July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH][next] ALSA: hda_codec: Use flex_array_size() helper in memcpy()
by Gustavo A. R. Silva 30 Jul '20

30 Jul '20

Make use of the flex_array_size() helper to calculate the size of a flexible array member within an enclosing structure. This helper offers defense-in-depth against potential integer overflows and makes it explicitly clear that we are dealing with a flexible array member. Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 58c698f4d131..453c4ec77c48 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -113,7 +113,7 @@ static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, return -ENOMEM; p->len = len; p->nid = nid; - memcpy(p->conns, list, len * sizeof(hda_nid_t)); + memcpy(p->conns, list, flex_array_size(p, conns, len)); list_add(&p->list, &codec->conn_list); return 0; } -- 2.27.0

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Race condition when closing the Jack plugin
by GitHub issues - opened 29 Jul '20

29 Jul '20

alsa-project/alsa-plugins issue #11 was opened from jeberger: There is a potential race condition that can cause segmentation faults if the Jack process callback is called between the call to `snd_pcm_jack_stop` and `snd_pcm_jack_hw_free`. This was introduced by revision badd921. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/11 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH 1/2] dt-bindings: tlv320adcx140: Add GPO config and drive config
by Dan Murphy 29 Jul '20

29 Jul '20

Add properties for configuring the General Purpose Outputs (GPO). The GPOs. There are 2 settings for each GPO, configuration and the output drive type. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../bindings/sound/tlv320adcx140.yaml | 28 +++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 2e6ac5d2ee96..0950eb69f2b2 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -108,6 +108,32 @@ properties: maximum: 7 default: [0, 0, 0, 0] +patternProperties: + '^ti,gpo-config-[1-4]$': + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + Defines the configuration and output driver for the general purpose + output pins (GPO). These values are pairs, the first value is for the + configuration type and the second value is for the output drive type. + The array is defined as <GPO_CFG GPO_DRV> + + GPO output configuration can be one of the following: + + 0 - (default) disabled + 1 - GPOX is configured as a general-purpose output (GPO) + 2 - GPOX is configured as a device interrupt output (IRQ) + 3 - GPOX is configured as a secondary ASI output (SDOUT2) + 4 - GPOX is configured as a PDM clock output (PDMCLK) + + GPO output drive configuration for the GPO pins can be one of the following: + + 0d - (default) Hi-Z output + 1d - Drive active low and active high + 2d - Drive active low and weak high + 3d - Drive active low and Hi-Z + 4d - Drive weak low and active high + 5d - Drive Hi-Z and active high + required: - compatible - reg @@ -124,6 +150,8 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpo-config-1 = <0 0>; + ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; }; -- 2.28.0

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[PATCH] ASoC: tlv320adcx140: Fix various style errors and warnings
by Dan Murphy 29 Jul '20

29 Jul '20

Fix white space issues and remove else case where it was not needed. Convert "static const char *" to "static const char * const" Fixes: 689c7655b50 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 49dcdd72e5c6..938c5ef17e61 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -218,8 +218,8 @@ static const struct snd_kcontrol_new in4_resistor_controls[] = { }; /* Analog/Digital Selection */ -static const char *adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"}; -static const char *adcx140_analog_sel_text[] = {"Analog", "Line In"}; +static const char * const adcx140_mic_sel_text[] = {"Analog", "Line In", "Digital"}; +static const char * const adcx140_analog_sel_text[] = {"Analog", "Line In"}; static SOC_ENUM_SINGLE_DECL(adcx140_mic1p_enum, ADCX140_CH1_CFG0, 5, @@ -598,7 +598,7 @@ static int adcx140_reset(struct adcx140_priv *adcx140) gpiod_direction_output(adcx140->gpio_reset, 1); } else { ret = regmap_write(adcx140->regmap, ADCX140_SW_RESET, - ADCX140_RESET); + ADCX140_RESET); } /* 8.4.2: wait >= 10 ms after entering sleep mode. */ @@ -841,7 +841,7 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) goto out; - if(adcx140->supply_areg == NULL) + if (adcx140->supply_areg == NULL) sleep_cfg_val |= ADCX140_AREG_INTERNAL; ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val); @@ -942,8 +942,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (IS_ERR(adcx140->supply_areg)) { if (PTR_ERR(adcx140->supply_areg) == -EPROBE_DEFER) return -EPROBE_DEFER; - else - adcx140->supply_areg = NULL; + + adcx140->supply_areg = NULL; } else { ret = regulator_enable(adcx140->supply_areg); if (ret) { -- 2.28.0

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[PATCH] ASoC: Intel: Atom: use hardware counter to update hw_ptr
by Brent Lu 29 Jul '20

