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

[PATCH] ASoC: mxs-saif: Fix refcount leak in mxs_saif_probe
by Miaoqian Lin 12 May '22

12 May '22

of_parse_phandle() returns a node pointer with refcount incremented, we should use of_node_put() on it when done. Fixes: 08641c7c74dd ("ASoC: mxs: add device tree support for mxs-saif") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mxs/mxs-saif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 879c1221a809..7afe1a1acc56 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -754,6 +754,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = saif->id; } else { ret = of_alias_get_id(master, "saif"); + of_node_put(master); if (ret < 0) return ret; else -- 2.25.1

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[PATCH 1/2] ASoC: ops: Fix bounds check for _sx controls
by Mark Brown 12 May '22

12 May '22

For _sx controls the semantics of the max field is not the usual one, max is the number of steps rather than the maximum value. This means that our check in snd_soc_put_volsw_sx() needs to just check against the maximum value. Fixes: 4f1e50d6a9cf9c1b ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx()") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..1ac7e2ce31a1 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -435,7 +435,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; - if (val > max - min) + if (val > max) return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; -- 2.30.2

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[PATCH] ASoC: fsl: Fix refcount leak in imx_sgtl5000_probe
by Miaoqian Lin 12 May '22

12 May '22

of_find_i2c_device_by_node() takes a reference, In error paths, we should call put_device() to drop the reference to aviod refount leak. Fixes: 81e8e4926167 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/fsl/imx-sgtl5000.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 8daced42d55e..580a0d963f0e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -120,19 +120,19 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; - goto fail; + goto put_device; } comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL); if (!comp) { ret = -ENOMEM; - goto fail; + goto put_device; } data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); - goto fail; + goto put_device; } data->clk_frequency = clk_get_rate(data->codec_clk); @@ -158,10 +158,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto fail; + goto put_device; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto fail; + goto put_device; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -174,7 +174,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err_probe(&pdev->dev, ret, "snd_soc_register_card failed\n"); - goto fail; + goto put_device; } of_node_put(ssi_np); @@ -182,6 +182,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; +put_device: + put_device(&codec_dev->dev); fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); -- 2.25.1

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[PATCH 0/6] ASOC: Fix the error handling code of the probe
by Zheyu Ma 12 May '22

12 May '22

These drivers mishandle the regulator resource in the probe function, failing to disable the regulator for probing failure. Zheyu Ma (6): ASoC: cs42l52: Fix the error handling of cs42l56_i2c_probe() ASoC: cs35l36: Fix the error handling of cs35l36_i2c_probe() ASoC: rt5645: Fix the error handling of rt5645_i2c_probe() ASoC: tas571x: Fix the error handling of tas571x_i2c_probe() ASoC: tas6424: Fix the error handling of tas6424_i2c_probe() ASoC: wm8903: Fix the error handling of wm8903_i2c_probe() sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/tas571x.c | 6 ++++-- sound/soc/codecs/tas6424.c | 8 ++++++-- sound/soc/codecs/wm8903.c | 2 +- 6 files changed, 15 insertions(+), 9 deletions(-) -- 2.25.1

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[PATCH] ASoC: fsl_ssi: Add support multi fifo script
by Shengjiu Wang 12 May '22

