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

[PATCH] ASoC: wm_adsp: Minor clean and redundant code removal
by Charles Keepax 30 Jun '22

30 Jun '22

The cs_dsp core will return an error if passed a NULL cs_dsp struct so there is no need for the wm_adsp_write|read_ctl functions to manually check that. The cs_dsp core will also check the data is within bounds of the control so the additional bounds check is redundant too. Simplify things a bit by removing said code. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index a7784ac15dde6..cfaa45ede916a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -675,21 +675,12 @@ static void wm_adsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; + struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); struct wm_coeff_ctl *ctl; struct snd_kcontrol *kcontrol; char ctl_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int ret; - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - ctl = cs_ctl->priv; - - if (len > cs_ctl->len) - return -EINVAL; - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); if (ret) return ret; @@ -697,6 +688,8 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) return 0; + ctl = cs_ctl->priv; + if (dsp->component->name_prefix) snprintf(ctl_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s", dsp->component->name_prefix, ctl->name); @@ -720,16 +713,8 @@ EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; - - cs_ctl = cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - if (len > cs_ctl->len) - return -EINVAL; - - return cs_dsp_coeff_read_ctrl(cs_ctl, 0, buf, len); + return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(&dsp->cs_dsp, name, type, alg), + 0, buf, len); } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); -- 2.30.2

2 1

Re: [PATCH] sound: delete a semicolon
by Mark Brown 30 Jun '22

30 Jun '22

On Thu, 30 Jun 2022 10:03:47 +0800, Li kunyu wrote: > extra semicolons could be deleted. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] sound: delete a semicolon commit: d8d6253b36f55d199590ef908712fe52bb39ee97 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: samsung: change neo1973_audio from a global to static
by Tom Rix 30 Jun '22

30 Jun '22

sparse reports sound/soc/samsung/neo1973_wm8753.c:347:24: warning: symbol 'neo1973_audio' was not declared. Should it be static? neo1973_audio is only used in neo1973_wm8753.c, so it's storage class specifier should be static. Fixes: e26a2abcc246 ("ASoC: samsung: neo1973: turn into platform driver") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/samsung/neo1973_wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index c98b68567a89..e9f2334028bf 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -344,7 +344,7 @@ static int neo1973_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, &neo1973); } -struct platform_driver neo1973_audio = { +static struct platform_driver neo1973_audio = { .driver = { .name = "neo1973-audio", .pm = &snd_soc_pm_ops, -- 2.27.0

3 2

[PATCH 0/2] ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus()
by Kuninori Morimoto 30 Jun '22

30 Jun '22

Hi Mark Current ASoC has fixup both snd_soc_of_get_dai_link_cpus/codecs(). I guess cpu was copied from codec, but it is using "codec" naming everwhere in "cpu" function. It is strange, and thus, error case will be issue (It should call cpu function instead of codec). This patch tidyup it, and try to cleanup. [1/2] is for bug-fix, [2/2] is for new feature. It needs Tested-by/Reviewed-by/Acked-by from someone. Kuninori Morimoto (2): ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() ASoC: soc-core.c: share code for snd_soc_of_get_dai_link_cpus/codecs() sound/soc/soc-core.c | 146 ++++++++++++++++++++++--------------------- 1 file changed, 75 insertions(+), 71 deletions(-) -- 2.25.1

3 4

Re: [PATCH] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph()
by Mark Brown 30 Jun '22

30 Jun '22

On Thu, 30 Jun 2022 02:25:20 +0800, Jianglei Nie wrote: > q6apm_get_audioreach_graph() allocates a memory chunk for graph->graph > with audioreach_alloc_graph_pkt(). When idr_alloc() fails, graph->graph > is not released, which will lead to a memory leak. > > We can release the graph->graph with kfree() when idr_alloc() fails to > fix the memory leak. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph() commit: 0063ae55ccd2ade509be828762addcfccee52c4c 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 1/4] ASoC: tas2764: Add post reset delays
by Martin Povišer 30 Jun '22

30 Jun '22

Make sure there is at least 1 ms delay from reset to first command as is specified in the datasheet. This is a fix similar to commit 307f31452078 ("ASoC: tas2770: Insert post reset delay"). Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/codecs/tas2764.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 42f0c1e449ba..039bf1900880 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -42,10 +42,12 @@ static void tas2764_reset(struct tas2764_priv *tas2764) gpiod_set_value_cansleep(tas2764->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2764->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2764->component, TAS2764_SW_RST, TAS2764_RST); + usleep_range(1000, 2000); } static int tas2764_set_bias_level(struct snd_soc_component *component, @@ -107,8 +109,10 @@ static int tas2764_codec_resume(struct snd_soc_component *component) struct tas2764_priv *tas2764 = snd_soc_component_get_drvdata(component); int ret; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } ret = snd_soc_component_update_bits(component, TAS2764_PWR_CTRL, TAS2764_PWR_CTRL_MASK, @@ -501,8 +505,10 @@ static int tas2764_codec_probe(struct snd_soc_component *component) tas2764->component = component; - if (tas2764->sdz_gpio) + if (tas2764->sdz_gpio) { gpiod_set_value_cansleep(tas2764->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2764_reset(tas2764); -- 2.33.0

2 4

[PATCH] ASoC: dt-bindings: Fix description for msm8916
by Bryan O'Donoghue 30 Jun '22

