diff options
58 files changed, 1149 insertions, 1706 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/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index d70c55f17df7..c11aaf8079fb 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -118,6 +118,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( * which do not use devicetree. * @process: Callback used to apply processing on samples transferred from/to * user space. + * @name: Component name. If null, dev_name will be used. * @compat_filter_fn: Will be used as the filter function when requesting a * channel for platforms which do not use devicetree. The filter parameter * will be the DAI's DMA data. @@ -143,6 +144,7 @@ struct snd_dmaengine_pcm_config { int (*process)(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, unsigned long bytes); + const char *name; dma_filter_fn compat_filter_fn; struct device *dma_dev; const char *chan_names[SNDRV_PCM_STREAM_LAST + 1]; 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/es8326.c b/sound/soc/codecs/es8326.c index 15289dadafea..e8526844337d 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -292,11 +292,6 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, - 4, 7, 0, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, - 0, 7, 0, 0), - SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -316,9 +311,6 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, - {"HPOR", NULL, "HPOR Supply"}, - {"HPOL", NULL, "HPOL Supply"}, - {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; @@ -1072,12 +1064,13 @@ static int es8326_suspend(struct snd_soc_component *component) es8326->calibrated = false; regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); - regcache_mark_dirty(es8326->regmap); /* reset register value to default */ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); usleep_range(1000, 3000); regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); + + regcache_mark_dirty(es8326->regmap); return 0; } @@ -1163,8 +1156,13 @@ static int es8326_set_jack(struct snd_soc_component *component, static void es8326_remove(struct snd_soc_component *component) { + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + es8326_disable_jack_detect(component); es8326_set_bias_level(component, SND_SOC_BIAS_OFF); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); + usleep_range(1000, 3000); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); } static const struct snd_soc_component_driver soc_component_dev_es8326 = { @@ -1236,6 +1234,29 @@ static int es8326_i2c_probe(struct i2c_client *i2c) &es8326_dai, 1); } + +static void es8326_i2c_shutdown(struct i2c_client *i2c) +{ + struct snd_soc_component *component; + struct es8326_priv *es8326; + + es8326 = i2c_get_clientdata(i2c); + component = es8326->component; + dev_dbg(component->dev, "Enter into %s\n", __func__); + cancel_delayed_work_sync(&es8326->jack_detect_work); + cancel_delayed_work_sync(&es8326->button_press_work); + + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); + usleep_range(1000, 3000); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); + +} + +static void es8326_i2c_remove(struct i2c_client *i2c) +{ + es8326_i2c_shutdown(i2c); +} + static const struct i2c_device_id es8326_i2c_id[] = { {"es8326", 0 }, {} @@ -1265,6 +1286,8 @@ static struct i2c_driver es8326_i2c_driver = { .of_match_table = of_match_ptr(es8326_of_match), }, .probe = es8326_i2c_probe, + .shutdown = es8326_i2c_shutdown, + .remove = es8326_i2c_remove, .id_table = es8326_i2c_id, }; module_i2c_driver(es8326_i2c_driver); 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_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 00852f174a69..bc41a0666856 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -135,7 +135,6 @@ static struct snd_soc_dai_driver fsl_rpmsg_dai = { static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", - .legacy_dai_naming = 1, }; static const struct fsl_rpmsg_soc_data imx7ulp_data = { @@ -190,19 +189,40 @@ MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); static int fsl_rpmsg_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct snd_soc_dai_driver *dai_drv; + const char *dai_name; struct fsl_rpmsg *rpmsg; int ret; + dai_drv = devm_kzalloc(&pdev->dev, sizeof(struct snd_soc_dai_driver), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + memcpy(dai_drv, &fsl_rpmsg_dai, sizeof(fsl_rpmsg_dai)); + rpmsg = devm_kzalloc(&pdev->dev, sizeof(struct fsl_rpmsg), GFP_KERNEL); if (!rpmsg) return -ENOMEM; rpmsg->soc_data = of_device_get_match_data(&pdev->dev); - fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; - fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; - fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; - fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + if (rpmsg->soc_data) { + dai_drv->playback.rates = rpmsg->soc_data->rates; + dai_drv->capture.rates = rpmsg->soc_data->rates; + dai_drv->playback.formats = rpmsg->soc_data->formats; + dai_drv->capture.formats = rpmsg->soc_data->formats; + } + + /* Use rpmsg channel name as cpu dai name */ + ret = of_property_read_string(np, "fsl,rpmsg-channel-name", &dai_name); + if (ret) { + if (ret == -EINVAL) { + dai_name = "rpmsg-audio-channel"; + } else { + dev_err(&pdev->dev, "Failed to get rpmsg channel name: %d!\n", ret); + return ret; + } + } + dai_drv->name = dai_name; if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; @@ -236,21 +256,10 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &fsl_rpmsg_dai, 1); + dai_drv, 1); if (ret) goto err_pm_disable; - rpmsg->card_pdev = platform_device_register_data(&pdev->dev, - "imx-audio-rpmsg", - PLATFORM_DEVID_AUTO, - NULL, - 0); - if (IS_ERR(rpmsg->card_pdev)) { - dev_err(&pdev->dev, "failed to register rpmsg card\n"); - ret = PTR_ERR(rpmsg->card_pdev); - goto err_pm_disable; - } - return 0; err_pm_disable: 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-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c index 289e47c03d40..38aafb8954c7 100644 --- a/sound/soc/fsl/imx-audio-rpmsg.c +++ b/sound/soc/fsl/imx-audio-rpmsg.c @@ -12,6 +12,7 @@ */ struct imx_audio_rpmsg { struct platform_device *rpmsg_pdev; + struct platform_device *card_pdev; }; static int imx_audio_rpmsg_cb(struct rpmsg_device *rpdev, void *data, int len, @@ -87,14 +88,24 @@ static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev) /* Register platform driver for rpmsg routine */ data->rpmsg_pdev = platform_device_register_data(&rpdev->dev, - IMX_PCM_DRV_NAME, - PLATFORM_DEVID_AUTO, + rpdev->id.name, + PLATFORM_DEVID_NONE, NULL, 0); if (IS_ERR(data->rpmsg_pdev)) { dev_err(&rpdev->dev, "failed to register rpmsg platform.