April 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: ab8500: add range to usleep_range
by Nicholas Mc Guire 07 Apr '19

07 Apr '19

Providing a range for usleep_range() allows the hrtimer subsystem to coalesce timers - as this delay has no upper limit anyway (interrupts or context switch is possible) it should not hurt to extend this from 2 to 2-4 milliseconds. Signed-off-by: Nicholas Mc Guire <hofrat(a)opentech.at> --- Problem located with an experimental coccinelle script Basically usleep_range() with min == max never makes much sense notably in non-atomic context. Patch was compile tested with: u8500_defconfig (implies CONFIG_SND_SOC_AB8500_CODEC=y) (with some sparse warnings about not implemented system calls) Patch is against 5.1-rc3 (localversion-next is next=20190405) sound/soc/codecs/ab8500-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 87616b1..19e7f03 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1062,10 +1062,10 @@ static void anc_iir(struct snd_soc_component *component, unsigned int bnk, snd_soc_component_update_bits(component, AB8500_ANCCONF1, BIT(AB8500_ANCCONF1_ANCIIRINIT), BIT(AB8500_ANCCONF1_ANCIIRINIT)); - usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY*2); snd_soc_component_update_bits(component, AB8500_ANCCONF1, BIT(AB8500_ANCCONF1_ANCIIRINIT), 0); - usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY*2); } else { snd_soc_component_update_bits(component, AB8500_ANCCONF1, BIT(AB8500_ANCCONF1_ANCIIRUPDATE), -- 2.1.4

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[alsa-devel] [PATCH v6 0/6] drm/bridge: sii902x: HDMI-audio support and some fixes
by Jyri Sarha 06 Apr '19

06 Apr '19

Changes since v5: - Add "#sound-dai-cells" to binding document. Configure audio if this property exsists. - Make sil,i2s-data-lanes property optional with default behaviour The already reviewed patches are in front. Not reviewed still: - "dt-bindings: display: sii902x: Add HDMI audio bindings" - "drm/bridge: sii902x: Implement HDMI audio support" The previous round: https://patchwork.kernel.org/cover/10865215/ Jyri Sarha (5): drm/bridge: sii902x: Set output mode to HDMI or DVI according to EDID drm/bridge: sii902x: pixel clock unit is 10kHz instead of 1kHz dt-bindings: display: sii902x: Remove trailing white space dt-bindings: display: sii902x: Add HDMI audio bindings drm/bridge: sii902x: Implement HDMI audio support Tomi Valkeinen (1): drm/bridge: sii902x: add input_bus_flags .../bindings/display/bridge/sii902x.txt | 39 +- drivers/gpu/drm/bridge/sii902x.c | 486 +++++++++++++++++- 2 files changed, 517 insertions(+), 8 deletions(-) -- Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 1/2] ASoC: core: conditionally increase module refcount on component open
by Ranjani Sridharan 05 Apr '19

05 Apr '19

Recently, for Intel platforms the "ignore_module_refcount" field was introduced for the component driver. In order to avoid a deadlock preventing the PCI modules from being removed even when the card was idle, the refcounts were not incremented for the device driver module during component probe. However, this change introduced a nasty side effect: the device driver module can be unloaded while a pcm stream is open. This patch proposes to change the field to be renamed as "module_get_upon_open". When this field is set, the module refcount should be incremented on pcm open amd decremented upon pcm close. This will enable modules to be removed when no PCM playback/capture happens and prevent removal when the component is actually in use. Also, align with the skylake component driver with the new name. Fixes: b450b878('ASoC: core: don't increase component module refcount unconditionally' Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/soc.h | 9 +++++++-- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/soc-core.c | 4 ++-- 3 files changed, 10 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1e2be35ed36f..482b4ea87c3c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -802,8 +802,13 @@ struct snd_soc_component_driver { int probe_order; int remove_order; - /* signal if the module handling the component cannot be removed */ - unsigned int ignore_module_refcount:1; + /* + * signal if the module handling the component should not be removed + * if a pcm is open. Setting this would prevent the module + * refcount being incremented in probe() but allow it be incremented + * when a pcm is opened and decremented when it is closed. + */ + unsigned int module_get_upon_open:1; /* bits */ unsigned int idle_bias_on:1; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 56099db8f86d..ee53d047c3bd 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1468,7 +1468,7 @@ static const struct snd_soc_component_driver skl_component = { .ops = &skl_platform_ops, .pcm_new = skl_pcm_new, .pcm_free = skl_pcm_free, - .ignore_module_refcount = 1, /* do not increase the refcount in core */ + .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; int skl_platform_register(struct device *dev) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6f4842977b8d..e78534bbd092 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -947,7 +947,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; - if (!component->driver->ignore_module_refcount) + if (!component->driver->module_get_upon_open) module_put(component->dev->driver->owner); } @@ -1381,7 +1381,7 @@ static int soc_probe_component(struct snd_soc_card *card, return 0; } - if (!component->driver->ignore_module_refcount && + if (!component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; -- 2.17.1

