @@ static int cs_dsp_coeff_parse_coeff(struct cs_dsp *dsp, blk->name = raw->name; - blk->name_len = strlen(raw->name); + blk->name_len = strnlen(raw->name, ARRAY_SIZE(raw->name)); blk->ctl_type = le16_to_cpu(raw->ctl_type); blk->flags = le16_to_cpu(raw->flags); -- 2.39.2

1 1

[PATCH v6 5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge
by Elinor Montmasson 08 Jul '24

08 Jul '24

The S/PDIF audio card support with compatible "fsl,imx-audio-spdif" was merged from imx-spdif into the fsl-asoc-card driver. It makes possible to use an S/PDIF with an ASRC. This merge introduces new DT bindings to use with compatible "fsl,imx-audio-spdif" to follow the way fsl-asoc-card works: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". In an upcoming commit, in-tree DTs will be modified to follow these new properties: * Property "spdif-controller" will be renamed "audio-cpu". * spdif_transmitter and spdif_receiver nodes will be declared and linked to the fsl-asoc-card node with the property "audio-codec". To keep backward compatibility with other DTs, support for "spdif-controller", "spdif-in" and "spdif-out" properties is kept. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still supported, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ------------------- .../bindings/sound/fsl-asoc-card.yaml | 53 +++++++++++++-- 2 files changed, 49 insertions(+), 70 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Freescale i.MX audio complex with S/PDIF transceiver - -maintainers: - - Shengjiu Wang <shengjiu.wang(a)nxp.com> - -properties: - compatible: - oneOf: - - items: - - enum: - - fsl,imx-sabreauto-spdif - - fsl,imx6sx-sdb-spdif - - const: fsl,imx-audio-spdif - - enum: - - fsl,imx-audio-spdif - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - spdif-controller: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of the i.MX S/PDIF controller - - spdif-out: - type: boolean - description: - If present, the transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector or jack on the - board or it's connecting to some other IP block, such as an HDMI - encoder or display-controller. - - spdif-in: - type: boolean - description: - If present, the receiving function of S/PDIF will be enabled, - indicating there is a physical S/PDIF in connector/jack on the board. - -required: - - compatible - - model - - spdif-controller - -anyOf: - - required: - - spdif-in - - required: - - spdif-out - -additionalProperties: false - -examples: - - | - sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..92aa47ec72c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -65,6 +65,11 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +86,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +99,15 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +163,10 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. + minItems: 1 + maxItems: 2 mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,6 +182,27 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + description: The phandle of an S/PDIF CPU DAI controller. + + spdif-out: + type: boolean + deprecated: true + description: | + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + deprecated: true + description: | + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + required: - compatible - model @@ -195,3 +231,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; -- 2.34.1

2 1

[PATCH v1] ASoc: TAS2781: replace beXX_to_cpup with get_unaligned_beXX for potentially broken alignment
by Shenghao Ding 08 Jul '24

