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

[alsa-devel] [PATCH] ASoC: SOF: control: return true when kcontrol values change
by Pierre-Louis Bossart 18 Oct '19

18 Oct '19

From: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> All the kcontrol put() functions are currently returning 0 when successful. This does not go well with alsamixer as it does not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks when values change for (some of) the sof kcontrols. This patch fixes that by returning true for volume, switch and enum type kcontrols when values do change in put(). Signed-off-by: Dragos Tarcatu <dragos_tarcatu(a)mentor.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 2c4abd406c4f..7baf7f1507c3 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -91,13 +91,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -107,8 +110,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -136,11 +138,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } if (scontrol->led_ctl.use_led) @@ -154,7 +160,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -182,11 +188,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -197,7 +207,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, -- 2.20.1

2 1

[alsa-devel] [PATCH v3 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 18 Oct '19

18 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Changes from v2: - rebase upon to "don't use snd_pcm_ops" series https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156170.h… - fix sparse errors https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156328.h… - use "reg" for SHM binding https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156657.h… Tzung-Bi Shih (10): WIP: platform/chrome: cros_ec: remove unused EC feature WIP: ASoC: cros_ec_codec: refactor I2S RX WIP: ASoC: cros_ec_codec: extract DMIC EC command from I2S RX WIP: platform/chrome: cros_ec: add common commands for EC codec WIP: ASoC: cros_ec_codec: read max DMIC gain from EC codec WIP: ASoC: dt-bindings: cros_ec_codec: add SHM bindings WIP: ASoC: cros_ec_codec: support WoV WIP: ASoC: mediatek: mt6358: support WoV WIP: ASoC: dt-bindings: mt8183: add ec-codec WIP: ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1128 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1296 insertions(+), 325 deletions(-) -- 2.23.0.700.g56cf767bdb-goog

5 28

[alsa-devel] [PATCH 1/2] ASoc: Add Texas Instruments TAS2562 amplifier binding
by Dan Murphy 18 Oct '19

18 Oct '19

Add the DT binding for the TAS2562 amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 +++++++++++++++++++ 1 file changed, 34 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt new file mode 100644 index 000000000000..658e1fb18a99 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -0,0 +1,34 @@ +Texas Instruments TAS2562 Smart PA + +The TAS2562 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - compatible: - Should contain "ti,tas2562". + - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. + - ti,imon-slot-no:- TDM TX current sense time slot. + +Optional properties: +- interrupt-parent: phandle to the interrupt controller which provides + the interrupt. +- interrupts: (GPIO) interrupt to which the chip is connected. +- shut-down: GPIO used to control the state of the device. + +Examples: +tas2562@4c { + #address-cells = <1>; + #size-cells = <0>; + compatible = "ti,tas2562"; + reg = <0x4c>; + + interrupt-parent = <&gpio1>; + interrupts = <14>; + + shut-down = <&gpio1 15 0>; + ti,imon-slot-no = <0>; +}; + -- 2.22.0.214.g8dca754b1e

3 4

[alsa-devel] [PATCH][next] ASoC: rt1011: fix spelling mistake "temperture" -> "temperature"
by Colin King 18 Oct '19

18 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_dbg message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1011.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index ad049cfddcb0..dcd397a83cb4 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2373,7 +2373,7 @@ static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) device_property_read_u32(dev, "realtek,r0_calib", &rt1011->r0_calib); - dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + dev_dbg(dev, "%s: r0_calib: 0x%x, temperature_calib: 0x%x", __func__, rt1011->r0_calib, rt1011->temperature_calib); return 0; -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: add a check for devm_clk_get
by Chuhong Yuan 18 Oct '19

18 Oct '19

aic32x4_set_dai_sysclk misses a check for devm_clk_get and may miss the failure. Add a check to fix it. Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/tlv320aic32x4.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 68165de1c8de..b4e9a6c73f90 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -573,6 +573,9 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct clk *pll; pll = devm_clk_get(component->dev, "pll"); + if (IS_ERR(pll)) + return PTR_ERR(pll); + mclk = clk_get_parent(pll); return clk_set_rate(mclk, freq); -- 2.20.1

2 1

[alsa-devel] [PATCH v3] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 18 Oct '19

18 Oct '19

The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <mripard(a)kernel.org> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 267 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------ 2 files changed, 267 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..b8f89c7258eb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,267 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + items: + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + items: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.23.0

3 2

[alsa-devel] [PATCH v6] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 18 Oct '19

