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

[PATCH] ASoC: inno_rk3036: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the inno_rk3036 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/inno_rk3036.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index e05c4f27486e..ca0f4c1911e4 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -200,12 +200,12 @@ static int rk3036_codec_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(component->dev, "rk3036_codec dai set fmt : %08x\n", fmt); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: reg01_val |= INNO_R01_PINDIR_IN_SLAVE | INNO_R01_I2SMODE_SLAVE; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: reg01_val |= INNO_R01_PINDIR_OUT_MASTER | INNO_R01_I2SMODE_MASTER; break; -- 2.30.2

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[PATCH] ASoC: es8328: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es8328 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..3f00ead97006 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -84,7 +84,7 @@ struct es8328_priv { int mclkdiv2; const struct snd_pcm_hw_constraint_list *sysclk_constraints; const int *mclk_ratios; - bool master; + bool provider; struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; }; @@ -462,7 +462,7 @@ static int es8328_startup(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8328_priv *es8328 = snd_soc_component_get_drvdata(component); - if (es8328->master && es8328->sysclk_constraints) + if (es8328->provider && es8328->sysclk_constraints) snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, es8328->sysclk_constraints); @@ -486,7 +486,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, else reg = ES8328_ADCCONTROL5; - if (es8328->master) { + if (es8328->provider) { if (!es8328->sysclk_constraints) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; @@ -590,19 +590,19 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 dac_mode = 0; u8 adc_mode = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: /* Master serial port mode, with BCLK generated automatically */ snd_soc_component_update_bits(component, ES8328_MASTERMODE, ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC); - es8328->master = true; + es8328->provider = true; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* Slave serial port mode */ snd_soc_component_update_bits(component, ES8328_MASTERMODE, ES8328_MASTERMODE_MSC, 0); - es8328->master = false; + es8328->provider = false; break; default: return -EINVAL; -- 2.30.2

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[PATCH] ASoC: es8316: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es8316 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8316.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8f30a3ea8bfe..ff33eab6f9de 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -401,8 +401,8 @@ static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 clksw; u8 mask; - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { - dev_err(component->dev, "Codec driver only supports slave mode\n"); + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { + dev_err(component->dev, "Codec driver only supports consumer mode\n"); return -EINVAL; } -- 2.30.2

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[PATCH] ASoC: es7241: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es7241 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es7241.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c index 9f20bfb855b3..0baa86241cf9 100644 --- a/sound/soc/codecs/es7241.c +++ b/sound/soc/codecs/es7241.c @@ -29,7 +29,7 @@ struct es7241_data { struct gpio_desc *m1; unsigned int fmt; unsigned int mclk; - bool is_slave; + bool is_consumer; const struct es7241_chip *chip; }; @@ -46,9 +46,9 @@ static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) gpiod_set_value_cansleep(priv->reset, 1); } -static int es7241_set_slave_mode(struct es7241_data *priv, - const struct es7241_clock_mode *mode, - unsigned int mfs) +static int es7241_set_consumer_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) { int j; @@ -67,9 +67,9 @@ static int es7241_set_slave_mode(struct es7241_data *priv, return 0; } -static int es7241_set_master_mode(struct es7241_data *priv, - const struct es7241_clock_mode *mode, - unsigned int mfs) +static int es7241_set_provider_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) { /* * We can't really set clock ratio, if the mclk/lrclk is different @@ -98,10 +98,10 @@ static int es7241_hw_params(struct snd_pcm_substream *substream, if (rate < mode->rate_min || rate >= mode->rate_max) continue; - if (priv->is_slave) - return es7241_set_slave_mode(priv, mode, mfs); + if (priv->is_consumer) + return es7241_set_consumer_mode(priv, mode, mfs); else - return es7241_set_master_mode(priv, mode, mfs); + return es7241_set_provider_mode(priv, mode, mfs); } /* should not happen */ @@ -136,12 +136,12 @@ static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - priv->is_slave = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + priv->is_consumer = true; break; - case SND_SOC_DAIFMT_CBM_CFM: - priv->is_slave = false; + case SND_SOC_DAIFMT_CBP_CFP: + priv->is_consumer = false; break; default: -- 2.30.2

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[PATCH] ASoC: es7134: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es7134 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es7134.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index e2b79879354b..f443351677df 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -94,7 +94,7 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) SND_SOC_DAIFMT_MASTER_MASK); if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS)) { + SND_SOC_DAIFMT_CBC_CFC)) { dev_err(codec_dai->dev, "Invalid DAI format\n"); return -EINVAL; } -- 2.30.2

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[PATCH 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 28 Feb '22

