diff options
| -rw-r--r-- | Documentation/devicetree/bindings/sound/fsl,audmix.txt | 50 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/fsl,audmix.yaml | 83 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/fsl,esai.txt | 68 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/fsl,esai.yaml | 116 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/wlf,wm8776.yaml | 41 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/wlf,wm8974.txt | 15 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/wlf,wm8974.yaml | 41 | ||||
| -rw-r--r-- | Documentation/devicetree/bindings/sound/wm8776.txt | 18 | ||||
| -rw-r--r-- | sound/soc/amd/Kconfig | 5 | ||||
| -rw-r--r-- | sound/soc/codecs/hdac_hda.c | 44 | ||||
| -rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 4 | ||||
| -rw-r--r-- | sound/soc/fsl/imx-es8328.c | 17 | ||||
| -rw-r--r-- | sound/soc/soc-dai.c | 2 |
13 files changed, 321 insertions, 183 deletions
diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt deleted file mode 100644 index 840b7e0d6a63..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt +++ /dev/null @@ -1,50 +0,0 @@ -NXP Audio Mixer (AUDMIX). - -The Audio Mixer is a on-chip functional module that allows mixing of two -audio streams into a single audio stream. Audio Mixer has two input serial -audio interfaces. These are driven by two Synchronous Audio interface -modules (SAI). Each input serial interface carries 8 audio channels in its -frame in TDM manner. Mixer mixes audio samples of corresponding channels -from two interfaces into a single sample. Before mixing, audio samples of -two inputs can be attenuated based on configuration. The output of the -Audio Mixer is also a serial audio interface. Like input interfaces it has -the same TDM frame format. This output is used to drive the serial DAC TDM -interface of audio codec and also sent to the external pins along with the -receive path of normal audio SAI module for readback by the CPU. - -The output of Audio Mixer can be selected from any of the three streams - - serial audio input 1 - - serial audio input 2 - - mixed audio - -Mixing operation is independent of audio sample rate but the two audio -input streams must have same audio sample rate with same number of channels -in TDM frame to be eligible for mixing. - -Device driver required properties: -================================= - - compatible : Compatible list, contains "fsl,imx8qm-audmix" - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "ipg" for register access. - - - power-domains : Must contain the phandle to AUDMIX power domain node - - - dais : Must contain a list of phandles to AUDMIX connected - DAIs. The current implementation requires two phandles - to SAI interfaces to be provided, the first SAI in the - list being used to route the AUDMIX output. - -Device driver configuration example: -====================================== - audmix: audmix@59840000 { - compatible = "fsl,imx8qm-audmix"; - reg = <0x0 0x59840000 0x0 0x10000>; - clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>; - clock-names = "ipg"; - power-domains = <&pd_audmix>; - dais = <&sai4>, <&sai5>; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml new file mode 100644 index 000000000000..9413b901cf77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Mixer (AUDMIX). + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: | + The Audio Mixer is a on-chip functional module that allows mixing of two + audio streams into a single audio stream. Audio Mixer has two input serial + audio interfaces. These are driven by two Synchronous Audio interface + modules (SAI). Each input serial interface carries 8 audio channels in its + frame in TDM manner. Mixer mixes audio samples of corresponding channels + from two interfaces into a single sample. Before mixing, audio samples of + two inputs can be attenuated based on configuration. The output of the + Audio Mixer is also a serial audio interface. Like input interfaces it has + the same TDM frame format. This output is used to drive the serial DAC TDM + interface of audio codec and also sent to the external pins along with the + receive path of normal audio SAI module for readback by the CPU. + + The output of Audio Mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - mixed audio + + Mixing operation is independent of audio sample rate but the two audio + input streams must have same audio sample rate with same number of channels + in TDM frame to be eligible for mixing. + +properties: + compatible: + const: fsl,imx8qm-audmix + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: ipg + + power-domains: + maxItems: 1 + + dais: + description: contain a list of phandles to AUDMIX connected DAIs. + $ref: /schemas/types.yaml#/definitions/phandle-array + minItems: 2 + items: + - description: the AUDMIX output + maxItems: 1 + - description: serial audio input 1 + maxItems: 1 + - description: serial audio input 2 + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + - power-domains + - dais + +unevaluatedProperties: false + +examples: + - | + audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x59840000 0x10000>; + clocks = <&amix_lpcg 0>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt deleted file mode 100644 index 90112ca1ff42..