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[alsa-devel] [PATCH 0/4] soundwire: few trival fixes and cleanups.
by Srinivas Kandagatla 05 Apr '19

05 Apr '19

Hi Vinod, Here are few trivial updates to soundwire which I found while trying out Qualcomm WCD934x codecs. Thanks, Srinivas Kandagatla (4): soundwire: add module_sdw_driver helper macro soundwire: cdns: fix check for number of messages soundwire: bus: check if pm runtime is enabled before calling soundwire: remove unused struct sdw_bus_conf definition drivers/soundwire/bus.c | 16 ++++++++++------ drivers/soundwire/cadence_master.c | 2 +- include/linux/soundwire/sdw.h | 15 --------------- include/linux/soundwire/sdw_type.h | 11 +++++++++++ 4 files changed, 22 insertions(+), 22 deletions(-) -- 2.21.0

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[alsa-devel] [PATCH] pcm: null: Do not allow a period size of 0
by twischer＠de.adit-jv.com 05 Apr '19

05 Apr '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Some applications do not expect that get_period_size_min() could return 0. Therefore these applications cannot use the null plugin without this patch. Due to there is no use case for having a period size of 0 this patch disallows a period size of 0 when using the null plugin. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/pcm_null.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index ff61624..c95fad8 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -261,7 +261,18 @@ static snd_pcm_sframes_t snd_pcm_null_mmap_commit(snd_pcm_t *pcm, static int snd_pcm_null_hw_refine(snd_pcm_t *pcm ATTRIBUTE_UNUSED, snd_pcm_hw_params_t *params) { - int err = snd_pcm_hw_refine_soft(pcm, params); + int err; + + /* Do not return a period size of 0 because for example portaudio cannot + * handle it. In addition such small buffers would only increase the CPU + * load but there is no benefit when throwing the samples anyway away. + */ + err = _snd_pcm_hw_param_set_min(params, SND_PCM_HW_PARAM_PERIOD_SIZE, + 32, 0); + if (err < 0) + return err; + + err = snd_pcm_hw_refine_soft(pcm, params); params->info = SND_PCM_INFO_MMAP | SND_PCM_INFO_MMAP_VALID | SND_PCM_INFO_RESUME | SND_PCM_INFO_PAUSE; params->fifo_size = 0; -- 2.7.4

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[alsa-devel] [PATCH 00/10] ALSA: aloop: Support selection of snd_timer
by twischer＠de.adit-jv.com 05 Apr '19

05 Apr '19

With the following patch set any sound timer can be used as timer source for the aloop devices if selected. It can be select by the additional timer_source kernel module parameter: -2 the jiffies timer will be used as done without this patch (default) -1 the sound timer will be selected depending on the linked sound cards (snd_pcm_link()). The sound timer of the first linked card which is not a aloop device nor a dummy device will be used. 0..N Select the sound timer by the card number. This patch set is required for forwarding audio data between a HW sound card and an aloop device without the usage of an asynchronous sample rate converter. Best regards, Timo

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[alsa-devel] [PATCH] ALSA: seq: Fix OOB-reads from strlcpy
by Zubin Mithra 05 Apr '19

05 Apr '19

When ioctl calls are made with non-null-terminated userspace strings, strlcpy causes an OOB-read from within strlen. Fix by changing to use strscpy instead. Signed-off-by: Zubin Mithra <zsm(a)chromium.org> --- sound/core/seq/seq_clientmgr.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 7d4640d1fe9fb..38e7deab63847 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1252,7 +1252,7 @@ static int snd_seq_ioctl_set_client_info(struct snd_seq_client *client, /* fill the info fields */ if (client_info->name[0]) - strlcpy(client->name, client_info->name, sizeof(client->name)); + strscpy(client->name, client_info->name, sizeof(client->name)); client->filter = client_info->filter; client->event_lost = client_info->event_lost; @@ -1530,7 +1530,7 @@ static int snd_seq_ioctl_create_queue(struct snd_seq_client *client, void *arg) /* set queue name */ if (!info->name[0]) snprintf(info->name, sizeof(info->name), "Queue-%d", q->queue); - strlcpy(q->name, info->name, sizeof(q->name)); + strscpy(q->name, info->name, sizeof(q->name)); snd_use_lock_free(&q->use_lock); return 0; @@ -1592,7 +1592,7 @@ static int snd_seq_ioctl_set_queue_info(struct snd_seq_client *client, queuefree(q); return -EPERM; } - strlcpy(q->name, info->name, sizeof(q->name)); + strscpy(q->name, info->name, sizeof(q->name)); queuefree(q); return 0; -- 2.21.0.392.gf8f6787159e-goog