08 Jul '24

Use get_unaligned_be16 instead of be16_to_cpup and get_unaligned_be32 instead of be32_to_cpup for potentially broken alignment. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add "#include <asm/unaligned.h>" for the declaration of get_unaligned_beXX --- sound/soc/codecs/tas2781-fmwlib.c | 71 +++++++++++++++---------------- 1 file changed, 35 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 838d29fead96..63626b982d04 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -21,7 +21,7 @@ #include <sound/soc.h> #include <sound/tlv.h> #include <sound/tas2781.h> - +#include <asm/unaligned.h> #define ERROR_PRAM_CRCCHK 0x0000000 #define ERROR_YRAM_CRCCHK 0x0000001 @@ -187,8 +187,7 @@ static struct tasdevice_config_info *tasdevice_add_config( /* convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - cfg_info->nblocks = - be32_to_cpup((__be32 *)&config_data[config_offset]); + cfg_info->nblocks = get_unaligned_be32(&config_data[config_offset]); config_offset += 4; /* Several kinds of dsp/algorithm firmwares can run on tas2781, @@ -232,14 +231,14 @@ static struct tasdevice_config_info *tasdevice_add_config( } bk_da[i]->yram_checksum = - be16_to_cpup((__be16 *)&config_data[config_offset]); + get_unaligned_be16(&config_data[config_offset]); config_offset += 2; bk_da[i]->block_size = - be32_to_cpup((__be32 *)&config_data[config_offset]); + get_unaligned_be32(&config_data[config_offset]); config_offset += 4; bk_da[i]->n_subblks = - be32_to_cpup((__be32 *)&config_data[config_offset]); + get_unaligned_be32(&config_data[config_offset]); config_offset += 4; @@ -289,7 +288,7 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) } buf = (unsigned char *)fmw->data; - fw_hdr->img_sz = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); offset += 4; if (fw_hdr->img_sz != fmw->size) { dev_err(tas_priv->dev, @@ -300,9 +299,9 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) goto out; } - fw_hdr->checksum = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); offset += 4; - fw_hdr->binary_version_num = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); if (fw_hdr->binary_version_num < 0x103) { dev_err(tas_priv->dev, "File version 0x%04x is too low", fw_hdr->binary_version_num); @@ -311,7 +310,7 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) goto out; } offset += 4; - fw_hdr->drv_fw_version = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); offset += 8; fw_hdr->plat_type = buf[offset]; offset += 1; @@ -339,11 +338,11 @@ int tasdevice_rca_parser(void *context, const struct firmware *fmw) for (i = 0; i < TASDEVICE_DEVICE_SUM; i++, offset++) fw_hdr->devs[i] = buf[offset]; - fw_hdr->nconfig = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); offset += 4; for (i = 0; i < TASDEVICE_CONFIG_SUM; i++) { - fw_hdr->config_size[i] = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); offset += 4; total_config_sz += fw_hdr->config_size[i]; } @@ -423,7 +422,7 @@ static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, /* convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - block->type = be32_to_cpup((__be32 *)&data[offset]); + block->type = get_unaligned_be32(&data[offset]); offset += 4; block->is_pchksum_present = data[offset]; @@ -438,10 +437,10 @@ static int fw_parse_block_data_kernel(struct tasdevice_fw *tas_fmw, block->ychksum = data[offset]; offset++; - block->blk_size = be32_to_cpup((__be32 *)&data[offset]); + block->blk_size = get_unaligned_be32(&data[offset]); offset += 4; - block->nr_subblocks = be32_to_cpup((__be32 *)&data[offset]); + block->nr_subblocks = get_unaligned_be32(&data[offset]); offset += 4; /* fixed m68k compiling issue: @@ -482,7 +481,7 @@ static int fw_parse_data_kernel(struct tasdevice_fw *tas_fmw, offset = -EINVAL; goto out; } - img_data->nr_blk = be32_to_cpup((__be32 *)&data[offset]); + img_data->nr_blk = get_unaligned_be32(&data[offset]); offset += 4; img_data->dev_blks = kcalloc(img_data->nr_blk, @@ -578,14 +577,14 @@ static int fw_parse_variable_header_kernel( offset = -EINVAL; goto out; } - fw_hdr->device_family = be16_to_cpup((__be16 *)&buf[offset]); + fw_hdr->device_family = get_unaligned_be16(&buf[offset]); if (fw_hdr->device_family != 0) { dev_err(tas_priv->dev, "%s:not TAS device\n", __func__); offset = -EINVAL; goto out; } offset += 2; - fw_hdr->device = be16_to_cpup((__be16 *)&buf[offset]); + fw_hdr->device = get_unaligned_be16(&buf[offset]); if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || fw_hdr->device == 6) { dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); @@ -603,7 +602,7 @@ static int fw_parse_variable_header_kernel( goto out; } - tas_fmw->nr_programs = be32_to_cpup((__be32 *)&buf[offset]); + tas_fmw->nr_programs = get_unaligned_be32(&buf[offset]); offset += 4; if (tas_fmw->nr_programs == 0 || tas_fmw->nr_programs > @@ -622,14 +621,14 @@ static int fw_parse_variable_header_kernel( for (i = 0; i < tas_fmw->nr_programs; i++) { program = &(tas_fmw->programs[i]); - program->prog_size = be32_to_cpup((__be32 *)&buf[offset]); + program->prog_size = get_unaligned_be32(&buf[offset]); offset += 4; } /* Skip the unused prog_size */ offset += 4 * (TASDEVICE_MAXPROGRAM_NUM_KERNEL - tas_fmw->nr_programs); - tas_fmw->nr_configurations = be32_to_cpup((__be32 *)&buf[offset]); + tas_fmw->nr_configurations = get_unaligned_be32(&buf[offset]); offset += 4; /* The max number of config in firmware greater than 4 pieces of @@ -661,7 +660,7 @@ static int fw_parse_variable_header_kernel( for (i = 0; i < tas_fmw->nr_programs; i++) { config = &(tas_fmw->configs[i]); - config->cfg_size = be32_to_cpup((__be32 *)&buf[offset]); + config->cfg_size = get_unaligned_be32(&buf[offset]); offset += 4; } @@ -699,7 +698,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, switch (subblk_typ) { case TASDEVICE_CMD_SING_W: { int i; - unsigned short len = be16_to_cpup((__be16 *)&data[2]); + unsigned short len = get_unaligned_be16(&data[2]); subblk_offset += 2; if (subblk_offset + 4 * len > sublocksize) { @@ -725,7 +724,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, } break; case TASDEVICE_CMD_BURST: { - unsigned short len = be16_to_cpup((__be16 *)&data[2]); + unsigned short len = get_unaligned_be16(&data[2]); subblk_offset += 2; if (subblk_offset + 4 + len > sublocksize) { @@ -766,7 +765,7 @@ static int tasdevice_process_block(void *context, unsigned char *data, is_err = true; break; } - sleep_time = be16_to_cpup((__be16 *)&data[2]) * 1000; + sleep_time = get_unaligned_be16(&data[2]) * 1000; usleep_range(sleep_time, sleep_time + 50); subblk_offset += 2; } @@ -910,7 +909,7 @@ static int fw_parse_variable_hdr(struct tasdevice_priv offset += len; - fw_hdr->device_family = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->device_family = get_unaligned_be32(&buf[offset]); if (fw_hdr->device_family != 0) { dev_err(tas_priv->dev, "%s: not TAS device\n", __func__); offset = -EINVAL; @@ -918,7 +917,7 @@ static int fw_parse_variable_hdr(struct tasdevice_priv } offset += 4; - fw_hdr->device = be32_to_cpup((__be32 *)&buf[offset]); + fw_hdr->device = get_unaligned_be32(&buf[offset]); if (fw_hdr->device >= TASDEVICE_DSP_TAS_MAX_DEVICE || fw_hdr->device == 6) { dev_err(tas_priv->dev, "Unsupported dev %d\n", fw_hdr->device); @@ -963,7 +962,7 @@ static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, offset = -EINVAL; goto out; } - block->type = be32_to_cpup((__be32 *)&data[offset]); + block->type = get_unaligned_be32(&data[offset]); offset += 4; if (tas_fmw->fw_hdr.fixed_hdr.drv_ver >= PPC_DRIVER_CRCCHK) { @@ -988,7 +987,7 @@ static int fw_parse_block_data(struct tasdevice_fw *tas_fmw, block->is_ychksum_present = 0; } - block->nr_cmds = be32_to_cpup((__be32 *)&data[offset]); + block->nr_cmds = get_unaligned_be32(&data[offset]); offset += 4; n = block->nr_cmds * 4; @@ -1039,7 +1038,7 @@ static int fw_parse_data(struct tasdevice_fw *tas_fmw, goto out; } offset += n; - img_data->nr_blk = be16_to_cpup((__be16 *)&data[offset]); + img_data->nr_blk = get_unaligned_be16(&data[offset]); offset += 2; img_data->dev_blks = kcalloc(img_data->nr_blk, @@ -1076,7 +1075,7 @@ static int fw_parse_program_data(struct tasdevice_priv *tas_priv, offset = -EINVAL; goto out; } - tas_fmw->nr_programs = be16_to_cpup((__be16 *)&buf[offset]); + tas_fmw->nr_programs = get_unaligned_be16(&buf[offset]); offset += 2; if (tas_fmw->nr_programs == 0) { @@ -1143,7 +1142,7 @@ static int fw_parse_configuration_data( offset = -EINVAL; goto out; } - tas_fmw->nr_configurations = be16_to_cpup((__be16 *)&data[offset]); + tas_fmw->nr_configurations = get_unaligned_be16(&data[offset]); offset += 2; if (tas_fmw->nr_configurations == 0) { @@ -1775,7 +1774,7 @@ static int fw_parse_header(struct tasdevice_priv *tas_priv, /* Convert data[offset], data[offset + 1], data[offset + 2] and * data[offset + 3] into host */ - fw_fixed_hdr->fwsize = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->fwsize = get_unaligned_be32(&buf[offset]); offset += 4; if (fw_fixed_hdr->fwsize != fmw->size) { dev_err(tas_priv->dev, "File size not match, %lu %u", @@ -1784,9 +1783,9 @@ static int fw_parse_header(struct tasdevice_priv *tas_priv, goto out; } offset += 4; - fw_fixed_hdr->ppcver = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->ppcver = get_unaligned_be32(&buf[offset]); offset += 8; - fw_fixed_hdr->drv_ver = be32_to_cpup((__be32 *)&buf[offset]); + fw_fixed_hdr->drv_ver = get_unaligned_be32(&buf[offset]); offset += 72; out: @@ -1828,7 +1827,7 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, offset = -EINVAL; goto out; } - tas_fmw->nr_calibrations = be16_to_cpup((__be16 *)&data[offset]); + tas_fmw->nr_calibrations = get_unaligned_be16(&data[offset]); offset += 2; if (tas_fmw->nr_calibrations != 1) { -- 2.34.1

