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

ACPI REV override for Dell XPS 13 9343
by Cezary Rojewski 15 Jun '22

15 Jun '22

Hello, It's been a while since catpt-driver [1] has been introduced to provide full support for Broadwell (BDW) machines with Intel DSP. For BDW, audio devices can make use of DSP only in I2S mode. In 2015 Rafael and Dominik provided quirk [2] for Dell XPS 13 9343. Given the description: _For example, based on what ACPI exports as the supported revision, Dell XPS 13 (2015) configures its audio device to either work in HDA mode or in I2S mode, where the former is supposed to be used on Linux until the latter is fully supported (in the kernel as well as in user space)._ It's clear that such configuration was not fully supported back then. I believe now it is. Perhaps it is time to let the quirk in mention go? By that I mean just the relevant entry, not the ACPI_REV_OVERRIDE_POSSIBLE functionality as a whole. [1]: https://lore.kernel.org/all/20200929141247.8058-1-cezary.rojewski@intel.com/ [2]: https://lore.kernel.org/alsa-devel/5115175.tbCl1JQVaV@vostro.rjw.lan/ Regards, Czarek

3 5

[bug report] ASoC: tegra: Add Tegra210 based OPE driver
by Dan Carpenter 15 Jun '22

15 Jun '22

Hello Sameer Pujar, The patch 7358a803c778: "ASoC: tegra: Add Tegra210 based OPE driver" from Jun 3, 2022, leads to the following Smatch static checker warning: sound/soc/tegra/tegra210_mbdrc.c:778 tegra210_mbdrc_hw_params() warn: bitwise AND condition is false here sound/soc/tegra/tegra210_mbdrc.c 769 int tegra210_mbdrc_hw_params(struct snd_soc_component *cmpnt) 770 { 771 struct tegra210_ope *ope = snd_soc_component_get_drvdata(cmpnt); 772 const struct tegra210_mbdrc_config *conf = &mbdrc_init_config; 773 u32 val = 0; 774 unsigned int i; 775 776 regmap_read(ope->mbdrc_regmap, TEGRA210_MBDRC_CFG, &val); 777 --> 778 if (val & TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS is zero so this can't be true. #define TEGRA210_MBDRC_CFG_MBDRC_MODE_BYPASS (0 << TEGRA210_MBDRC_CFG_MBDRC_MODE_SHIFT) The common false positive with this warning is that the macro is something which depends on the .config and in that case I just add it to the list in smatch_data/kernel.unconstant_macros. But in this case the macro is just always zero... Is there a plan to make it configurable or something? 779 return 0; 780 781 for (i = 0; i < MBDRC_NUM_BAND; i++) { 782 const struct tegra210_mbdrc_band_params *params = 783 &conf->band_params[i]; 784 785 u32 reg_off = i * TEGRA210_MBDRC_FILTER_PARAM_STRIDE; 786 787 tegra210_mbdrc_write_ram(ope->mbdrc_regmap, 788 reg_off + TEGRA210_MBDRC_CFG_RAM_CTRL, 789 reg_off + TEGRA210_MBDRC_CFG_RAM_DATA, 790 0, (u32 *)&params->biquad_params[0], 791 TEGRA210_MBDRC_MAX_BIQUAD_STAGES * 5); 792 } 793 return 0; 794 } regards, dan carpenter

2 3

pcm: Add error handler for `fgets`
by GitHub pull_request - opened 15 Jun '22

15 Jun '22

alsa-project/alsa-lib pull request #238 was opened from Korilakkuma: `fgets` function returns `NULL` if error occured. Therefore, end loop in the case. Request URL : https://github.com/alsa-project/alsa-lib/pull/238 Patch URL : https://github.com/alsa-project/alsa-lib/pull/238.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Any way to fuse static libasound with alsa-plugins so no dlopens are performed?
by GitHub issues - opened 14 Jun '22

14 Jun '22

alsa-project/alsa-plugins issue #45 was opened from MIvanchev: See title. I have a project that might benefit from such a mix. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/45 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: cs35l45: Add endianness flag in snd_soc_component_driver
by Charles Keepax 14 Jun '22

14 Jun '22

The endianness flag is used on the CODEC side to specify an ambivalence to endian, typically because it is lost over the hardware link. This device receives audio over an I2S DAI and as such should have endianness applied. Fixes: 0d463d016000 ("ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c94edfce4b720..d15b3b77c7eb0 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -500,6 +500,8 @@ static const struct snd_soc_component_driver cs35l45_component = { .num_controls = ARRAY_SIZE(cs35l45_controls), .name = "cs35l45", + + .endianness = 1, }; static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) -- 2.30.2

2 1

[PATCH] ASoC: ops: Fix boolean/integer detection for simple controls
by Mark Brown 14 Jun '22