\n"); ret = PTR_ERR(data->rpmsg_pdev); } + data->card_pdev = platform_device_register_data(&rpdev->dev, + "imx-audio-rpmsg", + PLATFORM_DEVID_AUTO, + rpdev->id.name, + strlen(rpdev->id.name) + 1); + if (IS_ERR(data->card_pdev)) { + dev_err(&rpdev->dev, "failed to register rpmsg card.\n"); + ret = PTR_ERR(data->card_pdev); + } + return ret; } @@ -105,6 +116,9 @@ static void imx_audio_rpmsg_remove(struct rpmsg_device *rpdev) if (data->rpmsg_pdev) platform_device_unregister(data->rpmsg_pdev); + if (data->card_pdev) + platform_device_unregister(data->card_pdev); + dev_info(&rpdev->dev, "audio rpmsg driver is removed\n"); } @@ -113,6 +127,7 @@ static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = { { .name = "rpmsg-micfil-channel" }, { }, }; +MODULE_DEVICE_TABLE(rpmsg, imx_audio_rpmsg_id_table); static struct rpmsg_driver imx_audio_rpmsg_driver = { .drv.name = "imx_audio_rpmsg", @@ -126,5 +141,5 @@ module_rpmsg_driver(imx_audio_rpmsg_driver); MODULE_DESCRIPTION("Freescale SoC Audio RPMSG interface"); MODULE_AUTHOR("Shengjiu Wang <shengjiu.wang@nxp.com>"); -MODULE_ALIAS("platform:imx_audio_rpmsg"); +MODULE_ALIAS("rpmsg:imx_audio_rpmsg"); MODULE_LICENSE("GPL v2"); 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/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index fb9244c1e9c5..b84d1dfddba2 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -732,9 +732,6 @@ static int imx_rpmsg_pcm_probe(struct platform_device *pdev) goto fail; } - /* platform component name is used by machine driver to link with */ - component->name = info->rpdev->id.name; - #ifdef CONFIG_DEBUG_FS component->debugfs_prefix = "rpmsg"; #endif @@ -822,9 +819,17 @@ static const struct dev_pm_ops imx_rpmsg_pcm_pm_ops = { imx_rpmsg_pcm_resume) }; +static const struct platform_device_id imx_rpmsg_pcm_id_table[] = { + { .name = "rpmsg-audio-channel" }, + { .name = "rpmsg-micfil-channel" }, + { }, +}; +MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table); + static struct platform_driver imx_pcm_rpmsg_driver = { .probe = imx_rpmsg_pcm_probe, .remove_new = imx_rpmsg_pcm_remove, + .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, .pm = &imx_rpmsg_pcm_pm_ops, diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index e5bd63dab10c..0f1ad7ad7d27 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -108,10 +108,8 @@ static int imx_rpmsg_late_probe(struct snd_soc_card *card) static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; - struct device *dev = pdev->dev.parent; - /* rpmsg_pdev is the platform device for the rpmsg node that probed us */ - struct platform_device *rpmsg_pdev = to_platform_device(dev); - struct device_node *np = rpmsg_pdev->dev.of_node; + struct snd_soc_dai *cpu_dai; + struct device_node *np = NULL; struct of_phandle_args args; const char *platform_name; struct imx_rpmsg *data; @@ -127,10 +125,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev) goto fail; } - ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); - if (ret) - dev_warn(&pdev->dev, "no reserved DMA memory\n"); - data->dai.cpus = &dlc[0]; data->dai.num_cpus = 1; data->dai.platforms = &dlc[1]; @@ -152,6 +146,23 @@ static int imx_rpmsg_probe(struct platform_device *pdev) */ data->dai.ignore_pmdown_time = 1; + data->dai.cpus->dai_name = pdev->dev.platform_data; + cpu_dai = snd_soc_find_dai(data->dai.cpus); + if (!cpu_dai) { + ret = -EPROBE_DEFER; + goto fail; + } + np = cpu_dai->dev->of_node; + if (!np) { + dev_err(&pdev->dev, "failed to parse CPU DAI device node\n"); + ret = -ENODEV; + goto fail; + } + + ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); + if (ret) + dev_warn(&pdev->dev, "no reserved DMA memory\n"); + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { @@ -170,7 +181,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->sysclk = clk_get_rate(clk); } - data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); if (!of_property_read_string(np, "fsl,rpmsg-channel-name", &platform_name)) data->dai.platforms->name = platform_name; else diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 28f254eb0d03..282256d18cc6 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -54,76 +54,13 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_create_dapm_routes(struct device *dev, int ssp_port, int tdm_slot, - struct snd_soc_dapm_route **routes, int *num_routes) -{ - struct snd_soc_dapm_route *dr; - const int num_dr = 2; - - dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); - if (!dr) - return -ENOMEM; - - dr[0].sink = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot)); - dr[0].source = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", " Tx", ssp_port, tdm_slot)); - if (!dr[0].sink || !dr[0].source) - return -ENOMEM; - - dr[1].sink = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", " Rx", ssp_port, tdm_slot)); - dr[1].source = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot)); - if (!dr[1].sink || !dr[1].source) - return -ENOMEM; - - *routes = dr; - *num_routes = num_dr; - - return 0; -} - -static int avs_create_dapm_widgets(struct device *dev, int ssp_port, int tdm_slot, - struct snd_soc_dapm_widget **widgets, int *num_widgets) -{ - struct snd_soc_dapm_widget *dw; - const int num_dw = 2; - - dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL); - if (!dw) - return -ENOMEM; - - dw[0].id = snd_soc_dapm_hp; - dw[0].reg = SND_SOC_NOPM; - dw[0].name = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot)); - if (!dw[0].name) - return -ENOMEM; - - dw[1].id = snd_soc_dapm_mic; - dw[1].reg = SND_SOC_NOPM; - dw[1].name = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot)); - if (!dw[1].name) - return -ENOMEM; - - *widgets = dw; - *num_widgets = num_dw; - - return 0; -} - static int avs_i2s_test_probe(struct platform_device *pdev) { - struct snd_soc_dapm_widget *widgets; - struct snd_soc_dapm_route *routes; struct snd_soc_dai_link *dai_link; struct snd_soc_acpi_mach *mach; struct snd_soc_card *card; struct device *dev = &pdev->dev; const char *pname; - int num_routes, num_widgets; int ssp_port, tdm_slot, ret; mach = dev_get_platdata(dev); @@ -156,26 +93,10 @@ static int avs_i2s_test_probe(struct platform_device *pdev) return ret; } - ret = avs_create_dapm_routes(dev, ssp_port, tdm_slot, &routes, &num_routes); - if (ret) { - dev_err(dev, "Failed to create dapm routes: %d\n", ret); - return ret; - } - - ret = avs_create_dapm_widgets(dev, ssp_port, tdm_slot, &widgets, &num_widgets); - if (ret) { - dev_err(dev, "Failed to create dapm widgets: %d\n", ret); - return ret; - } - card->dev = dev; card->owner = THIS_MODULE; card->dai_link = dai_link; card->num_links = 1; - card->dapm_routes = routes; - card->num_dapm_routes = num_routes; - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; card->fully_routed = true; ret = snd_soc_fixup_dai_links_platform_name(card, pname); diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 4a0e136835ff..abb87bb88fff 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -37,8 +37,6 @@ static const struct snd_kcontrol_new card_controls[] = { static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_SPK("Left Speaker", NULL), SND_SOC_DAPM_SPK("Right Speaker", NULL), - SND_SOC_DAPM_SPK("DP1", NULL), - SND_SOC_DAPM_SPK("DP2", NULL), }; static const struct snd_soc_dapm_route card_base_routes[] = { @@ -158,7 +156,7 @@ static int avs_ssm4567_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = "avs_ssm4567-adi"; + card->name = "avs_ssm4567"; card->dev = dev; card->owner = THIS_MODULE; card->dai_link = dai_link; @@ -172,7 +170,6 @@ static int avs_ssm4567_probe(struct platform_device *pdev) card->dapm_routes = card_base_routes; card->num_dapm_routes = ARRAY_SIZE(card_base_routes); card->fully_routed = true; - card->disable_route_checks = true; ret = snd_soc_fixup_dai_links_platform_name(card, pname); if (ret) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 18ac3ce0752e..e5df64fec319 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -624,6 +624,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH select SND_SOC_MAX98357A select SND_SOC_MAX98373_I2C select SND_SOC_DMIC + select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_SSP_COMMON help diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 540f7a29310a..3fe3f38c6cb6 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -768,6 +768,7 @@ static struct snd_soc_card broxton_audio_card = { .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index c0eb65c14aa9..afc499be8db2 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -574,6 +574,7 @@ static struct snd_soc_card broxton_rt298 = { .dapm_routes = broxton_rt298_map, .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 657e4658234c..4098b2d32f9b 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -649,6 +649,8 @@ static int geminilake_audio_probe(struct platform_device *pdev) card = &glk_audio_card_rt5682_m98357a; card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, ctx); + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + card->disable_route_checks = true; /* override platform name, if required */ mach = pdev->dev.platform_data; diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index a5d8965303a8..9dbc15f9d1c9 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -639,6 +639,7 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .dapm_routes = kabylake_map, .num_dapm_routes = ARRAY_SIZE(kabylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 98c11ec0adc0..e662da5af83b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -1036,6 +1036,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1054,6 +1055,7 @@ static struct snd_soc_card kbl_audio_card_max98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1071,6 +1073,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1088,6 +1091,7 @@ static struct snd_soc_card kbl_audio_card_max98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 30e0aca161cd..894d127c482a 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -518,6 +518,7 @@ static struct snd_soc_card kabylake_audio_card_rt5660 = { .dapm_routes = kabylake_rt5660_map, .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 9071b1f1cbd0..646e8ff8e961 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -966,6 +966,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -982,6 +983,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663 = { .dapm_routes = kabylake_5663_map, .num_dapm_routes = ARRAY_SIZE(kabylake_5663_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 178fe9c37df6..924d5d1de03a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -791,6 +791,7 @@ static struct snd_soc_card kabylake_audio_card = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6e172719c979..4aa7fd2a05e4 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -227,6 +227,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; hda_soc_card.dev = &pdev->dev; + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + hda_soc_card.disable_route_checks = true; if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0e7025834594..e4630c33176e 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -654,6 +654,7 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index c59c60e28091..9a8044274908 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -523,6 +523,7 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 088894ff4165..a5135be94f32 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -74,6 +74,11 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) * DAI Link Helpers */ +enum sof_dmic_be_type { + SOF_DMIC_01, + SOF_DMIC_16K, +}; + /* DEFAULT_LINK_ORDER: the order used in sof_rt5682 */ #define DEFAULT_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_CODEC, \ SOF_LINK_DMIC01, \ @@ -97,13 +102,13 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; -int sof_intel_board_set_codec_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec codec_type, int ssp_codec) +static int set_ssp_codec_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_ssp_codec codec_type, + int ssp_codec) { struct snd_soc_dai_link_component *cpus; - dev_dbg(dev, "link %d: codec %s, ssp %d\n", be_id, + dev_dbg(dev, "link %d: ssp codec %s, ssp %d\n", be_id, sof_ssp_get_codec_name(codec_type), ssp_codec); /* link name */ @@ -144,11 +149,9 @@ int