30 Jun '22

For the existing msm8916 bindings the minimum reg/reg-names is 1 not 2. Similarly the minimum interrupt/interrupt-names is 1 not 2. Fixes: f3fc4fbfa2d2 ("ASoC: dt-bindings: Add SC7280 lpass cpu bindings") Signed-off-by: Bryan O'Donoghue <bryan.odonoghue(a)linaro.org> --- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index e9a533080b32d..ef18a572a1ff3 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -25,12 +25,12 @@ properties: - qcom,sc7280-lpass-cpu reg: - minItems: 2 + minItems: 1 maxItems: 6 description: LPAIF core registers reg-names: - minItems: 2 + minItems: 1 maxItems: 6 clocks: @@ -42,12 +42,12 @@ properties: maxItems: 10 interrupts: - minItems: 2 + minItems: 1 maxItems: 4 description: LPAIF DMA buffer interrupt interrupt-names: - minItems: 2 + minItems: 1 maxItems: 4 qcom,adsp: -- 2.36.1

3 2

Re: [PATCH] ASoC: qdsp6: fix potential memory leak in q6apm_get_audioreach_graph()
by Srinivas Kandagatla 30 Jun '22

30 Jun '22

On 29/06/2022 19:25, Jianglei Nie wrote: > q6apm_get_audioreach_graph() allocates a memory chunk for graph->graph > with audioreach_alloc_graph_pkt(). When idr_alloc() fails, graph->graph > is not released, which will lead to a memory leak. > > We can release the graph->graph with kfree() when idr_alloc() fails to > fix the memory leak. thanks for catching this, wondering did you hit this bug while testing or was it a some tooling that discovered this bug? > > Signed-off-by: Jianglei Nie <niejianglei2021(a)163.com> > --- Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > sound/soc/qcom/qdsp6/q6apm.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c > index f424d7aa389a..794019286c70 100644 > --- a/sound/soc/qcom/qdsp6/q6apm.c > +++ b/sound/soc/qcom/qdsp6/q6apm.c > @@ -75,6 +75,7 @@ static struct audioreach_graph *q6apm_get_audioreach_graph(struct q6apm *apm, ui > id = idr_alloc(&apm->graph_idr, graph, graph_id, graph_id + 1, GFP_KERNEL); > if (id < 0) { > dev_err(apm->dev, "Unable to allocate graph id (%d)\n", graph_id); > + kfree(graph->graph); > kfree(graph); > mutex_unlock(&apm->lock); > return ERR_PTR(id);

1 0

[PATCH 0/6] Add support of two Audio PLL source
by Shengjiu Wang 30 Jun '22

30 Jun '22

i.MX8MQ/MN/MM/MP platforms typically have 2 AUDIO PLLs being configured to handle 8kHz and 11kHz series audio rates. The patches implement the functionality to select at runtime the appropriate AUDIO PLL for root clock, if there is no two PLL registered, then no action taken. Shengjiu Wang (6): ASoC: fsl_utils: Add function to handle PLL clock source ASoC: fsl_spdif: Add support for PLL switch at runtime. ASoC: fsl_micfil: Add support for PLL switch at runtime ASoC: fsl_sai: Add support for PLL switch at runtime ASoC: dt-bindings: fsl_spdif: Add two PLL clock source ASoC: dt-bindings: fsl-sai: Add two PLL clock source .../devicetree/bindings/sound/fsl,spdif.yaml | 4 ++ .../devicetree/bindings/sound/fsl-sai.txt | 3 + sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/fsl_micfil.c | 41 +++++++++++ sound/soc/fsl/fsl_sai.c | 54 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 2 + sound/soc/fsl/fsl_spdif.c | 57 +++++++++++++-- sound/soc/fsl/fsl_utils.c | 69 +++++++++++++++++++ sound/soc/fsl/fsl_utils.h | 9 +++ 9 files changed, 237 insertions(+), 5 deletions(-) -- 2.17.1

3 8

[PATCH v2] pinctrl: qcom: sc7280: Fix compile bug
by Srinivasa Rao Mandadapu 30 Jun '22

30 Jun '22

Fix the compilation error, caused by updating constant variable. Hence remove redundant constant variable, which is no more useful as per new design. The issue is due to some unstaged changes. Fix it up. Fixes: 36fe26843d6d ("pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Reviewed-by: Stephen Boyd <swboyd(a)chromium.org> --- Changes Since V1: -- Fix typo errors in commit message. drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 5 +---- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c index b5d1b99..e97ce45 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c @@ -388,9 +388,6 @@ int lpi_pinctrl_probe(struct platform_device *pdev) pctrl->data = data; pctrl->dev = &pdev->dev; - data->is_clk_optional = of_property_read_bool(dev->of_node, - "qcom,adsp-bypass-mode"); - pctrl->clks[0].id = "core"; pctrl->clks[1].id = "audio"; @@ -404,7 +401,7 @@ int lpi_pinctrl_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(pctrl->slew_base), "Slew resource not provided\n"); - if (data->is_clk_optional) + if (of_property_read_bool(dev->of_node, "qcom,adsp-bypass-mode")) ret = devm_clk_bulk_get_optional(dev, MAX_LPI_NUM_CLKS, pctrl->clks); else ret = devm_clk_bulk_get(dev, MAX_LPI_NUM_CLKS, pctrl->clks); diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h index 759d5d8..afbac2a 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.h @@ -77,7 +77,6 @@ struct lpi_pinctrl_variant_data { int ngroups; const struct lpi_function *functions; int nfunctions; - bool is_clk_optional; }; int lpi_pinctrl_probe(struct platform_device *pdev); -- 2.7.4

2 1