14 Jun '22

The standard snd_soc_info_volsw() detects if a control is a volume control and needs to be reported as an integer even if it only has two values by looking for the string " Volume" in the control name. This results in false positives if the control has a name like "HP Volume Ramp Switch" since any " Volume" is matched, not just a trailing one. Fix this by making sure that we only match at the end of the control name. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e11109dcaa50..0267e39de2a8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,13 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + const char *vol_string = NULL; int max; max = uinfo->value.integer.max = mc->max - mc->min; if (mc->platform_max && mc->platform_max < max) max = mc->platform_max; - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) + /* Even two value controls ending in Volume should always be integer */ + if (max == 1) { + vol_string = strstr(kcontrol->id.name, " Volume"); + if (vol_string && strcmp(vol_string, " Volume")) + vol_string = NULL; + } + + if (!vol_string) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; -- 2.30.2

1 1

Re: [PATCH] ASoC: ti: omap-mcbsp: duplicate sysfs failure after PROBE_DEFER
by Mark Brown 14 Jun '22

14 Jun '22

On Tue, Jun 14, 2022 at 10:59:31AM -0500, David Owens wrote: > The call to sdma_pcm_platform_register() can return PROBE_DEFER, leading > to omap_mcbsp_init() being called multiple times. sysfs node creation > fails in subsequent calls to omap_mcbsp_init(), which prevents > the driver from ever successfully probing. The resulting errors can be > seen during boot: > > [ 1.749328] sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' > [ 1.759643] CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 5.18.0-yocto-standard #1 > [ 1.767181] Hardware name: Generic OMAP36xx (Flattened Device Tree) > [ 1.773498] Workqueue: events_unbound deferred_probe_work_func > [ 1.779449] unwind_backtrace from show_stack+0x10/0x14 > [ 1.784729] show_stack from sysfs_warn_dup+0x4c/0x60 Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections. > +++ b/sound/soc/ti/omap-mcbsp.c > @@ -1403,6 +1403,10 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > mcbsp->dev = &pdev->dev; > platform_set_drvdata(pdev, mcbsp); > > + ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > + if (ret) > + return ret; > + > ret = omap_mcbsp_init(pdev); > if (ret) > return ret; > @@ -1412,13 +1416,9 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) > omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; > } > > - ret = devm_snd_soc_register_component(&pdev->dev, > + return devm_snd_soc_register_component(&pdev->dev, > &omap_mcbsp_component, > &omap_mcbsp_dai, 1); > - if (ret) > - return ret; > - > - return sdma_pcm_platform_register(&pdev->dev, "tx", "rx"); > } It's not clear to me how this fixes the problem, your commit message doesn't mention how? I was expecting to see more error handling paths being added to unwind the sysfs allocation, or a conversion to devm. As things stand it's not clear to me that the error won't persist in the case where we defer registering the component.

1 0

[PATCH 0/2] ASoC/SoundWire: Intel: add sdw BE dai trigger
by Bard Liao 14 Jun '22

14 Jun '22

For SOF IPC4, we need to set pipeline state in BE DAI trigger. @Vinod, Could you ack if the soundwire patch looks good to you? And we can go through the ASoC tree since the change is mainly on ASoC. Bard Liao (2): soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback drivers/soundwire/intel.c | 8 ++++++++ include/linux/soundwire/sdw_intel.h | 1 + sound/soc/sof/intel/hda-dai.c | 15 ++++++++++++--- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 1 + 5 files changed, 23 insertions(+), 4 deletions(-) -- 2.17.1

3 6

[PATCH] ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support
by Bard Liao 14 Jun '22