29 Jul '20

The ring buffer counter runs faster than hardware counter if the period size in hw_param is larger than 240. Although the differce is not much (around 2k frames), it causes false underrun in CRAS sometimes because it's using 256 frames as period size in hw_param. Using the hardware counter could provide precise hw_ptr to user space and avoid the false underrun in CRAS. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 7624953..1949ad9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -485,7 +485,6 @@ static inline int sst_calc_tstamp(struct intel_sst_drv *ctx, struct snd_pcm_substream *substream, struct snd_sst_tstamp *fw_tstamp) { - size_t delay_bytes, delay_frames; size_t buffer_sz; u32 pointer_bytes, pointer_samples; @@ -493,22 +492,14 @@ static inline int sst_calc_tstamp(struct intel_sst_drv *ctx, fw_tstamp->ring_buffer_counter); dev_dbg(ctx->dev, "mrfld hardware_counter %llu in bytes\n", fw_tstamp->hardware_counter); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - delay_bytes = (size_t) (fw_tstamp->ring_buffer_counter - - fw_tstamp->hardware_counter); - else - delay_bytes = (size_t) (fw_tstamp->hardware_counter - - fw_tstamp->ring_buffer_counter); - delay_frames = bytes_to_frames(substream->runtime, delay_bytes); + buffer_sz = snd_pcm_lib_buffer_bytes(substream); - div_u64_rem(fw_tstamp->ring_buffer_counter, buffer_sz, &pointer_bytes); + div_u64_rem(fw_tstamp->hardware_counter, buffer_sz, &pointer_bytes); pointer_samples = bytes_to_samples(substream->runtime, pointer_bytes); - dev_dbg(ctx->dev, "pcm delay %zu in bytes\n", delay_bytes); - info->buffer_ptr = pointer_samples / substream->runtime->channels; + info->pcm_delay = 0; - info->pcm_delay = delay_frames; dev_dbg(ctx->dev, "buffer ptr %llu pcm_delay rep: %llu\n", info->buffer_ptr, info->pcm_delay); return 0; -- 2.7.4

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MOTU Ultralite AVB: Modulo-8 output channel hopping
by Holger Marzen 28 Jul '20

28 Jul '20

Hi all, I am using the MOTU Ultralite AVB USB audio interface. With recent firmwares MOTU uses implicit feedback. Kernel 5.8-rc6 can handle it when I apply the attached patch with the MOTU signatures. But both with the old and the new firmware there is an occasional modulo 8 output channel hopping. What goes out to system:playback_1 in jackd ends on channel 1, 9 or 17 in the interface. Same for system:playback_2. I can force a hop when changing jackd's buffer size on the fly with jack_bufsize. One user reports this issue with ALSA clients, too, so it's probably no jackd issue. Since I never found this issue reported from a Windows or Mac user it may be buried deep in the ALSA kernel code. For me it seems that flaws/timeouts in the USB communication force this hopping in the ALSA kernel. Any ideas how to find out where the hopping happens? sound/usb/endpoint.c ? sound/usb/stream.c? Regards Holger

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[PATCH 1/2] dt-bindings: sound: add DT bindings for Microchip S/PDIF TX Controller
by Codrin Ciubotariu 28 Jul '20

28 Jul '20

This patch adds DT bindings for the new Microchip S/PDIF TX Controller embedded inside sama7g5 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- .../bindings/sound/mchp,spdiftx.yaml | 76 +++++++++++++++++++ 1 file changed, 76 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..8e2acfbbe621 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: https://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: https://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a + serial port compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + + dma-names: + const: tx + + pinctrl-0: true + + pinctrl-names: + const: default + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x100>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&spdiftx_clk>, <&spdiftx_gclk>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; -- 2.25.1

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[PATCH v2] dt-bindings: sound: convert Everest ES8316 binding to yaml
by Katsuhiro Suzuki 28 Jul '20

28 Jul '20

This patch converts Everest Semiconductor ES8316 low power audio CODEC binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- Changes in v2: - Change maintainers from Mark to Daniel and me --- .../bindings/sound/everest,es8316.txt | 23 --------- .../bindings/sound/everest,es8316.yaml | 50 +++++++++++++++++++ 2 files changed, 50 insertions(+), 23 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt deleted file mode 100644 index 1bf03c5f2af4..000000000000 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ /dev/null @@ -1,23 +0,0 @@ -Everest ES8316 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : should be "everest,es8316" - - reg : the I2C address of the device for I2C - -Optional properties: - - - clocks : a list of phandle, should contain entries for clock-names - - clock-names : should include as follows: - "mclk" : master clock (MCLK) of the device - -Example: - -es8316: codec@11 { - compatible = "everest,es8316"; - reg = <0x11>; - clocks = <&clks 10>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..3b752bba748b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Daniel Drake <drake(a)endlessm.com> + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; -- 2.27.0

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[PATCH 0/2] SOF Fixes for S0iX suspend/resume sequence
by Ranjani Sridharan 28 Jul '20

28 Jul '20

This set of patches is required for facilitating system S0ix entry when the DSP is in D0I3. This first patch adds the missing CORB/RIRB DMA stop and restart to the suspend/resume sequence along with powering up/down the links. The second patch ensures that the FW traces are disabled when the system enters S0ix with the DSP in D0I3. Marcin Rajwa (2): ASoC: SOF: Intel: fix the suspend procedure to support s0ix entry ASoC: SOF: Intel: disable traces when switching to S0Ix D0I3 sound/soc/sof/intel/hda-dsp.c | 48 ++++++++++++++++++++++++++++++++--- 1 file changed, 44 insertions(+), 4 deletions(-) -- 2.25.1

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[PATCH] ASoC: bcm2835: Silence clk_get() error on -EPROBE_DEFER
by Seung-Woo Kim 28 Jul '20

28 Jul '20

Silence clk_get() error with dev_dbg() on -EPROBE_DEFER. Signed-off-by: Seung-Woo Kim <sw0312.kim(a)samsung.com> --- sound/soc/bcm/bcm2835-i2s.c | 9 ++++++--- 1 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index d80b570..dc34fe1 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -841,9 +841,12 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) dev->clk_prepared = false; dev->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { - dev_err(&pdev->dev, "could not get clk: %ld\n", - PTR_ERR(dev->clk)); - return PTR_ERR(dev->clk); + ret = PTR_ERR(dev->clk); + if (ret == -EPROBE_DEFER) + dev_dbg(&pdev->dev, "could not get clk: %d\n", ret); + else + dev_err(&pdev->dev, "could not get clk: %d\n", ret); + return ret; } /* Request ioarea */ -- 1.7.4.1

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