2 1

[PATCH alsa-cum-conf v3 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

1 1

Numark Scratch Mixer USB Audio Interface
by amhadebe＠gmail.com 08 Jul '24

08 Jul '24

Hi All, I have a Numark Scratch mixer which I would like to use on Linux with Mixxx. The mixer should be class compliant but there is an issue and the Audio interface does not work. I hope to get your help in getting this fixed in the kernel. I'm on fedora 49 running kernel 6.9.5-200 Here's some info, I have found: $ dmesg [ 74.473642] usb 1-3: new high-speed USB device number 5 using xhci_hcd [ 74.600370] usb 1-3: language id specifier not provided by device, defaulting to English [ 74.600541] usb 1-3: New USB device found, idVendor=15e4, idProduct=0052, bcdDevice= 0.00 [ 74.600556] usb 1-3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.600566] usb 1-3: Product: Numark Scratch [ 74.600574] usb 1-3: Manufacturer: Numark [ 74.600580] usb 1-3: SerialNumber: N31911167105329 [ 74.604272] usb 1-3: Quirk or no altset; falling back to MIDI 1.0 [ 74.606706] usb 1-3: 3:1 : bogus bTerminalLink 1 [ 74.606802] usb 1-3: 4:1 : bogus bTerminalLink 4 [ 156.432125] usb 1-3: USB disconnect, device number 5 $ amidi -l Dir Device Name IO hw:0,0,0 Numark Scratch MIDI 1 $ aplay -l **** List of PLAYBACK Hardware Devices **** card 1: PCH [HDA Intel PCH], device 0: CX20753/4 Analog [CX20753/4 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] $ lsusb -v Bus 001 Device 045: ID 15e4:0052 Numark Numark Scratch Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 [unknown] bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x15e4 Numark idProduct 0x0052 Numark Scratch bcdDevice 0.00 iManufacturer 1 Numark iProduct 2 Numark Scratch iSerial 3 N31911167105329 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 0x0164 bNumInterfaces 5 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0009 bInCollection 1 baInterfaceNr(0) 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 0x0032 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 2 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (Invalid) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (Invalid) bNumEmbMIDIJack 1 baAssocJackID( 0) 2 Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 2 bInterfaceCount 3 bFunctionClass 1 Audio bFunctionSubClass 0 [unknown] bFunctionProtocol 32 iFunction 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 2 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 8 wTotalLength 0x0087 bmControls 0x00 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 5 bmAttributes 1 Internal fixed clock bmControls 0x05 Clock Frequency Control (read-only) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 2 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 5 bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 4 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 26 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 1 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 bmaControls(3) 0x00000000 bmaControls(4) 0x00000000 iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 6 bCSourceID 5 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 5 bNrChannels 6 bmChannelConfig 0x00000000 iChannelNames 8 bmControls 0x0000 iTerminal 2 AudioControl Interface Descriptor: bLength 34 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 7 bSourceID 3 bmaControls(0) 0x00000000 bmaControls(1) 0x00000000 bmaControls(2) 0x00000000 bmaControls(3) 0x00000000 bmaControls(4) 0x00000000 bmaControls(5) 0x00000000 bmaControls(6) 0x00000000 iFeature 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 7 bCSourceID 5 bmControls 0x0000 iTerminal 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 4 bmChannelConfig 0x00000000 iChannelNames 4 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 2 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bmControls 0x00 bFormatType 1 bmFormats 0x00000001 PCM bNrChannels 6 bmChannelConfig 0x00000000 iChannelNames 8 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 4 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x00a8 1x 168 bytes bInterval 1 AudioStreaming Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x00 bLockDelayUnits 0 Undefined wLockDelay 0x0000 Regards Al

1 0

Fix labeling of BOSS Katana ports
by GitHub pull_request - opened 06 Jul '24

06 Jul '24

alsa-project/alsa-ucm-conf pull request #430 was opened from aledomu: I'd like to rename the MIDI ports as well but I don't know how. If you can point me where I can learn about it, I'll be glad to do so and mark this PR as finished. I guess this fix works the same for all Katana models from all generations, but I don't have the PID available to enable it, nor the devices for testing. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/430.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-utils 1/5] aplaymidi2: Add initial version
by Takashi Iwai 06 Jul '24