14 Jun '22

Some SoundWire hardware topologies rely on different amplifiers or capture devices connected on different links. These devices need to be 'aggregated', remain synchronized and be handled as a single logical device. In the IPC3 solution, the aggregation for amplifiers was handled by a firmware 'demux' component. In the IPC4 solution, the demux component is not needed, the gateway component can handle multiple ALH/DMA transfers at the same time. This change makes the topology slightly more complicated in that only one ALH DAI will be connected in the topology with the gateway. The other DAIs that are part of the 'aggregated' dailink are not shown in the DAPM graph as connected to the gateway, but they will however be activated thanks to a feature in soc-dapm.c where events are forwarded to all DAIs in the dailink (see soc_dapm_stream_event). The topology also sets the same stream name for all widgets, dais and dailinks, so a search for the stream name helps identify cases where SoundWire/ALH aggregation is needed. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 68 ++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-topology.h | 11 ++++++ 2 files changed, 74 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index cb0f0823b8eb..3c949298e007 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -19,6 +19,8 @@ #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 +static DEFINE_IDA(alh_group_ida); + static const struct sof_topology_token ipc4_sched_tokens[] = { {SOF_TKN_SCHED_LP_MODE, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_pipeline, lp_mode)} @@ -478,7 +480,9 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_ALH: { + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_alh_configuration_blob *blob; + struct snd_sof_widget *w; blob = kzalloc(sizeof(*blob), GFP_KERNEL); if (!blob) { @@ -486,6 +490,14 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) goto err; } + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + blob->alh_cfg.count++; + } + ipc4_copier->copier_config = (uint32_t *)blob; ipc4_copier->data.gtw_cfg.config_length = sizeof(*blob) >> 2; break; @@ -844,6 +856,17 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) struct snd_sof_dai *dai = swidget->private; ipc4_copier = dai->private; + if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_alh_configuration_blob *blob; + unsigned int group_id; + + blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; + if (blob->alh_cfg.count > 1) { + group_id = SOF_IPC4_NODE_INDEX(ipc4_copier->data.gtw_cfg.node_id) - + ALH_MULTI_GTW_BASE; + ida_free(&alh_group_ida, group_id); + } + } } if (ipc4_copier) { @@ -973,6 +996,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct sof_ipc4_copier_data *copier_data; struct snd_pcm_hw_params *ref_params; struct sof_ipc4_copier *ipc4_copier; + struct snd_sof_dai *dai; struct snd_mask *fmt; int out_sample_valid_bits; size_t ref_audio_fmt_size; @@ -1022,7 +1046,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: { - struct snd_sof_dai *dai = swidget->private; + dai = swidget->private; ipc4_copier = (struct sof_ipc4_copier *)dai->private; copier_data = &ipc4_copier->data; @@ -1077,22 +1101,56 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { struct sof_ipc4_alh_configuration_blob *blob; + struct sof_ipc4_copier_data *alh_data; + struct sof_ipc4_copier *alh_copier; + struct snd_sof_widget *w; + u32 ch_mask = 0; u32 ch_map; int i; blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; - /* TODO: add aggregation mode support */ - blob->alh_cfg.count = 1; - blob->alh_cfg.mapping[0].alh_id = copier_data->gtw_cfg.node_id; + blob->gw_attr.lp_buffer_alloc = 0; /* Get channel_mask from ch_map */ ch_map = copier_data->base_config.audio_fmt.ch_map; for (i = 0; ch_map; i++) { if ((ch_map & 0xf) != 0xf) - blob->alh_cfg.mapping[0].channel_mask |= BIT(i); + ch_mask |= BIT(i); ch_map >>= 4; } + + /* + * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[] + * for all widgets with the same stream name + */ + i = 0; + list_for_each_entry(w, &sdev->widget_list, list) { + if (w->widget->sname && + strcmp(w->widget->sname, swidget->widget->sname)) + continue; + + dai = w->private; + alh_copier = (struct sof_ipc4_copier *)dai->private; + alh_data = &alh_copier->data; + blob->alh_cfg.mapping[i].alh_id = alh_data->gtw_cfg.node_id; + blob->alh_cfg.mapping[i].channel_mask = ch_mask; + i++; + } + if (blob->alh_cfg.count > 1) { + int group_id; + + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + GFP_KERNEL); + + if (group_id < 0) + return group_id; + + /* add multi-gateway base */ + group_id += ALH_MULTI_GTW_BASE; + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(group_id); + } } } } diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 1a9c0627bae9..3bc2fe38c733 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -42,6 +42,17 @@ #define ALH_MAX_NUMBER_OF_GTW 16 +/* + * The base of multi-gateways. Multi-gateways addressing starts from + * ALH_MULTI_GTW_BASE and there are ALH_MULTI_GTW_COUNT multi-sources + * and ALH_MULTI_GTW_COUNT multi-sinks available. + * Addressing is continuous from ALH_MULTI_GTW_BASE to + * ALH_MULTI_GTW_BASE + ALH_MULTI_GTW_COUNT - 1. + */ +#define ALH_MULTI_GTW_BASE 0x50 +/* A magic number from FW */ +#define ALH_MULTI_GTW_COUNT 8 + /** * struct sof_ipc4_pipeline - pipeline config data * @priority: Priority of this pipeline -- 2.17.1

2 1

[PATCH] ALSA: pcm: Test for "silence" field in struct "pcm_format_data"
by Fabio M. De Francesco 14 Jun '22

14 Jun '22

Syzbot reports "KASAN: null-ptr-deref Write in snd_pcm_format_set_silence".[1] It is due to missing validation of the "silence" field of struct "pcm_format_data" in "pcm_formats" array. Add a test for valid "pat" and, if it is not so, return -EINVAL. [1] https://lore.kernel.org/lkml/000000000000d188ef05dc2c7279@google.com/ Reported-and-tested-by: syzbot+205eb15961852c2c5974(a)syzkaller.appspotmail.com Signed-off-by: Fabio M. De Francesco <fmdefrancesco(a)gmail.com> --- I wasn't able to figure out the commit for the "Fixes:" tag. If this patch is good, can someone please help with providing this missing information? sound/core/pcm_misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4866aed97aac..5588b6a1ee8b 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -433,7 +433,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; - if (! width) + if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { -- 2.34.1

3 8