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Enhanced Serial Audio Interface (ESAI) Controller - -The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port -for serial communication with a variety of serial devices, including industry -standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and -other DSPs. It has up to six transmitters and four receivers. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-esai", - "fsl,vf610-esai", - "fsl,imx6ull-esai", - "fsl,imx8qm-esai", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock used to access registers - "extal" The esai baud clock for esai controller used to - derive HCK, SCK and FS. - "fsys" The system clock derived from ahb clock used to - derive HCK, SCK and FS. - "spba" The spba clock is required when ESAI is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - - - fsl,fifo-depth : The number of elements in the transmit and receive - FIFOs. This number is the maximum allowed value for - TFCR[TFWM] or RFCR[RFWM]. - - - fsl,esai-synchronous: This is a boolean property. If present, indicating - that ESAI would work in the synchronous mode, which - means all the settings for Receiving would be - duplicated from Transmission related registers. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -esai: esai@2024000 { - compatible = "fsl,imx35-esai"; - reg = <0x02024000 0x4000>; - interrupts = <0 51 0x04>; - clocks = <&clks 208>, <&clks 118>, <&clks 208>; - clock-names = "core", "extal", "fsys"; - dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; - dma-names = "rx", "tx"; - fsl,fifo-depth = <128>; - fsl,esai-synchronous; - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml new file mode 100644 index 000000000000..f167f1634d7e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,esai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Enhanced Serial Audio Interface (ESAI) Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang@nxp.com> + - Frank Li <Frank.Li@nxp.com> + +description: + The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port + for serial communication with a variety of serial devices, including industry + standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and + other DSPs. It has up to six transmitters and four receivers. + +properties: + compatible: + enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,imx8qm-esai + - fsl,vf610-esai + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + items: + - description: + The core clock used to access registers. + - description: + The esai baud clock for esai controller used to + derive HCK, SCK and FS. + - description: + The system clock derived from ahb clock used to + derive HCK, SCK and FS. + - description: + The spba clock is required when ESAI is placed as a + bus slave of the Shared Peripheral Bus and when two + or more bus masters (CPU, DMA or DSP) try to access + it. This property is optional depending on the SoC + design. + + clock-names: + minItems: 3 + items: + - const: core + - const: extal + - const: fsys + - const: spba + + dmas: + minItems: 2 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive + FIFOs. This number is the maximum allowed value for + TFCR[TFWM] or RFCR[RFWM]. + + fsl,esai-synchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: + This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which + means all the settings for Receiving would be + duplicated from Transmission related registers. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: + If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - fsl,fifo-depth + - fsl,esai-synchronous + +unevaluatedProperties: false + +examples: + - | + esai@2024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + big-endian; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml new file mode 100644 index 000000000000..7bbc96ee81be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8776 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8776 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8776"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt deleted file mode 100644 index 01d3a7c83419..000000000000 --- a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt +++ /dev/null @@ -1,15 +0,0 @@ -WM8974 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - compatible: "wlf,wm8974" - - reg: the I2C address or SPI chip select number of the device - -Examples: - -codec: wm8974@1a { - compatible = "wlf,wm8974"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml new file mode 100644 index 000000000000..d27300207c67 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8974 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8974 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt deleted file mode 100644 index 01173369c3ed..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8776.