18 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: integrated fix for minor spelling issues] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- v6: - integrated fix for minor spelling issues - collected tags --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 138 ++++++++++++++++++ 2 files changed, 138 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..53e3bad4178c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,138 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel (5.1) I2S with + secondary FIFO, s/w reset control and internal mux for root clock + source. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel (5.1) I2S for + playback, stereo channel capture, secondary FIFO using internal + or external DMA, s/w reset control, internal mux for root clock + source and 7.1 channel TDM support for playback; TDM (Time division + multiplexing) is to allow transfer of multiple channel audio data on + single data line. + + samsung,exynos7-i2s: with all the available features of Exynos5 I2S. + Exynos7 I2S has 7.1 channel TDM support for capture, secondary FIFO + with only external DMA and more number of root clock sampling + frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. Exynos7 I2S1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for I2S0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for I2S1 and I2S2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the I2S bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clock. I2S0 has internal mux to select the source + of root clock and I2S1 and I2S2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for I2S0 + - const: i2s_cdclk0 + - items: # for I2S1 + - const: i2s_cdclk1 + - items: # for I2S2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + subsystem (used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; -- 2.17.1

2 1

[alsa-devel] [PATCH 00/12] ALSA: firewire: configure interval of hw IRQ via ALSA PCM interface
by Takashi Sakamoto 18 Oct '19

18 Oct '19

Hi, ALSA IEC 61883-1/6 packet streaming engine controls OHCI 1394 controller to generate hardware IRQ for fixed number (=16) of isoc packets. Additionally, the size of packet buffer is fixed for triple of interval of hardware IRQ (=48). This commit allows ALSA PCM applications to configure these parameters by ALSA PCM interface. As a result, the size of PCM period is available in the range between 125 usec and 64 msec (or 32 msec). The hardware IRQ is scheduled for isoc packet which includes event across PCM period boundary. Takashi Sakamoto (12): ALSA: firewire-lib: use variable size of queue for isoc packets instead of fixed size ALSA: bebob: share PCM buffer size for both direction ALSA: fireworks: share PCM buffer size for both direction ALSA: oxfw: share PCM buffer size for both direction ALSA: dice: share PCM buffer size for both direction ALSA: firewire-digi00x: share PCM buffer size for both direction ALSA: firewire-tascam: share PCM buffer size for both direction ALSA: firewire-motu: share PCM buffer size for both direction ALSA: fireface: share PCM buffer size for both direction ALSA: firewire-lib: schedule hardware IRQ according to the size of PCM period ALSA: firewire-lib: tune the maximum available size of PCM period ALSA: firewire-lib: tune the minimum available size of PCM period sound/firewire/amdtp-stream.c | 154 ++++++++++++++------ sound/firewire/amdtp-stream.h | 9 +- sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 5 +- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 20 ++- sound/firewire/dice/dice-stream.c | 5 +- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 12 +- sound/firewire/digi00x/digi00x-stream.c | 5 +- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 13 +- sound/firewire/fireface/ff-stream.c | 5 +- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 12 +- sound/firewire/fireworks/fireworks_stream.c | 5 +- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 12 +- sound/firewire/motu/motu-stream.c | 5 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 17 ++- sound/firewire/oxfw/oxfw-stream.c | 5 +- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 12 +- sound/firewire/tascam/tascam-stream.c | 5 +- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 266 insertions(+), 85 deletions(-) -- 2.20.1

2 13

[alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 17 Oct '19

17 Oct '19

For P2P output, the output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 40 +++++++++++++++++++++++++++++++--------- 1 file changed, 31 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..44d05ec28bd3 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; if (div[IN] == 0) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); @@ -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; - else - div[OUT] = clk_get_rate(clk) / outrate; + /* + * When P2P mode, output rate should align with the out samplerate. + * if set too high output rate, there will be lots of Overload. + * When M2M mode, output rate should also need to align with the out + * samplerate, but M2M must use less time to achieve good performance. + */ + clk_rate = clk_get_rate(clk); + if (p2p || !ideal) { + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; + } else { + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); + div[OUT] = clk_rate; + } if (div[OUT] == 0) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || + rem[IN] != 0 || rem[OUT] != 0)) { + pair_err("The divider can't be used for non ideal mode\n"); + return -EINVAL; + } + + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +582,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, true); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v3] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 17 Oct '19

17 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties, added list of values for widgets, minor other fixes] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 -------- .../bindings/sound/samsung,tm2-audio.yaml | 99 +++++++++++++++++++ 2 files changed, 99 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..c9178d928ad4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are: + the WM5110's pins: + "HPOUT1L", + "HPOUT1R", + "HPOUT2L", + "HPOUT2R", + "HPOUT3L", + "HPOUT3R", + MAX98504's pins: + "SPKOUT" + and the jacks on the board: + "HP", + "SPK", + "RCV", + "Main Mic", + "Sub Mic", + "Third Mic", + "Headset Mic". + +required: + - compatible + - model + - i2s-controller + - audio-codec + - audio-amplifier + - mic-bias-gpios + - samsung,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-codec = <&wm5110>, <&hdmi>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

2 1