06 Jul '24

aplaymidi2 is a program similar like aplaymidi, but intended for playing back a MIDI Clip file that was introduced for handling UMP. MIDI Clip file contains UMP packets, and its structure is much simpler than SMF. The options are mostly same as aplaymidi, but I omitted -l option for simplifying the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- configure.ac | 2 +- seq/Makefile.am | 2 +- seq/aplaymidi2/Makefile.am | 5 + seq/aplaymidi2/aplaymidi2.1 | 71 ++++++ seq/aplaymidi2/aplaymidi2.c | 447 ++++++++++++++++++++++++++++++++++++ 5 files changed, 525 insertions(+), 2 deletions(-) create mode 100644 seq/aplaymidi2/Makefile.am create mode 100644 seq/aplaymidi2/aplaymidi2.1 create mode 100644 seq/aplaymidi2/aplaymidi2.c diff --git a/configure.ac b/configure.ac index 173b9ed1644c..708380cf31dc 100644 --- a/configure.ac +++ b/configure.ac @@ -486,7 +486,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ bat/Makefile bat/tests/Makefile bat/tests/asound_state/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ - seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ + seq/aplaymidi/Makefile seq/aplaymidi2/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ seq/aseqsend/Makefile speaker-test/Makefile speaker-test/samples/Makefile \ alsaloop/Makefile alsa-info/Makefile \ axfer/Makefile axfer/test/Makefile \ diff --git a/seq/Makefile.am b/seq/Makefile.am index b0f628aac4da..a6171268186b 100644 --- a/seq/Makefile.am +++ b/seq/Makefile.am @@ -1 +1 @@ -SUBDIRS=aconnect aplaymidi aseqdump aseqnet aseqsend +SUBDIRS=aconnect aplaymidi aplaymidi2 aseqdump aseqnet aseqsend diff --git a/seq/aplaymidi2/Makefile.am b/seq/aplaymidi2/Makefile.am new file mode 100644 index 000000000000..0c5c743f7eb8 --- /dev/null +++ b/seq/aplaymidi2/Makefile.am @@ -0,0 +1,5 @@ +AM_CPPFLAGS = -I$(top_srcdir)/include +EXTRA_DIST = aplaymidi2.1 + +bin_PROGRAMS = aplaymidi2 +man_MANS = aplaymidi2.1 diff --git a/seq/aplaymidi2/aplaymidi2.1 b/seq/aplaymidi2/aplaymidi2.1 new file mode 100644 index 000000000000..99be608f107d --- /dev/null +++ b/seq/aplaymidi2/aplaymidi2.1 @@ -0,0 +1,71 @@ +.TH APLAYMIDI2 1 "4 July 2024" + +.SH NAME +aplaymidi2 \- play MIDI Clip Files + +.SH SYNOPSIS +.B aplaymidi2 +\-p client:port[,...] midi2file ... + +.SH DESCRIPTION +.B aplaymidi2 +is a command-line utility that plays the specified MIDI Clip file(s) to one +or more ALSA sequencer ports. + +.SH OPTIONS + +.TP +.I \-h, \-\-help +Prints a list of options. + +.TP +.I \-V, \-\-version +Prints the current version. + +.TP +.I \-p, \-\-port=client:port,... +Sets the sequencer port(s) to which the events in the MIDI Clip file(s) are +sent. + +A client can be specified by its number, its name, or a prefix of its +name. A port is specified by its number; for port 0 of a client, the +":0" part of the port specification can be omitted. + +Multiple ports can be specified to allow playback of MIDI Clip file(s) that +contain events for multiple devices (ports) corresponding to the +multiple UMP Groups. + +For compatibility with +.B pmidi(1), +the port specification is taken from the +.I ALSA_OUTPUT_PORTS +environment variable if none is given on the command line. + +.B aplaymidi2 +supports only basic UMP events: in addition to the standard MIDI1 and +MIDI2 CVMs and 7bit SysEx, only the following are supported: +DCTPQ, DC, Set Tempo, Start Clip and End Clip. +Lyrics and other meta data in Flex Data are skipped, so far. + +The multiple output ports are useful when the given MIDI Clip file +contains the UMP packets for multiple Groups. +When the destination port is a UMP MIDI 2.0 port, the single +connection should suffice, though, since a MIDI 2.0 port can process +the inputs for multiple Groups. For other cases (e.g. connecting to a +legacy MIDI port), you would need to specify the destination port per +Group. If undefined, it's sent to the first destination port as +default. + +.TP +.I \-d, \-\-delay=seconds +Specifies how long to wait after the end of each MIDI Clip file, +to allow the last notes to die away. +Default is 2 seconds. + +.SH SEE ALSO +pmidi(1) +.br +aplaymidi(1) + +.SH AUTHOR +Takashi Iwai <tiwai(a)suse.de> diff --git a/seq/aplaymidi2/aplaymidi2.c b/seq/aplaymidi2/aplaymidi2.c new file mode 100644 index 000000000000..435b7de1aa95 --- /dev/null +++ b/seq/aplaymidi2/aplaymidi2.c @@ -0,0 +1,447 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * aplaymidi2.c - simple player of a MIDI Clip File over ALSA sequencer + */ + +#include "aconfig.h" +#include <stdio.h> +#include <stdlib.h> +#include <stdarg.h> +#include <string.h> +#include <getopt.h> +#include <unistd.h> +#include <alsa/asoundlib.h> +#include <alsa/ump_msg.h> +#include "version.h" + +static snd_seq_t *seq; +static int client; +static int port_count; +static snd_seq_addr_t ports[16]; +static int queue; +static int end_delay = 2; + +static unsigned int _current_tempo = 50000000; /* default 120 bpm */ +static unsigned int tempo_base = 10; +static unsigned int current_tick; + +/* prints an error message to stderr */ +static void errormsg(const char *msg, ...) +{ + va_list ap; + + va_start(ap, msg); + vfprintf(stderr, msg, ap); + va_end(ap); + fputc('\n', stderr); +} + +/* prints an error message to stderr, and dies */ +static void fatal(const char *msg, ...) +{ + va_list ap; + + va_start(ap, msg); + vfprintf(stderr, msg, ap); + va_end(ap); + fputc('\n', stderr); + exit(EXIT_FAILURE); +} + +/* memory allocation error handling */ +static void check_mem(void *p) +{ + if (!p) + fatal("Out of memory"); +} + +/* error handling for ALSA functions */ +static void check_snd(const char *operation, int err) +{ + if (err < 0) + fatal("Cannot %s - %s", operation, snd_strerror(err)); +} + +/* open and initialize the sequencer client */ +static void init_seq(void) +{ + int err; + + err = snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); + check_snd("open sequencer", err); + + err = snd_seq_set_client_name(seq, "aplaymidi2"); + check_snd("set client name", err); + + client = snd_seq_client_id(seq); + check_snd("get client id", client); + + err = snd_seq_set_client_midi_version(seq, SND_SEQ_CLIENT_UMP_MIDI_2_0); + check_snd("set midi version", err); +} + +/* parses one or more port addresses from the string */ +static void parse_ports(const char *arg) +{ + char *buf, *s, *port_name; + int err; + + /* make a copy of the string because we're going to modify it */ + buf = strdup(arg); + check_mem(buf); + + for (port_name = s = buf; s; port_name = s + 1) { + /* Assume that ports are separated by commas. We don't use + * spaces because those are valid in client names. */ + s = strchr(port_name, ','); + if (s) + *s = '\0'; + + ++port_count; + if (port_count > 16) + fatal("Too many ports specified"); + + err = snd_seq_parse_address(seq, &ports[port_count - 1], port_name); + if (err < 0) + fatal("Invalid port %s - %s", port_name, snd_strerror(err)); + } + + free(buf); +} + +/* create a source port to send from */ +static void create_source_port(void) +{ + snd_seq_port_info_t *pinfo; + int err; + + snd_seq_port_info_alloca(&pinfo); + + /* the first created port is 0 anyway, but let's make sure ... */ + snd_seq_port_info_set_port(pinfo, 0); + snd_seq_port_info_set_port_specified(pinfo, 