sof_intel_board_set_codec_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_codec_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_dmic_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_dmic_be_type be_type) +static int set_dmic_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_dmic_be_type be_type) { struct snd_soc_dai_link_component *cpus; @@ -196,16 +199,14 @@ int sof_intel_board_set_dmic_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_dmic_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_intel_hdmi_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int hdmi_id, bool idisp_codec) +static int set_idisp_hdmi_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int hdmi_id, bool idisp_codec) { struct snd_soc_dai_link_component *cpus, *codecs; - dev_dbg(dev, "link %d: intel hdmi, hdmi id %d, idisp codec %d\n", - be_id, hdmi_id, idisp_codec); + dev_dbg(dev, "link %d: idisp hdmi %d, idisp codec %d\n", be_id, hdmi_id, + idisp_codec); /* link name */ link->name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", hdmi_id); @@ -256,11 +257,9 @@ int sof_intel_board_set_intel_hdmi_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_intel_hdmi_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_ssp_amp_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec amp_type, int ssp_amp) +static int set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_ssp_codec amp_type, int ssp_amp) { struct snd_soc_dai_link_component *cpus; @@ -298,11 +297,9 @@ int sof_intel_board_set_ssp_amp_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_ssp_amp_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_bt_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_bt) +static int set_bt_offload_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int ssp_bt) { struct snd_soc_dai_link_component *cpus; @@ -341,11 +338,9 @@ int sof_intel_board_set_bt_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_bt_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_hdmi_in_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_hdmi) +static int set_hdmi_in_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int ssp_hdmi) { struct snd_soc_dai_link_component *cpus; @@ -383,7 +378,6 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_hdmi_in_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); static int calculate_num_links(struct sof_card_private *ctx) { @@ -427,6 +421,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, int ret; int ssp_hdmi_in = 0; unsigned long link_order, link; + unsigned long link_ids, be_id; num_links = calculate_num_links(ctx); @@ -440,22 +435,34 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, else link_order = DEFAULT_LINK_ORDER; - dev_dbg(dev, "create dai links, link_order 0x%lx\n", link_order); + if (ctx->link_id_overwrite) + link_ids = ctx->link_id_overwrite; + else + link_ids = 0; + + dev_dbg(dev, "create dai links, link_order 0x%lx, id_overwrite 0x%lx\n", + link_order, link_ids); while (link_order) { link = link_order & SOF_LINK_ORDER_MASK; link_order >>= SOF_LINK_ORDER_SHIFT; + if (ctx->link_id_overwrite) { + be_id = link_ids & SOF_LINK_IDS_MASK; + link_ids >>= SOF_LINK_IDS_SHIFT; + } else { + /* use array index as link id */ + be_id = idx; + } + switch (link) { case SOF_LINK_CODEC: /* headphone codec */ if (ctx->codec_type == CODEC_NONE) continue; - ret = sof_intel_board_set_codec_link(dev, &links[idx], - idx, - ctx->codec_type, - ctx->ssp_codec); + ret = set_ssp_codec_link(dev, &links[idx], be_id, + ctx->codec_type, ctx->ssp_codec); if (ret) { dev_err(dev, "fail to set codec link, ret %d\n", ret); @@ -471,8 +478,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], - idx, SOF_DMIC_01); + ret = set_dmic_link(dev, &links[idx], be_id, SOF_DMIC_01); if (ret) { dev_err(dev, "fail to set dmic01 link, ret %d\n", ret); @@ -487,8 +493,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], - idx, SOF_DMIC_16K); + ret = set_dmic_link(dev, &links[idx], be_id, + SOF_DMIC_16K); if (ret) { dev_err(dev, "fail to set dmic16k link, ret %d\n", ret); @@ -500,10 +506,9 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, case SOF_LINK_IDISP_HDMI: /* idisp HDMI */ for (i = 1; i <= ctx->hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, - &links[idx], - idx, i, - ctx->hdmi.idisp_codec); + ret = set_idisp_hdmi_link(dev, &links[idx], + be_id, i, + ctx->hdmi.idisp_codec); if (ret) { dev_err(dev, "fail to set hdmi link, ret %d\n", ret); @@ -511,6 +516,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } idx++; + be_id++; } break; case SOF_LINK_AMP: @@ -518,10 +524,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (ctx->amp_type == CODEC_NONE) continue; - ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], - idx, - ctx->amp_type, - ctx->ssp_amp); + ret = set_ssp_amp_link(dev, &links[idx], be_id, + ctx->amp_type, ctx->ssp_amp); if (ret) { dev_err(dev, "fail to set amp link, ret %d\n", ret); @@ -536,8 +540,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (!ctx->bt_offload_present) continue; - ret = sof_intel_board_set_bt_link(dev, &links[idx], idx, - ctx->ssp_bt); + ret = set_bt_offload_link(dev, &links[idx], be_id, + ctx->ssp_bt); if (ret) { dev_err(dev, "fail to set bt link, ret %d\n", ret); @@ -549,10 +553,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, case SOF_LINK_HDMI_IN: /* HDMI-In */ for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, - &links[idx], - idx, - ssp_hdmi_in); + ret = set_hdmi_in_link(dev, &links[idx], be_id, + ssp_hdmi_in); if (ret) { dev_err(dev, "fail to set hdmi-in link, ret %d\n", ret); @@ -560,6 +562,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } idx++; + be_id++; } break; case SOF_LINK_NONE: @@ -584,6 +587,49 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +struct sof_card_private * +sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk) +{ + struct sof_card_private *ctx; + + dev_dbg(dev, "create ctx, board_quirk 0x%lx\n", board_quirk); + + ctx = devm_kzalloc(dev, sizeof(struct sof_card_private), GFP_KERNEL); + if (!ctx) + return NULL; + + ctx->codec_type = sof_ssp_detect_codec_type(dev); + ctx->amp_type = sof_ssp_detect_amp_type(dev); + + ctx->dmic_be_num = 2; + ctx->hdmi_num = (board_quirk & SOF_NUM_IDISP_HDMI_MASK) >> + SOF_NUM_IDISP_HDMI_SHIFT; + /* default number of HDMI DAI's */ + if (!ctx->hdmi_num) + ctx->hdmi_num = 3; + + /* port number/mask of peripherals attached to ssp interface */ + if (ctx->codec_type != CODEC_NONE) + ctx->ssp_codec = (board_quirk & SOF_SSP_PORT_CODEC_MASK) >> + SOF_SSP_PORT_CODEC_SHIFT; + + if (ctx->amp_type != CODEC_NONE) + ctx->ssp_amp = (board_quirk & SOF_SSP_PORT_AMP_MASK) >> + SOF_SSP_PORT_AMP_SHIFT; + + if (board_quirk & SOF_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; + ctx->ssp_bt = (board_quirk & SOF_SSP_PORT_BT_OFFLOAD_MASK) >> + SOF_SSP_PORT_BT_OFFLOAD_SHIFT; + } + + ctx->ssp_mask_hdmi_in = (board_quirk & SOF_SSP_MASK_HDMI_CAPTURE_MASK) >> + SOF_SSP_MASK_HDMI_CAPTURE_SHIFT; + + return ctx; +} +EXPORT_SYMBOL_NS(sof_intel_board_get_ctx, SND_SOC_INTEL_SOF_BOARD_HELPERS); + struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais) { diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index f42d5d640321..2f27ad8726f8 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -10,6 +10,44 @@ #include "sof_hdmi_common.h" #include "sof_ssp_common.h" +/* + * Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine + * drivers + */ + +/* SSP port number for headphone codec: 3 bits */ +#define SOF_SSP_PORT_CODEC_SHIFT 8 +#define SOF_SSP_PORT_CODEC_MASK (GENMASK(10, 8)) +#define SOF_SSP_PORT_CODEC(quirk) \ + (((quirk) << SOF_SSP_PORT_CODEC_SHIFT) & SOF_SSP_PORT_CODEC_MASK) + +/* SSP port number for speaker amplifier: 3 bits */ +#define SOF_SSP_PORT_AMP_SHIFT 11 +#define SOF_SSP_PORT_AMP_MASK (GENMASK(13, 11)) +#define SOF_SSP_PORT_AMP(quirk) \ + (((quirk) << SOF_SSP_PORT_AMP_SHIFT) & SOF_SSP_PORT_AMP_MASK) + +/* SSP port number for BT audio offload: 3 bits */ +#define SOF_SSP_PORT_BT_OFFLOAD_SHIFT 14 +#define SOF_SSP_PORT_BT_OFFLOAD_MASK (GENMASK(16, 14)) +#define SOF_SSP_PORT_BT_OFFLOAD(quirk) \ + (((quirk) << SOF_SSP_PORT_BT_OFFLOAD_SHIFT) & SOF_SSP_PORT_BT_OFFLOAD_MASK) + +/* SSP port mask for HDMI capture: 6 bits */ +#define SOF_SSP_MASK_HDMI_CAPTURE_SHIFT 17 +#define SOF_SSP_MASK_HDMI_CAPTURE_MASK (GENMASK(22, 17)) +#define SOF_SSP_MASK_HDMI_CAPTURE(quirk) \ + (((quirk) << SOF_SSP_MASK_HDMI_CAPTURE_SHIFT) & SOF_SSP_MASK_HDMI_CAPTURE_MASK) + +/* Number of idisp HDMI BE link: 3 bits */ +#define SOF_NUM_IDISP_HDMI_SHIFT 23 +#define SOF_NUM_IDISP_HDMI_MASK (GENMASK(25, 23)) +#define SOF_NUM_IDISP_HDMI(quirk) \ + (((quirk) << SOF_NUM_IDISP_HDMI_SHIFT) & SOF_NUM_IDISP_HDMI_MASK) + +/* Board uses BT audio offload */ +#define SOF_BT_OFFLOAD_PRESENT BIT(26) + enum { SOF_LINK_NONE = 0, SOF_LINK_CODEC, @@ -33,6 +71,31 @@ enum { (((k6) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 5)) | \ (((k7) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 6))) +#define SOF_LINK_IDS_MASK (0xF) +#define SOF_LINK_IDS_SHIFT (4) + +#define SOF_LINK_IDS(k1, k2, k3, k4, k5, k6, k7) \ + ((((k1) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 0)) | \ + (((k2) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 1)) | \ + (((k3) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 2)) | \ + (((k4) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 3)) | \ + (((k5) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 4)) | \ + (((k6) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 5)) | \ + (((k7) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 6))) + +/* + * sof_da7219_private: private data for da7219 machine driver + * + * @is_jsl_board: true for JSL boards + * @mclk_en: true for mclk pin is connected + * @pll_bypass: true for PLL bypass mode + */ +struct sof_da7219_private { + bool is_jsl_board; + bool mclk_en; + bool pll_bypass; +}; + /* * sof_rt5682_private: private data for rt5682 machine driver * @@ -61,6 +124,8 @@ struct sof_rt5682_private { * @codec_link: pointer to headset codec dai link * @amp_link: pointer to speaker amplifier dai link * @link_order_overwrite: custom DAI link order + * @link_id_overwrite: custom DAI link ID + * @da7219: private data for da7219 machine driver * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -84,39 +149,23 @@ struct sof_card_private { struct snd_soc_dai_link *amp_link; unsigned long link_order_overwrite; + /* + * A variable stores id for all BE DAI links, use SOF_LINK_IDS macro to + * build the value; use DAI link array index as id if zero. + */ + unsigned long link_id_overwrite; union { + struct sof_da7219_private da7219; struct sof_rt5682_private rt5682; }; }; -enum sof_dmic_be_type { - SOF_DMIC_01, - SOF_DMIC_16K, -}; - int sof_intel_board_card_late_probe(struct snd_soc_card *card); int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, struct sof_card_private *ctx); - -int sof_intel_board_set_codec_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec codec_type, int ssp_codec); -int sof_intel_board_set_dmic_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_dmic_be_type be_type); -int sof_intel_board_set_intel_hdmi_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int hdmi_id, bool idisp_codec); -int sof_intel_board_set_ssp_amp_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec amp_type, int ssp_amp); -int sof_intel_board_set_bt_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_bt); -int sof_intel_board_set_hdmi_in_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_hdmi); +struct sof_card_private * +sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk); struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais); diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 323b86c42ef9..40ecfeaa1d26 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -22,23 +22,6 @@ #include "../common/soc-intel-quirks.h" #include "sof_board_helpers.h" #include "sof_maxim_common.h" -#include "sof_ssp_common.h" - -#define SOF_CS42L42_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_CS42L42_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_CS42L42_SSP_AMP_SHIFT 4 -#define SOF_CS42L42_SSP_AMP_MASK (GENMASK(6, 4)) -#define SOF_CS42L42_SSP_AMP(quirk) \ - (((quirk) << SOF_CS42L42_SSP_AMP_SHIFT) & SOF_CS42L42_SSP_AMP_MASK) -#define SOF_CS42L42_NUM_HDMIDEV_SHIFT 7 -#define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) -#define SOF_CS42L42_NUM_HDMIDEV(quirk) \ - (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_BT_OFFLOAD_PRESENT BIT(25) -#define SOF_CS42L42_SSP_BT_SHIFT 26 -#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) -#define SOF_CS42L42_SSP_BT(quirk) \ - (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) static struct snd_soc_jack_pin jack_pins[] = { { @@ -52,7 +35,7 @@ static struct snd_soc_jack_pin jack_pins[] = { }; /* Default: SSP2 */ -static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2); +static unsigned long sof_cs42l42_quirk = SOF_SSP_PORT_CODEC(2); static int sof_cs42l42_init(struct snd_soc_pcm_runtime *rtd) { @@ -229,48 +212,26 @@ static int sof_audio_probe(struct platform_device *pdev) struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_cs42l42_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_cs42l42_quirk); + if (!ctx) + return -ENOMEM; if (soc_intel_is_glk()) { ctx->dmic_be_num = 1; - ctx->hdmi_num = 3; /* overwrite the DAI link order for GLK boards */ ctx->link_order_overwrite = GLK_LINK_ORDER; - } else { - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_cs42l42_quirk & SOF_CS42L42_NUM_HDMIDEV_MASK) >> - SOF_CS42L42_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; } if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); - - /* port number of peripherals attached to ssp interface */ - ctx->ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> - SOF_CS42L42_SSP_BT_SHIFT; - - ctx->ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> - SOF_CS42L42_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; - - if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_cs42l42, ctx); if (ret) @@ -293,21 +254,21 @@ static int sof_audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { .name = "glk_cs4242_mx98357a", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(2) | - SOF_CS42L42_SSP_AMP(1)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(1)), }, { .name = "jsl_cs4242_mx98360a", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | - SOF_CS42L42_SSP_AMP(1)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "adl_mx98360a_cs4242", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | - SOF_CS42L42_SSP_AMP(1) | - SOF_CS42L42_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_PRESENT | - SOF_CS42L42_SSP_BT(2)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_SSP_PORT_BT_OFFLOAD(2)), }, { } }; @@ -329,4 +290,3 @@ MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>"); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 6eb5a6144e97..fd3a7be993c1 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -15,35 +15,25 @@ #include <sound/soc-acpi.h> #include <sound/sof.h> #include "../../codecs/da7219.h" -#include "hda_dsp_common.h" -#include "sof_hdmi_common.h" +#include "sof_board_helpers.h" #include "sof_maxim_common.h" -#include "sof_ssp_common.h" -/* Board Quirks */ -#define SOF_DA7219_JSL_BOARD BIT(2) +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_DA7219_JSL_BOARD BIT(0) +#define SOF_DA7219_MCLK_EN BIT(1) #define DIALOG_CODEC_DAI "da7219-hifi" -struct card_private { - struct snd_soc_jack headset_jack; - struct sof_hdmi_private hdmi; - enum sof_ssp_codec codec_type; - enum sof_ssp_codec amp_type; - - unsigned int pll_bypass:1; -}; - static int platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; - struct card_private *ctx = snd_soc_card_get_drvdata(card); + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai *codec_dai; int ret = 0; - if (ctx->pll_bypass) + if (ctx->da7219.pll_bypass) return ret; /* PLL SRM mode */ @@ -74,8 +64,6 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Line Out"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), }; static const struct snd_soc_dapm_widget widgets[] = { @@ -83,14 +71,9 @@ static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), - - SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -102,9 +85,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, { "Line Out", NULL, "Platform Clock" }, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, }; static struct snd_soc_jack_pin jack_pins[] = { @@ -124,7 +104,7 @@ static struct snd_soc_jack_pin jack_pins[] = { static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; struct snd_soc_jack *jack = &ctx->headset_jack; @@ -147,7 +127,8 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Use PLL bypass mode if MCLK is available, be sure to set the * frequency of MCLK to 12.288 or 24.576MHz on topology side. */ - if (mclk_rate == 12288000 || mclk_rate == 24576000) { + if (ctx->da7219.mclk_en && + (mclk_rate == 12288000 || mclk_rate == 24576000)) { /* PLL bypass mode */ dev_dbg(rtd->dev, "pll bypass mode, mclk rate %d\n", mclk_rate); @@ -157,7 +138,7 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ctx->pll_bypass = 1; + ctx->da7219.pll_bypass = true; } /* @@ -188,6 +169,13 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static void da7219_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int max98373_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -222,215 +210,11 @@ static const struct snd_soc_ops max98373_ops = { .hw_params = max98373_hw_params, }; -static int hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - - ctx->hdmi.hdmi_comp = dai->component; - - return 0; -} - static int card_late_probe(struct snd_soc_card *card) { - struct card_private *ctx = snd_soc_card_get_drvdata(card); - - if (!ctx->hdmi.idisp_codec) - return 0; - - if (!ctx->hdmi.hdmi_comp) - return -EINVAL; - - return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp); + return sof_intel_board_card_late_probe(card); } -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); - -SND_SOC_DAILINK_DEF(ssp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(dummy_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai"))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(dmic16k_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(idisp4_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp4 Pin"))); -SND_SOC_DAILINK_DEF(idisp4_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi4"))); - -SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */ - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -static struct snd_soc_dai_link jsl_dais[] = { - /* Back End DAI links */ - { - .name = "SSP1-Codec", - .id = 0, - .ignore_pmdown_time = 1, - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, /* IV feedback */ - SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform), - }, - { - .name = "SSP0-Codec", - .id = 1, - .no_pcm = 1, - .init = da7219_codec_init, - .ignore_pmdown_time = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - .name = "dmic16k", - .id = 6, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - } -}; - -static struct snd_soc_dai_link adl_dais[] = { - /* Back End DAI links */ - { - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .init = da7219_codec_init, - .ignore_pmdown_time = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 1, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "dmic16k", - .id = 2, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - .name = "iDisp4", - .id = 6, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp4_pin, idisp4_codec, platform), - }, - { - .name = "SSP1-Codec", - .id = 7, - .no_pcm = 1, - .dpcm_playback = 1, - /* feedback stream or firmware-generated echo reference */ - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform), - }, - { - .name = "SSP2-BT", - .id = 8, - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, dummy_codec, platform), - }, -}; - static struct snd_soc_card card_da7219 = { .name = "da7219", /* the sof- prefix is added by the core */ .owner = THIS_MODULE, @@ -444,28 +228,101 @@ static struct snd_soc_card card_da7219 = { .late_probe = card_late_probe, }; +static struct snd_soc_dai_link_component da7219_component[] = { + { + .name = "i2c-DLGS7219:00", + .dai_name = DIALOG_CODEC_DAI, + } +}; + +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) +{ + int ret; + + ret = sof_intel_board_set_dai_link(dev, card, ctx); + if (ret) + return ret; + + if (!ctx->codec_link) { + dev_err(dev, "codec link not available"); + return -EINVAL; + } + + /* codec-specific fields for headphone codec */ + ctx->codec_link->codecs = da7219_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(da7219_component); + ctx->codec_link->init = da7219_codec_init; + ctx->codec_link->exit = da7219_codec_exit; + + if (ctx->amp_type == CODEC_NONE) + return 0; + + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; + } + + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_MAX98360A: + max_98360a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98373: + ctx->amp_link->codecs = max_98373_components; + ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); + ctx->amp_link->init = max_98373_spk_codec_init; + if (ctx->da7219.is_jsl_board) { + ctx->amp_link->ops = &max98373_ops; /* use local ops */ + } else { + /* TBD: implement the amp for later platform */ + dev_err(dev, "max98373 not support yet\n"); + return -EINVAL; + } + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; + } + + return 0; +} + +#define JSL_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + static int audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; - struct card_private *ctx; + struct sof_card_private *ctx; unsigned long board_quirk = 0; - int ret, amp_idx; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; + int ret; if (pdev->id_entry && pdev->id_entry->driver_data) board_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, board_quirk); + if (!ctx) + return -ENOMEM; if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; if (board_quirk & SOF_DA7219_JSL_BOARD) { + ctx->da7219.is_jsl_board = true; + + /* overwrite the DAI link order for JSL boards */ + ctx->link_order_overwrite = JSL_LINK_ORDER; + /* backward-compatible with existing devices */ switch (ctx->amp_type) { case CODEC_MAX98360A: @@ -480,46 +337,30 @@ static int audio_probe(struct platform_device *pdev) default: break; } - - dai_links = jsl_dais; - amp_idx = 0; - - card_da7219.num_links = ARRAY_SIZE(jsl_dais); - } else { - dai_links = adl_dais; - amp_idx = 7; - - card_da7219.num_links = ARRAY_SIZE(adl_dais); } - dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk); + if (board_quirk & SOF_DA7219_MCLK_EN) + ctx->da7219.mclk_en = true; + + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &card_da7219, ctx); + if (ret) + return ret; - /* speaker amp */ + /* update codec_conf */ switch (ctx->amp_type) { - case CODEC_MAX98360A: - max_98360a_dai_link(&dai_links[amp_idx]); - break; case CODEC_MAX98373: - dai_links[amp_idx].codecs = max_98373_components; - dai_links[amp_idx].num_codecs = ARRAY_SIZE(max_98373_components); - dai_links[amp_idx].init = max_98373_spk_codec_init; - if (board_quirk & SOF_DA7219_JSL_BOARD) { - dai_links[amp_idx].ops = &max98373_ops; /* use local ops */ - } else { - /* TBD: implement the amp for later platform */ - dev_err(&pdev->dev, "max98373 not support yet\n"); - return -EINVAL; - } - max_98373_set_codec_conf(&card_da7219); break; + case CODEC_MAX98360A: + case CODEC_NONE: + /* no codec conf required */ + break; default: dev_err(&pdev->dev, "invalid amp type %d\n", ctx->amp_type); return -EINVAL; } - card_da7219.dai_link = dai_links; - card_da7219.dev = &pdev->dev; ret = snd_soc_fixup_dai_links_platform_name(&card_da7219, @@ -534,16 +375,28 @@ static int audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { - .name = "jsl_mx98373_da7219", - .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD), + .name = "jsl_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { - .name = "jsl_mx98360_da7219", - .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD), + .name = "adl_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { - .name = "adl_mx98360_da7219", - /* no quirk needed for this board */ + .name = "rpl_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -564,6 +417,5 @@ MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver for Dialog codec"); MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 719c2fbaf515..23fe8b4015cc 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -24,27 +24,8 @@ #include "sof_realtek_common.h" #include "sof_maxim_common.h" #include "sof_nuvoton_common.h" -#include "sof_ssp_common.h" - -#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_NAU8825_SSP_AMP_SHIFT 4 -#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) -#define SOF_NAU8825_SSP_AMP(quirk) \ - (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) -#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 -#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) -#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ - (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) - -/* BT audio offload: reserve 3 bits for future */ -#define SOF_BT_OFFLOAD_SSP_SHIFT 10 