28 Feb '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 49 ++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 15 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 226 ++++++++++ sound/soc/intel/avs/cldma.c | 328 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 410 +++++++++++++++++ sound/soc/intel/avs/loader.c | 594 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 642 +++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 762 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 282 ++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3803 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

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[PATCH v4 1/2] ALSA: core: remove initialise static variables to 0
by Meng Tang 28 Feb '22

28 Feb '22

Initializing the static variable to 0 causes the following error when exec checkpatch: ERROR: do not initialise statics to 0 FILE: sound/sound_core.c:142: static int preclaim_oss = 0; In addition, considering the following way of writing 139: #ifdef config_sound_oss_core_preclaim 140: Static int preclaim_oss = 1; 141: #ELSE 142: Static int preclaim_oss = 0; 143: #ENDIF We can optimize it by IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM), so modified it to static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/sound_core.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 90d118cd9164..aa4a57e488e5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -136,11 +136,7 @@ struct sound_unit * All these clutters are scheduled to be removed along with * sound-slot/service-* module aliases. */ -#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM -static int preclaim_oss = 1; -#else -static int preclaim_oss = 0; -#endif +static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); module_param(preclaim_oss, int, 0444); -- 2.20.1

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[PATCH] ALSA: spi: Add check for clk_enable()
by Jiasheng Jiang 28 Feb '22

28 Feb '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: 3568459a5113 ("ALSA: at73c213: manage SSC clock") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/spi/at73c213.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 76c0e37a838c..8a2da6b1012e 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -218,7 +218,9 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_at73c213_playback_hw; chip->substream = substream; - clk_enable(chip->ssc->clk); + err = clk_enable(chip->ssc->clk); + if (err) + return err; return 0; } @@ -776,7 +778,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip) goto out; /* Enable DAC master clock. */ - clk_enable(chip->board->dac_clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + goto out; /* Initialize at73c213 on SPI bus. */ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); @@ -889,7 +893,9 @@ static int snd_at73c213_dev_init(struct snd_card *card, chip->card = card; chip->irq = -1; - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + return retval; retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); if (retval) { @@ -1008,7 +1014,9 @@ static int snd_at73c213_remove(struct spi_device *spi) int retval; /* Stop playback. */ - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + goto out; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); clk_disable(chip->ssc->clk); @@ -1088,9 +1096,16 @@ static int snd_at73c213_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; + int retval; - clk_enable(chip->board->dac_clk); - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + return retval; + retval = clk_enable(chip->ssc->clk); + if (retval) { + clk_disable(chip->board->dac_clk); + return retval; + } ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); return 0; -- 2.25.1

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[PATCH] sound/mips: Use platform_get_irq() to get the interrupt
by Meng Tang 28 Feb '22

28 Feb '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/mips/snd-n64.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index 463a6fe589eb..bff6d85b8fe2 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -289,8 +289,7 @@ static int __init n64audio_probe(struct platform_device *pdev) struct snd_card *card; struct snd_pcm *pcm; struct n64audio *priv; - struct resource *res; - int err; + int err, irq; err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, @@ -337,12 +336,12 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->shortname, "N64 Audio"); strcpy(card->longname, "N64 Audio"); - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { + irq = platform_get_irq(pdev, 0); + if (irq < 0) { err = -EINVAL; goto fail_dma_alloc; } - if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, + if (devm_request_irq(&pdev->dev, irq, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; goto fail_dma_alloc; -- 2.20.1

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[PATCH V0 0/1] ALSA: pcm: fix blocking while loop in
by Raghu Bankapur 28 Feb '22

28 Feb '22

When period interrupts are disabled, while loop in snd_pcm_update_hw_ptr0() results in the machine locking up if runtime->hw_ptr_buffer_jiffies is 0. Validate runtime->hw_ptr_buffer_jiffies value before while loop to avoid delta check. when runtime->hw_ptr_buffer_jiffies is 0 then update of while condition hdelta > xrun_threshold will be always true because hdelta will not change. This will lead to infinite loop causing lock-up This issue found during stability test, below log explains CPU lock-up on CPU1 [ 922.249028] sched: RT throttling activated for rt_rq ffffffc0f9970778 (cpu 1) [ 922.249028] potential CPU hogs: [ 922.249028] **FastCapture (11426) [ 922.249105] ------------[ cut here ]--------- To fix this lock-up issue, added new condition to check if runtime-> hw_ptr_buffer_jiffies is ZERO or negative. Raghu Bankapur (1): ALSA: pcm: fix blocking while loop in snd_pcm_update_hw_ptr0() sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) -- 2.17.1

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