1); + + snd_seq_port_info_set_name(pinfo, "aplaymidi2"); + + snd_seq_port_info_set_capability(pinfo, 0); /* sic */ + snd_seq_port_info_set_type(pinfo, + SND_SEQ_PORT_TYPE_MIDI_GENERIC | + SND_SEQ_PORT_TYPE_APPLICATION); + + err = snd_seq_create_port(seq, pinfo); + check_snd("create port", err); +} + +/* create a queue */ +static void create_queue(void) +{ + if (!snd_seq_has_queue_tempo_base(seq)) + tempo_base = 1000; + + queue = snd_seq_alloc_named_queue(seq, "aplaymidi2"); + check_snd("create queue", queue); +} + +/* connect to destination ports */ +static void connect_ports(void) +{ + int i, err; + + for (i = 0; i < port_count; ++i) { + err = snd_seq_connect_to(seq, 0, ports[i].client, ports[i].port); + if (err < 0) + fatal("Cannot connect to port %d:%d - %s", + ports[i].client, ports[i].port, snd_strerror(err)); + } +} + +/* read 32bit word and convert to native endian: + * return 0 on success, -1 on error + */ +static int read_word(FILE *file, uint32_t *dest) +{ + uint32_t v; + + if (fread(&v, 4, 1, file) != 1) + return -1; + *dest = be32toh(v); + return 0; +} + +/* read a UMP packet: return the number of packets, -1 on error */ +static int read_ump_packet(FILE *file, uint32_t *buf) +{ + snd_ump_msg_hdr_t *h = (snd_ump_msg_hdr_t *)buf; + + int i, num; + + if (read_word(file, buf) < 0) + return -1; + num = snd_ump_packet_length(h->type); + for (i = 1; i < num; i++) { + if (read_word(file, buf + i) < 0) + return -1; + } + return num; +} + +/* read the file header and verify it's MIDI Clip File: return 0 on success */ +static int verify_file_header(FILE *file) +{ + unsigned char buf[8]; + + if (fread(buf, 1, 8, file) != 8) + return -1; + if (memcmp(buf, "SMF2CLIP", 8)) + return -1; + return 0; +} + +/* return the current tempo, corrected to be sent to host */ +static int current_tempo(void) +{ + if (tempo_base != 10) + return _current_tempo / 100; /* down to us */ + return _current_tempo; +} + +/* send a timer event */ +static void send_timer_event(unsigned int type, unsigned int val) +{ + snd_seq_ump_event_t ev = { + .type = type, + .flags = SND_SEQ_TIME_STAMP_TICK | SND_SEQ_EVENT_LENGTH_FIXED, + }; + + ev.queue = queue; + ev.source.port = 0; + ev.time.tick = current_tick; + + ev.dest.client = SND_SEQ_CLIENT_SYSTEM; + ev.dest.port = SND_SEQ_PORT_SYSTEM_TIMER; + ev.data.queue.queue = queue; + ev.data.queue.param.value = val; + + snd_seq_ump_event_output(seq, &ev); +} + +/* set DCTPQ */ +static void set_dctpq(unsigned int ppq) +{ + snd_seq_queue_tempo_t *queue_tempo; + + snd_seq_queue_tempo_alloca(&queue_tempo); + snd_seq_queue_tempo_set_tempo(queue_tempo, current_tempo()); + snd_seq_queue_tempo_set_ppq(queue_tempo, ppq); + snd_seq_queue_tempo_set_tempo_base(queue_tempo, tempo_base); + + if (snd_seq_set_queue_tempo(seq, queue, queue_tempo) < 0) + errormsg("Cannot set queue tempo (%d)", queue); +} + +/* set DC */ +static void set_dc(unsigned int ticks) +{ + current_tick += ticks; +} + +/* set tempo event */ +static void set_tempo(unsigned int tempo) +{ + _current_tempo = tempo; + send_timer_event(SND_SEQ_EVENT_TEMPO, current_tempo()); +} + +/* start clip */ +static void start_clip(void) +{ + if (snd_seq_start_queue(seq, queue, NULL) < 0) + errormsg("Cannot start queue (%d)", queue); +} + +/* end clip */ +static void end_clip(void) +{ + send_timer_event(SND_SEQ_EVENT_STOP, 0); +} + +/* send a UMP packet */ +static void send_ump(const uint32_t *ump, int len) +{ + snd_seq_ump_event_t ev = { + .flags = SND_SEQ_TIME_STAMP_TICK | SND_SEQ_EVENT_LENGTH_FIXED | + SND_SEQ_EVENT_UMP, + }; + int group; + + memcpy(ev.ump, ump, len * 4); + + ev.queue = queue; + ev.source.port = 0; + ev.time.tick = current_tick; + group = snd_ump_msg_group(ump); + if (group >= port_count) + ev.dest = ports[0]; + else + ev.dest = ports[group]; + + snd_seq_ump_event_output(seq, &ev); +} + +/* play the given MIDI Clip File content */ +static void play_midi(FILE *file) +{ + uint32_t ump[4]; + int len; + + current_tick = 0; + + while ((len = read_ump_packet(file, ump)) > 0) { + const snd_ump_msg_hdr_t *h = (snd_ump_msg_hdr_t *)ump; + + if (h->type == SND_UMP_MSG_TYPE_UTILITY) { + const snd_ump_msg_utility_t *uh = + (const snd_ump_msg_utility_t *)ump; + switch (h->status) { + case SND_UMP_UTILITY_MSG_STATUS_DCTPQ: + set_dctpq(uh->dctpq.ticks); + continue; + case SND_UMP_UTILITY_MSG_STATUS_DC: + set_dc(uh->dctpq.ticks); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_FLEX_DATA) { + const snd_ump_msg_flex_data_t *fh = + (const snd_ump_msg_flex_data_t *)ump; + if (fh->meta.status_bank == SND_UMP_FLEX_DATA_MSG_BANK_SETUP && + fh->meta.status == SND_UMP_FLEX_DATA_MSG_STATUS_SET_TEMPO) { + set_tempo(fh->set_tempo.tempo); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_STREAM) { + const snd_ump_msg_stream_t *sh = + (const snd_ump_msg_stream_t *)ump; + switch (sh->gen.status) { + case SND_UMP_STREAM_MSG_STATUS_START_CLIP: + start_clip(); + continue; + case SND_UMP_STREAM_MSG_STATUS_END_CLIP: + end_clip(); + continue; + } + } else if (h->type == SND_UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE || + h->type == SND_UMP_MSG_TYPE_DATA || + h->type == SND_UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE) { + send_ump(ump, len); + } + } + + snd_seq_drain_output(seq); + snd_seq_sync_output_queue(seq); + + /* give the last notes time to die away */ + if (end_delay > 0) + sleep(end_delay); +} + +static void play_file(const char *file_name) +{ + FILE *file; + + if (!strcmp(file_name, "-")) + file = stdin; + else + file = fopen(file_name, "rb"); + if (!file) { + errormsg("Cannot open %s - %s", file_name, strerror(errno)); + return; + } + + if (verify_file_header(file) < 0) { + errormsg("%s is not a MIDI Clip File", file_name); + goto error; + } + + play_midi(file); + + error: + if (file != stdin) + fclose(file); +} + +static void usage(const char *argv0) +{ + printf( + "Usage: %s -p client:port[,...] [-d delay] midifile ...\n" + "-h, --help this help\n" + "-V, --version print current version\n" + "-p, --port=client:port,... set port(s) to play to\n" + "-d, --delay=seconds delay after song ends\n", + argv0); +} + +static void version(void) +{ + puts("aplaymidi2 version " SND_UTIL_VERSION_STR); +} + +int main(int argc, char *argv[]) +{ + static const struct option long_options[] = { + {"help", 0, NULL, 'h'}, + {"version", 0, NULL, 'V'}, + {"port", 1, NULL, 'p'}, + {"delay", 1, NULL, 'd'}, + {0} + }; + int c; + + init_seq(); + + while ((c = getopt_long(argc, argv, "hVp:d:", + long_options, NULL)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'V': + version(); + return 0; + case 'p': + parse_ports(optarg); + break; + case 'd': + end_delay = atoi(optarg); + break; + default: + usage(argv[0]); + return 1; + } + } + + + if (port_count < 1) { + /* use env var for compatibility with pmidi */ + const char *ports_str = getenv("ALSA_OUTPUT_PORTS"); + if (ports_str) + parse_ports(ports_str); + if (port_count < 1) { + errormsg("Please specify at least one port with --port."); + return 1; + } + } + if (optind >= argc) { + errormsg("Please specify a file to play."); + return 1; + } + + create_source_port(); + create_queue(); + connect_ports(); + + for (; optind < argc; optind++) + play_file(argv[optind]); + + snd_seq_close(seq); + return 0; +} -- 2.43.0