12 May '22

With dual fifo enabled, the case that recording mono sound in the background, playback mono sound twice in parallal, at second time playback sound may distort, the possible reason is using dual fifo to playback mono sound is not recommended. This patch is to provide a option to use multi fifo script, which can be dynamically configured as one fifo or two fifo mode. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 34 ++++++++++++++++++++++++++++++++-- 1 file changed, 32 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ca30a4ede076..84cb36d9dfea 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -40,6 +40,7 @@ #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/dma/imx-dma.h> #include <sound/core.h> #include <sound/pcm.h> @@ -214,6 +215,7 @@ struct fsl_ssi_soc_data { * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode + * @use_dyna_fifo: DMA with support for multi FIFO script * @has_ipg_clk_name: If "ipg" is in the clock name list of device tree * @fifo_depth: Depth of the SSI FIFOs * @slot_width: Width of each DAI slot @@ -243,6 +245,7 @@ struct fsl_ssi_soc_data { * @dma_maxburst: Max number of words to transfer in one go. So far, * this is always the same as fifo_watermark. * @ac97_reg_lock: Mutex lock to serialize AC97 register access operations + * @audio_config: configure for dma multi fifo script */ struct fsl_ssi { struct regmap *regs; @@ -255,6 +258,7 @@ struct fsl_ssi { bool synchronous; bool use_dma; bool use_dual_fifo; + bool use_dyna_fifo; bool has_ipg_clk_name; unsigned int fifo_depth; unsigned int slot_width; @@ -287,6 +291,7 @@ struct fsl_ssi { u32 dma_maxburst; struct mutex ac97_reg_lock; + struct sdma_peripheral_config audio_config[2]; }; /* @@ -643,7 +648,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * task from fifo0, fifo1 would be neglected at the end of each * period. But SSI would still access fifo1 with an invalid data. */ - if (ssi->use_dual_fifo) + if (ssi->use_dual_fifo || ssi->use_dyna_fifo) snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2); @@ -802,6 +807,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, { bool tx2, tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(dai); + struct fsl_ssi_regvals *vals = ssi->regvals; struct regmap *regs = ssi->regs; unsigned int channels = params_channels(hw_params); unsigned int sample_size = params_width(hw_params); @@ -856,6 +862,28 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, tx2 = tx || ssi->synchronous; regmap_update_bits(regs, REG_SSI_SxCCR(tx2), SSI_SxCCR_WL_MASK, wl); + if (ssi->use_dyna_fifo) { + if (channels == 1) { + ssi->audio_config[0].n_fifos_dst = 1; + ssi->audio_config[1].n_fifos_src = 1; + vals[RX].srcr &= ~SSI_SRCR_RFEN1; + vals[TX].stcr &= ~SSI_STCR_TFEN1; + vals[RX].scr &= ~SSI_SCR_TCH_EN; + vals[TX].scr &= ~SSI_SCR_TCH_EN; + } else { + ssi->audio_config[0].n_fifos_dst = 2; + ssi->audio_config[1].n_fifos_src = 2; + vals[RX].srcr |= SSI_SRCR_RFEN1; + vals[TX].stcr |= SSI_STCR_TFEN1; + vals[RX].scr |= SSI_SCR_TCH_EN; + vals[TX].scr |= SSI_SCR_TCH_EN; + } + ssi->dma_params_tx.peripheral_config = &ssi->audio_config[0]; + ssi->dma_params_tx.peripheral_size = sizeof(ssi->audio_config[0]); + ssi->dma_params_rx.peripheral_config = &ssi->audio_config[1]; + ssi->dma_params_rx.peripheral_size = sizeof(ssi->audio_config[1]); + } + return 0; } @@ -1353,7 +1381,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi->dma_params_rx.addr = ssi->ssi_phys + REG_SSI_SRX0; /* Use even numbers to avoid channel swap due to SDMA script design */ - if (ssi->use_dual_fifo) { + if (ssi->use_dual_fifo || ssi->use_dyna_fifo) { ssi->dma_params_tx.maxburst &= ~0x1; ssi->dma_params_rx.maxburst &= ~0x1; } @@ -1446,6 +1474,8 @@ static int fsl_ssi_probe_from_dt(struct fsl_ssi *ssi) if (ssi->use_dma && !ret && dmas[2] == IMX_DMATYPE_SSI_DUAL) ssi->use_dual_fifo = true; + if (ssi->use_dma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + ssi->use_dyna_fifo = true; /* * Backward compatible for older bindings by manually triggering the * machine driver's probe(). Use /compatible property, including the -- 2.17.1

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[PATCH 1/2] ASoC: wm2000: Remove redundant endianness flag
by Charles Keepax 12 May '22

12 May '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 has no DAI links and as such the flag would have no effect, remove the redundant flag. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm2000.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 075ee852eb724..6a86ce3fb182d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -803,7 +803,6 @@ static const struct snd_soc_component_driver soc_component_dev_wm2000 = { .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), .idle_bias_on = 1, .use_pmdown_time = 1, - .endianness = 1, .non_legacy_dai_naming = 1, }; -- 2.30.2

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[PATCH 0/3] firewire: fix minor issues
by Takashi Sakamoto 12 May '22

12 May '22

Hi, This patchset includes below patches posted by several developers. * [PATCH v2] firewire: convert sysfs sprintf/snprintf family to sysfs_emit * https://lore.kernel.org/lkml/1612424420-96871-1-git-send-email-jiapeng.chon… * [PATCH V2] drivers/firewire: use struct_size over open coded arithmetic * https://lore.kernel.org/lkml/20220210060805.1608198-1-chi.minghao@zte.com.c… * [PATCH] firewire: Fix using uninitialized value * https://lore.kernel.org/lkml/20220411105205.2520784-1-lv.ruyi@zte.com.cn/ They are for some minor issues, while preferable. I expect the patches are going to be sent to Linus for next merge window by maintainer of either Linux FireWire subsystem or sound subsystem. Regards Jiapeng Chong (1): firewire: convert sysfs sprintf/snprintf family to sysfs_emit Lv Ruyi (1): firewire: Fix using uninitialized value Minghao Chi (CGEL ZTE) (1): firewire: use struct_size over open coded arithmetic drivers/firewire/core-device.c | 6 ++---- drivers/firewire/core-transaction.c | 2 +- 2 files changed, 3 insertions(+), 5 deletions(-) -- 2.34.1

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[PATCH v7 0/2] Add generic serial MIDI driver using serial bus API
by Daniel Kaehn 12 May '22