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[alsa-devel] [PATCH 1/3] ASoC: wcd9335: fix a leaked reference by adding missing of_node_put
by Wen Yang 05 Apr '19

05 Apr '19

The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> (commit_signer:1/11=9%,authored:1/11=9%) Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/codecs/wcd9335.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 981f88a..a04a7ce 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5188,6 +5188,7 @@ static int wcd9335_slim_status(struct slim_device *sdev, wcd->slim = sdev; wcd->slim_ifc_dev = of_slim_get_device(sdev->ctrl, ifc_dev_np); + of_node_put(ifc_dev_np); if (!wcd->slim_ifc_dev) { dev_err(dev, "Unable to get SLIM Interface device\n"); return -EINVAL; -- 2.9.5

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[alsa-devel] [PATCH V3] ASoC: fsl_esai: Support synchronous mode
by S.j. Wang 05 Apr '19

05 Apr '19

In ESAI synchronous mode, the clock is generated by Tx, So we should always set registers of Tx which relate with the bit clock and frame clock generation (TCCR, TCR, ECR), even there is only Rx is working. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changes in v3 - fix the indentation sound/soc/fsl/fsl_esai.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3623aa9a6f2e..c7410bbfd2af 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -218,7 +218,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); struct clk *clksrc = esai_priv->extalclk; - bool tx = clk_id <= ESAI_HCKT_EXTAL; + bool tx = (clk_id <= ESAI_HCKT_EXTAL || esai_priv->synchronous); bool in = dir == SND_SOC_CLOCK_IN; u32 ratio, ecr = 0; unsigned long clk_rate; @@ -253,7 +253,7 @@ static int fsl_esai_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, ecr |= ESAI_ECR_ETI; /* fall through */ case ESAI_HCKR_EXTAL: - ecr |= ESAI_ECR_ERI; + ecr |= esai_priv->synchronous ? ESAI_ECR_ETI : ESAI_ECR_ERI; break; default: return -EINVAL; @@ -537,10 +537,18 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, bclk = params_rate(params) * slot_width * esai_priv->slots; - ret = fsl_esai_set_bclk(dai, tx, bclk); + ret = fsl_esai_set_bclk(dai, esai_priv->synchronous || tx, bclk); if (ret) return ret; + mask = ESAI_xCR_xSWS_MASK; + val = ESAI_xCR_xSWS(slot_width, width); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + /* Recording in synchronous mode needs to set TCR also */ + if (!tx && esai_priv->synchronous) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, val); + /* Use Normal mode to support monaural audio */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), ESAI_xCR_xMOD_MASK, params_channels(params) > 1 ? @@ -556,10 +564,9 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val); - mask = ESAI_xCR_xSWS_MASK | (tx ? ESAI_xCR_PADC : 0); - val = ESAI_xCR_xSWS(slot_width, width) | (tx ? ESAI_xCR_PADC : 0); - - regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), mask, val); + if (tx) + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_PADC, ESAI_xCR_PADC); /* Remove ESAI personal reset by configuring ESAI_PCRC and ESAI_PRRC */ regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, -- 1.9.1

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[alsa-devel] [PATCH 0/7] ASoC: cs42l51: add some features
by Olivier Moysan 05 Apr '19

05 Apr '19

This patchset add support of following features to Cirrus CS42L51 audio codec driver: - reset - regulators - master mode - power management - adc volume control Olivier Moysan (7): ASoC: dt-bindings: update cs42l51 bindings ASoC: cs42l51: add regulator management ASoC: cs42l51: add reset management ASoC: cs42l51: add support of master mode ASoC: cs42l51: change mic bias DAPM ASoC: cs42l51: add power management ASoC: cs42l51: add adc volume control .../devicetree/bindings/sound/cs42l51.txt | 16 ++ sound/soc/codecs/cs42l51-i2c.c | 13 +- sound/soc/codecs/cs42l51.c | 217 ++++++++++++++++++++- sound/soc/codecs/cs42l51.h | 3 + 4 files changed, 242 insertions(+), 7 deletions(-) -- 2.7.4

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