1 4

[PATCH alsa-lib] seq: Add API functions to set different tempo base values
by Takashi Iwai 06 Jul '24

06 Jul '24

MIDI2 Set Tempo message uses 10ns-based values, and we need to update the API to change the base time unit. This patch adds a few new API functions: - snd_seq_has_queue_tempo_base() returns 1 if the client supports a new tempo-base value; if 0, it's an old system and application has to use the tempo in the fixed 1us unit - the tempo base can be changed with snd_seq_queue_tempo_set_tempo_base(), provided in nsec unit; the value has to be either 10 or 1000 (or 0 as default, equivalent with 1000) The protocol version is checked and fallback to the fixed 1us base for the old clients. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/seq.h | 3 ++ include/sound/uapi/asequencer.h | 7 ++-- src/Versions.in.in | 3 ++ src/seq/seq.c | 58 +++++++++++++++++++++++++++++++++ src/seq/seq_hw.c | 5 +++ src/seq/seq_local.h | 1 + 6 files changed, 74 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index e55f5c16fada..3bad5dd9762c 100644 --- a/include/seq.h +++ b/include/seq.h @@ -506,13 +506,16 @@ unsigned int snd_seq_queue_tempo_get_tempo(const snd_seq_queue_tempo_t *info); int snd_seq_queue_tempo_get_ppq(const snd_seq_queue_tempo_t *info); unsigned int snd_seq_queue_tempo_get_skew(const snd_seq_queue_tempo_t *info); unsigned int snd_seq_queue_tempo_get_skew_base(const snd_seq_queue_tempo_t *info); +unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t *info); void snd_seq_queue_tempo_set_tempo(snd_seq_queue_tempo_t *info, unsigned int tempo); void snd_seq_queue_tempo_set_ppq(snd_seq_queue_tempo_t *info, int ppq); void snd_seq_queue_tempo_set_skew(snd_seq_queue_tempo_t *info, unsigned int skew); void snd_seq_queue_tempo_set_skew_base(snd_seq_queue_tempo_t *info, unsigned int base); +void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t *info, unsigned int tempo_base); int snd_seq_get_queue_tempo(snd_seq_t *handle, int q, snd_seq_queue_tempo_t *tempo); int snd_seq_set_queue_tempo(snd_seq_t *handle, int q, snd_seq_queue_tempo_t *tempo); +int snd_seq_has_queue_tempo_base(snd_seq_t *handle); /* */ diff --git a/include/sound/uapi/asequencer.h b/include/sound/uapi/asequencer.h index b913f31daa2d..923dfdddfc76 100644 --- a/include/sound/uapi/asequencer.h +++ b/include/sound/uapi/asequencer.h @@ -26,7 +26,7 @@ #include <sound/asound.h> /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 3) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION(1, 0, 4) /** * definition of sequencer event types @@ -539,11 +539,12 @@ struct snd_seq_queue_status { /* queue tempo */ struct snd_seq_queue_tempo { int queue; /* sequencer queue */ - unsigned int tempo; /* current tempo, us/tick */ + unsigned int tempo; /* current tempo, us/tick (or different time-base below) */ int ppq; /* time resolution, ticks/quarter */ unsigned int skew_value; /* queue skew */ unsigned int skew_base; /* queue skew base */ - char reserved[24]; /* for the future */ + unsigned short tempo_base; /* tempo base in nsec unit; either 10 or 1000 */ + char reserved[22]; /* for the future */ }; diff --git a/src/Versions.in.in b/src/Versions.in.in index 1c40d461dce4..90849277c983 100644 --- a/src/Versions.in.in +++ b/src/Versions.in.in @@ -206,6 +206,9 @@ ALSA_1.2.13 { #ifdef HAVE_SEQ_SYMS @SYMBOL_PREFIX@snd_seq_create_ump_endpoint; @SYMBOL_PREFIX@snd_seq_create_ump_block; + @SYMBOL_PREFIX@snd_seq_queue_tempo_get_tempo_base; + @SYMBOL_PREFIX@snd_seq_queue_tempo_set_tempo_base; + @SYMBOL_PREFIX@snd_seq_has_tempo_base; #endif #ifdef HAVE_RAWMIDI_SYMS @SYMBOL_PREFIX@snd_ump_endpoint_info_clear; diff --git a/src/seq/seq.c b/src/seq/seq.c index a18f67e2632e..d0ecedf0838f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -490,6 +490,21 @@ the buffer is flushed by #snd_seq_drain_output() call. You can schedule the event in a certain queue so that the tempo change happens at the scheduled time, too. +The tempo is set as default in microsecond unit as defined for +Standard MIDI Format 1. But since the value in MIDI2 Set Tempo message +is based on 10-nanosecand unit, sequencer queue also allows to set up +in 10-nanosecond unit. For that, change the tempo-base value in +#snd_seq_queue_tempo_t to 10 via #snd_seq_queue_tempo_set_tempo_base() +along with the 10-nanobased tempo value. The default tempo base is 1000, +i.e. 1 microsecond. +Currently the API supports only either 0, 10 or 1000 as the tempo-base +(where 0 is treated as 1000). + +The older kernel might not support the different tempo-base, and setting a +different value from 1000 would fail. The application may heck the +availability of tempo-base change via #snd_seq_has_tempo_base() function +beforehand, and re-calculate in microsecond unit as fallback. + \subsection seq_ev_start Starting and stopping a queue To start, stop, or continue a queue, you need to send a queue-control @@ -3878,6 +3893,19 @@ unsigned int snd_seq_queue_tempo_get_skew_base(const snd_seq_queue_tempo_t *info return info->skew_base; } +/** + * \brief Get the tempo base of a queue_status container + * \param info queue_status container + * \return tempo base time in nsec unit + * + * \sa snd_seq_get_queue_tempo() + */ +unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t *info) +{ + assert(info); + return info->tempo_base; +} + /** * \brief Set the tempo of a queue_status container * \param info queue_status container @@ -3933,6 +3961,21 @@ void snd_seq_queue_tempo_set_skew_base(snd_seq_queue_tempo_t *info, unsigned int info->skew_base = base; } +/** + * \brief Set the tempo base of a queue_status container + * \param info queue_status container + * \param tempo_base tempo base time in nsec unit + * + * \sa snd_seq_get_queue_tempo() + */ +void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t *info, unsigned int tempo_base) +{ + assert(info); + if (!tempo_base) + tempo_base = 1000; + info->tempo_base = tempo_base; +} + /** * \brief obtain the current tempo of the queue * \param seq sequencer handle @@ -3962,10 +4005,25 @@ int snd_seq_get_queue_tempo(snd_seq_t *seq, int q, snd_seq_queue_tempo_t * tempo int snd_seq_set_queue_tempo(snd_seq_t *seq, int q, snd_seq_queue_tempo_t * tempo) { assert(seq && tempo); + if (!seq->has_queue_tempo_base && + tempo->tempo_base && tempo->tempo_base != 1000) + return -EINVAL; tempo->queue = q; return seq->ops->set_queue_tempo(seq, tempo); } +/** + * \brief inquiry the support of tempo base change + * \param seq sequencer handle + * \return 1 if the client supports the tempo base change, 0 if not + * + * \sa snd_seq_get_queue_tempo() + */ +int snd_seq_has_queue_tempo_base(snd_seq_t *seq) +{ + assert(seq); + return seq->has_queue_tempo_base; +} /*----------------------------------------------------------------*/ diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index eeaf26e16d1c..e88a7b22b052 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -275,12 +275,15 @@ static int snd_seq_hw_get_queue_tempo(snd_seq_t *seq, snd_seq_queue_tempo_t * te /*SYSERR("SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO failed");*/ return -errno; } + if (!seq->has_queue_tempo_base) + tempo->tempo_base = 1000; return 0; } static int snd_seq_hw_set_queue_tempo(snd_seq_t *seq, snd_seq_queue_tempo_t * tempo) { snd_seq_hw_t *hw = seq->private_data; + if (ioctl(hw->fd, SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO, tempo) < 0) { /*SYSERR("SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO failed");*/ return -errno; @@ -587,6 +590,8 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) seq->ops = &snd_seq_hw_ops; seq->private_data = hw; seq->packet_size = sizeof(snd_seq_event_t); + seq->has_queue_tempo_base = ver >= SNDRV_PROTOCOL_VERSION(1, 0, 4); + client = snd_seq_hw_client_id(seq); if (client < 0) { snd_seq_close(seq); diff --git a/src/seq/seq_local.h b/src/seq/seq_local.h index 263029702739..00a7615afeae 100644 --- a/src/seq/seq_local.h +++ b/src/seq/seq_local.h @@ -94,6 +94,7 @@ struct _snd_seq { size_t tmpbufsize; /* size of errbuf */ size_t packet_size; /* input packet alignment size */ int midi_version; /* current protocol version */ + int has_queue_tempo_base; /* support queue tempo-base? */ unsigned int num_ump_groups; /* number of UMP groups */ snd_ump_endpoint_info_t *ump_ep; /* optional UMP info */ -- 2.43.0