12 May '22

Generic serial MIDI driver adding support for using serial devices compatible with the serial bus as raw MIDI devices, allowing using additional serial devices not compatible with the existing serial-u16550 driver. Supports only setting standard serial baudrates on the underlying serial device; however, the underlying serial device can be configured so that a requested 38.4 kBaud is actually the standard MIDI 31.25 kBaud. Supports DeviceTree configuration. Changes in v7: - Separate examples in dt-binding to remove need to specify unit name (fixing dt_binding_check error) Changes in v6: - Change compatible "serialmidi" -> serial-midi" - Default current-speed to 38400 (closest baud to MIDI standard speed) - Appropriately stop reading or writing MIDI if input or output _trigger() is called with a parameter of zero, respectively - Zero out corresponding triggered state on close to ensure input and output closing results in the serial port being closed - Fix order of operations in _probe() - Remove "DEBUG" literal from debug messages - Remove unused dt-parsing patch checking for existence of node - Whitespace / tabbing fixes / improvements Changes in v5: - Reword description in dt-binding for clarity - Change 'speed' dt property to standard 'current-speed' - Move MIDI output loop onto workqueue (since this could loop quite a while, if ALSA provides a continuous stream of bytes) - Add tx_state bit flags to snd_serial_generic struct - Safegard critical section in tx_work with atomic bit ops on tx_state - Switch operations on filemode to use atomic bit ops Changes in v4: - Fix regressed typo - Correct 3.84 kBaud -> 38.4 kBaud in DT & Kconfig (sorry about spam - noticed after sending v3 and didn't want to let the error sit around for too long) Changes in v3: - Replace use of snd_printk() with dev_* alternatives - Removed unnecessary initialization of err variables - Replaced instances of `== SERIAL_MODE_NOT_OPENED` with zero check - Loop on output_write to completely fill output buffer if data available - Depend on CONFIG_OF in Kconfig - Replace use of devm_kzalloc() with extra_size allocation in snd_devm_card_new() - Use module_serdev_device_driver() instead of module_init() and module_exit(0) Changes in v2: - Fix 'snd_serial_generic_write_wakeup' missing static keyword - Correct 3.125 kBaud > 31.25 kBaud in documentation for MIDI The need for this driver arose from a project using a Raspberry Pi4 which needed to receive and send raw MIDI with low latency. The pl011 UART used is not compatible with the existing serial MIDI driver made for u16550-style devices. Using a userspace program such as ttymidi to feed input from the TTY device to a virtual ALSA MIDI device was functional, but not ideal. I am not sure if a MIDI driver needing the mentioned 'hack' to clock 38.4 kBaud down to the standard MIDI baud is permissible in the mainline kernel, but am submitting nevertheless in case it is useful. To my knowledge, it doesn't seem that there would be any way for this driver to manually configure a serial port to 31.25 kBaud using the serial bus API (please correct me f I'm wrong). In my use case, I am actually configuring one port to run at 115.2 kBaud for faster communication with a custom onboard MIDI controller. Daniel Kaehn (2): dt-bindings: sound: Add generic serial MIDI device Add generic serial MIDI driver using serial bus API .../bindings/sound/serial-midi.yaml | 51 +++ sound/drivers/Kconfig | 18 + sound/drivers/Makefile | 2 + sound/drivers/serial-generic.c | 374 ++++++++++++++++++ 4 files changed, 445 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/serial-midi.yaml create mode 100644 sound/drivers/serial-generic.c base-commit: c5eb0a61238dd6faf37f58c9ce61c9980aaffd7a -- 2.33.0

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[PATCH v2 0/3] ALSA: hda/cs8409: Add support for Odin Laptop Variants
by Stefan Binding 12 May '22

12 May '22

Add support for new Odin Laptop Variants into CS8409 HDA Driver. Since these laptops require Speaker Playback Switch, support must be added into CS8409 HDA driver. Since CS8409 does not have amplifier parameters for the NID associated with the speaker, the HDA driver does not add the switch automatically, so the driver needs to add this support manually. changes since v1: - add missing break in switch statement Stefan Binding (3): ALSA: hda/cs8409: Support new Odin Variants ALSA: hda/cs8409: Add Speaker Playback Switch for Cyborg ALSA: hda/cs8409: Add Speaker Playback Switch for Warlock sound/pci/hda/patch_cs8409-tables.c | 19 +++++++ sound/pci/hda/patch_cs8409.c | 86 ++++++++++++++++++++++++----- sound/pci/hda/patch_cs8409.h | 5 ++ 3 files changed, 96 insertions(+), 14 deletions(-) -- 2.25.1

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[PATCH -next] ALSA: ppc: tumbler: use for_each_child_of_node() macro
by Yang Yingliang 12 May '22

12 May '22

Use for_each_child_of_node() macro instead of open coding it. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/ppc/tumbler.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index c65e74d7cd0a..c7e150f82bcc 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1060,8 +1060,7 @@ static struct device_node *find_audio_device(const char *name) if (! gpiop) return NULL; - for (np = of_get_next_child(gpiop, NULL); np; - np = of_get_next_child(gpiop, np)) { + for_each_child_of_node(gpiop, np) { const char *property = of_get_property(np, "audio-gpio", NULL); if (property && strcmp(property, name) == 0) break; -- 2.25.1

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