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8776 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8776" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8776: codec@1a { - compatible = "wlf,wm8776"; - reg = <0x1a>; -}; diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08..3508f5a96b75 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 6aa3223985be..29c88de5508b 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -230,7 +230,8 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); if (!format_val) { dev_err(dai->dev, - "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + "%s: invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + __func__, params_rate(params), params_channels(params), params_format(params), maxbps); @@ -266,14 +267,12 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct hdac_device *hdev; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; int ret = 0; hda_pvt = snd_soc_component_get_drvdata(component); - hdev = &hda_pvt->codec->core; pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); if (!pcm) return -EINVAL; @@ -286,7 +285,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream, stream, format_val, substream); if (ret < 0) - dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); return ret; } @@ -298,6 +297,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, struct hdac_hda_priv *hda_pvt; struct hda_pcm_stream *hda_stream; struct hda_pcm *pcm; + int ret; hda_pvt = snd_soc_component_get_drvdata(component); pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); @@ -308,7 +308,11 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + ret = hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + if (ret < 0) + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); + + return ret; } static void hdac_hda_dai_close(struct snd_pcm_substream *substream, @@ -367,7 +371,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, pcm_name = "HDMI 3"; break; default: - dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + dev_err(dai->dev, "%s: invalid dai id %d\n", __func__, dai->id); return NULL; } @@ -381,7 +385,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, } } - dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + dev_err(dai->dev, "%s: didn't find PCM for DAI %s\n", __func__, dai->name); return NULL; } @@ -411,7 +415,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } @@ -429,7 +433,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, hdev->addr, hcodec, true); if (ret < 0) { - dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to create hda codec %d\n", __func__, ret); goto error_no_pm; } @@ -446,7 +450,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (fw) { ret = snd_hda_load_patch(hcodec->bus, fw->size, fw->data); if (ret < 0) { - dev_err(&hdev->dev, "failed to load hda patch %d\n", ret); + dev_err(&hdev->dev, "%s: failed to load hda patch %d\n", __func__, ret); goto error_no_pm; } release_firmware(fw); @@ -470,13 +474,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); if (ret < 0) { - dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + dev_err(&hdev->dev, "%s: name failed %s\n", __func__, hcodec->preset->name); goto error_pm; } ret = snd_hdac_regmap_init(&hcodec->core); if (ret < 0) { - dev_err(&hdev->dev, "regmap init failed\n"); + dev_err(&hdev->dev, "%s: regmap init failed\n", __func__); goto error_pm; } @@ -484,16 +488,16 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (patch) { ret = patch(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "patch failed %d\n", ret); + dev_err(&hdev->dev, "%s: patch failed %d\n", __func__, ret); goto error_regmap; } } else { - dev_dbg(&hdev->dev, "no patch file found\n"); + dev_dbg(&hdev->dev, "%s: no patch file found\n", __func__); } ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + dev_err(&hdev->dev, "%s: unable to map pcms to dai %d\n", __func__, ret); goto error_patch; } @@ -501,8 +505,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (!is_hdmi_codec(hcodec)) { ret = snd_hda_codec_build_controls(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to create controls %d\n", - ret); + dev_err(&hdev->dev, "%s: unable to create controls %d\n", + __func__, ret); goto error_patch; } } @@ -548,7 +552,7 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return; } @@ -624,7 +628,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } snd_hdac_ext_bus_link_get(hdev->bus, hlink); @@ -640,7 +644,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { - dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to register HDA codec %d\n", __func__, ret); return ret; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab6ec1974807..4ca3a16f7ac0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1401,8 +1401,10 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, goto error_pcm; } else { ret = imx_pcm_dma_init(pdev); - if (ret) + if (ret) { + dev_err_probe(dev, ret, "Failed to init PCM DMA\n"); goto error_pcm; + } } return 0; diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..5b9648f3b087 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,12 +23,11 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { { - .gpio = -1, .name = "headset-gpio", .report = SND_JACK_HEADSET, .invert = 0, @@ -54,8 +53,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +63,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +173,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack", GPIOD_IN); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6f8773a8fc05..fefe394dce72 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -45,7 +45,7 @@ static inline int _soc_dai_ret(struct snd_soc_dai *dai, * @dai: DAI * @clk_id: DAI specific clock ID * @freq: new clock frequency in Hz - * @dir: new clock direction - input/output. + * @dir: new clock direction (SND_SOC_CLOCK_IN or SND_SOC_CLOCK_OUT) * * Configures the DAI master (MCLK) or system (SYSCLK) clocking. */ |