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Add support for a new DAC
by Pino Abete 05 Jul '24

05 Jul '24

Hi, I'm not an expert and I'm sorry if this is the wrong place for thsi kind of requests. I'm trying to add native DSD support for this integrated DAC/amplifier: https://www.mission.co.uk/778x/ Unfortunately, vendor is not collaborative. It should support up to 384kHz (PCM) and DSD256 and have a XMOS USB chip and a ESS Sabre32 ES9018K2M DAC chip. I tried simply to add: case USB_ID(0x2622, 0x0011): /* Mission 778X */ in the "switch" section under /* XMOS based USB DACs */ and it works with the DSD_U32_BE format (tried using MPD - Music Player Demo). Is it possible to officially add this to source or do you need a request from OEM? Thanks --

1 0

Re: [PATCH] Fix swapped l/r audio channels for Lenovo ThinBook 13x Gen4
by Takashi Iwai 05 Jul '24

05 Jul '24

On Thu, 04 Jul 2024 23:14:01 +0200, Dmitry Savin wrote: > > Fixes audio channel assignment in configuration table for ThinkBook 13x Gen4. > > Signed-off-by: Dmitry Savin <envelsavinds(a)gmail.com> > Reviewed-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Thanks, applied now. Takashi

1 0

[tiwai-sound:for-next 5/6] ERROR: modpost: "snd_pcm_set_sync_per_card" [sound/pci/emu10k1/snd-emu10k1.ko] undefined!
by kernel test robot 05 Jul '24

05 Jul '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 5f9f982dd71b418aeba7a0b37f87312545c06df4 commit: d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 [5/6] ALSA: pcm: optimize and clarify stream synchronization ID API config: i386-randconfig-r064-20240705 (https://download.01.org/0day-ci/archive/20240705/202407051343.1Sz8AKFt-lkp@…) compiler: gcc-11 (Ubuntu 11.4.0-4ubuntu1) 11.4.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240705/202407051343.1Sz8AKFt-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202407051343.1Sz8AKFt-lkp@intel.com/ All errors (new ones prefixed by >>, old ones prefixed by <<): WARNING: modpost: missing MODULE_DESCRIPTION() in vmlinux.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/locking/locktorture.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/locking/test-ww_mutex.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/rcutorture.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/rcuscale.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/rcu/refscale.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/time/test_udelay.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/time/clocksource-wdtest.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/backtracetest.o WARNING: modpost: missing MODULE_DESCRIPTION() in kernel/torture.o WARNING: modpost: missing MODULE_DESCRIPTION() in mm/dmapool_test.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nfs/nfsv4.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp737.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp855.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp863.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp866.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp936.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_cp950.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_ascii.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_iso8859-7.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_iso8859-13.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-r.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-u.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_koi8-ru.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-croatian.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-inuit.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-romanian.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/mac-turkish.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/nls/nls_ucs2_utils.o WARNING: modpost: missing MODULE_DESCRIPTION() in fs/binfmt_misc.o WARNING: modpost: missing MODULE_DESCRIPTION() in security/keys/encrypted-keys/encrypted-keys.o WARNING: modpost: missing MODULE_DESCRIPTION() in crypto/ecc.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/crypto/libdes.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_hexdump.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_dhry.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_firmware.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_min_heap.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_user_copy.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_static_keys.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_static_key_base.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_printf.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_scanf.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_bitmap.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_uuid.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_xarray.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_maple_tree.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_memcat_p.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_meminit.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_ref_tracker.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/test_objpool.o WARNING: modpost: missing MODULE_DESCRIPTION() in lib/atomic64_test.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/virtio/virtio_dma_buf.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/ali-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/amd64-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/intel-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/intel-gtt.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/nvidia-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/sis-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/agp/sworks-agp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/ttyprintk.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/char/lp.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/panel/panel-orisetech-ota5601a.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/tiny/cirrus.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/drm_panel_orientation_quirks.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/gpu/drm/drm_mipi_dbi.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/base/regmap/regmap-spi-avmm.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mfd/arizona.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mfd/pcf50633-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/cxl/cxl_pci.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/media/dvb-frontends/mb86a16.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/mmc/core/sdio_uart.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-spilib.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-light.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-raw.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-audio-manager.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-gbphy.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-i2c.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-pwm.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/staging/greybus/gb-sdio.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/devfreq/governor_simpleondemand.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/hwtracing/intel_th/intel_th_msu_sink.o WARNING: modpost: missing MODULE_DESCRIPTION() in drivers/siox/siox-bus-gpio.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/vfio-mdev/mdpy-fb.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/vfio-mdev/mbochs.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/kobject/kobject-example.o WARNING: modpost: missing MODULE_DESCRIPTION() in samples/kobject/kset-example.o >> ERROR: modpost: "snd_pcm_set_sync_per_card" [sound/pci/emu10k1/snd-emu10k1.ko] undefined! -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[tiwai-sound:for-next 5/6] sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length
by kernel test robot 05 Jul '24

05 Jul '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 5f9f982dd71b418aeba7a0b37f87312545c06df4 commit: d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 [5/6] ALSA: pcm: optimize and clarify stream synchronization ID API config: x86_64-defconfig (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-lkp@…) compiler: gcc-13 (Ubuntu 13.2.0-4ubuntu3) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202407050501.o5Z3bibi-lkp@intel.com/ All warnings (new ones prefixed by >>): In function 'snd_pcm_set_sync_per_card', inlined from 'snd_pcm_lib_ioctl_sync_id' at sound/core/pcm_lib.c:1837:3, inlined from 'snd_pcm_lib_ioctl' at sound/core/pcm_lib.c:1863:10: >> sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length [-Wstringop-truncation] 547 | strncpy(params->sync + 4, id, len); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/strncpy +547 sound/core/pcm_lib.c 517 518 /** 519 * snd_pcm_set_sync_per_card - set the PCM sync id with card number 520 * @substream: the pcm substream 521 * @params: modified hardware parameters 522 * @id: identifier (max 12 bytes) 523 * @len: identifier length (max 12 bytes) 524 * 525 * Sets the PCM sync identifier for the card with zero padding. 526 * 527 * User space or any user should use this 16-byte identifier for a comparison only 528 * to check if two IDs are similar or different. Special case is the identifier 529 * containing only zeros. Interpretation for this combination is - empty (not set). 530 * The contents of the identifier should not be interpreted in any other way. 531 * 532 * The synchronization ID must be unique per clock source (usually one sound card, 533 * but multiple soundcard may use one PCM word clock source which means that they 534 * are fully synchronized). 535 * 536 * This routine composes this ID using card number in first four bytes and 537 * 12-byte additional ID. When other ID composition is used (e.g. for multiple 538 * sound cards), make sure that the composition does not clash with this 539 * composition scheme. 540 */ 541 void snd_pcm_set_sync_per_card(struct snd_pcm_substream *substream, 542 struct snd_pcm_hw_params *params, 543 const unsigned char *id, unsigned int len) 544 { 545 *(__u32 *)params->sync = cpu_to_le32(substream->pcm->card->number); 546 len = max(12, len); > 547 strncpy(params->sync + 4, id, len); 548 memset(params->sync + 4 + len, 0, 12 - len); 549 } 550 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 0/6] ASoC: codecs: lpass-rx-macro: Few code cleanups
by Krzysztof Kozlowski 04 Jul '24

04 Jul '24

Hi, Changes in v3: - New patch #1 to fix clang jump warning ("ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm") - Link to v2: https://lore.kernel.org/r/20240628-b4-qcom-audio-lpass-codec-cleanups-v2-0-… Changes in v2: - Use cleanup.h instead of devm(), therefore not adding Dmitry's review. - New patch #5. - Link to v1: https://lore.kernel.org/r/20240627-b4-qcom-audio-lpass-codec-cleanups-v1-0-… Improve a bit the Qualcomm LPASS RX macro driver and align similar parts of code with LPASS WSA macro driver for consistency. No external dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm ASoC: codecs: lpass-rx-macro: Simplify with cleanup.h ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgets ASoC: codecs: lpass-wsa-macro: Simplify with cleanup.h include/sound/soc-dapm.h | 2 +- sound/soc/codecs/lpass-macro-common.h | 5 +++ sound/soc/codecs/lpass-rx-macro.c | 63 +++++++++++++++++------------------ sound/soc/codecs/lpass-wsa-macro.c | 22 ++++++------ sound/soc/soc-dapm.c | 2 +- 5 files changed, 47 insertions(+), 47 deletions(-) --- base-commit: feca1ff0cd5ab7bc3990ec5a387d81d4dff88068 change-id: 20240627-b4-qcom-audio-lpass-codec-cleanups-27175f1d069f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 7

[A52] Pipeware errors
by GitHub issues - opened 04 Jul '24

04 Jul '24

alsa-project/alsa-plugins issue #59 was opened from ManuLinares: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4085 I'm getting this errors: ``` jul 03 21:12:38 thought pipewire[63110]: ALSA lib confmisc.c:165:(snd_config_get_card) Cannot get card index for %f jul 03 21:12:38 thought pipewire[63110]: spa.alsa: 'plug:{SLAVE="a52:0,'hw:%f,3'"}': playback open failed: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: 0x5b856abf1c30: can't get format: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: usage: node.name=<string> jul 03 21:14:21 thought wireplumber[63111]: wplua: [string "alsa.lua"]:182: attempt to concatenate a nil value (local 'node_name') stack traceback: [string "alsa.lua"]:182: in function <[string "alsa.lua"]:175> ``` could this be related to the this commit? https://github.com/alsa-project/alsa-plugins/commit/dfc65f2d1f239256c201c68… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/59 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0