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) - -static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +static unsigned long sof_nau8825_quirk = SOF_SSP_PORT_CODEC(0); static struct snd_soc_jack_pin jack_pins[] = { { @@ -264,41 +245,19 @@ static int sof_audio_probe(struct platform_device *pdev) struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); - dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); - /* default number of DMIC DAI's */ - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> - SOF_NAU8825_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_nau8825_quirk); + if (!ctx) + return -ENOMEM; if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - /* port number of peripherals attached to ssp interface */ - ctx->ssp_bt = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ctx->ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> - SOF_NAU8825_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; - - if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_nau8825, ctx); if (ret) @@ -312,10 +271,10 @@ static int sof_audio_probe(struct platform_device *pdev) case CODEC_RT1015P: sof_rt1015p_codec_conf(&sof_audio_card_nau8825); break; - case CODEC_NONE: case CODEC_MAX98360A: case CODEC_NAU8318: case CODEC_RT1019P: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -339,34 +298,26 @@ static int sof_audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { - .name = "sof_nau8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - - }, - { .name = "adl_rt1019p_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(2) | - SOF_NAU8825_NUM_HDMIDEV(4)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "adl_nau8825_def", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "rpl_nau8825_def", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -392,4 +343,3 @@ MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_NUVOTON_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 640d17c6cd35..aadd341a202c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -27,37 +27,14 @@ #include "sof_board_helpers.h" #include "sof_maxim_common.h" #include "sof_realtek_common.h" -#include "sof_ssp_common.h" - -#define SOF_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_RT5682_MCLK_EN BIT(3) -#define SOF_RT5682_SSP_AMP_SHIFT 6 -#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6)) -#define SOF_RT5682_SSP_AMP(quirk) \ - (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) -#define SOF_RT5682_MCLK_BYTCHT_EN BIT(9) -#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10 -#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) -#define SOF_RT5682_NUM_HDMIDEV(quirk) \ - ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) - -/* BT audio offload: reserve 3 bits for future */ -#define SOF_BT_OFFLOAD_SSP_SHIFT 19 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(21, 19)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) - -/* HDMI capture*/ -#define SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT 27 -#define SOF_SSP_HDMI_CAPTURE_PRESENT_MASK (GENMASK(30, 27)) -#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \ - (((quirk) << SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT) & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) + +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_RT5682_MCLK_EN BIT(0) +#define SOF_RT5682_MCLK_BYTCHT_EN BIT(1) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0); + SOF_SSP_PORT_CODEC(0); static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) { @@ -72,7 +49,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max"), }, - .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)), + .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)), }, { .callback = sof_rt5682_quirk_cb, @@ -80,7 +57,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"), }, - .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)), + .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)), }, { .callback = sof_rt5682_quirk_cb, @@ -89,7 +66,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1)), + SOF_SSP_PORT_CODEC(1)), }, { .callback = sof_rt5682_quirk_cb, @@ -97,8 +74,8 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .callback = sof_rt5682_quirk_cb, @@ -107,7 +84,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0)), + SOF_SSP_PORT_CODEC(0)), }, { .callback = sof_rt5682_quirk_cb, @@ -116,9 +93,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -128,9 +105,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98373_ALC5682I_I2S"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -139,9 +116,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -150,9 +127,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S_AMP_SSP2"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -160,11 +137,10 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT ), }, {} @@ -630,19 +606,29 @@ static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; + bool is_legacy_cpu; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; dmi_check_system(sof_rt5682_quirk_table); - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + if (soc_intel_is_byt() || soc_intel_is_cht()) { + is_legacy_cpu = true; + + /* default quirk for legacy cpu */ + sof_rt5682_quirk = SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_BYTCHT_EN | + SOF_SSP_PORT_CODEC(2); + } + + dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_rt5682_quirk); + if (!ctx) + return -ENOMEM; if (ctx->codec_type == CODEC_RT5650) { sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", @@ -653,23 +639,12 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->amp_type = CODEC_RT5650; } - if (soc_intel_is_byt() || soc_intel_is_cht()) { + if (is_legacy_cpu) { ctx->rt5682.is_legacy_cpu = true; ctx->dmic_be_num = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ ctx->hdmi_num = 0; - /* default quirk for legacy cpu */ - sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_BYTCHT_EN | - SOF_RT5682_SSP_CODEC(2); } else { - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >> - SOF_RT5682_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; - if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; } @@ -694,23 +669,6 @@ static int sof_audio_probe(struct platform_device *pdev) } } - dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); - - /* port number/mask of peripherals attached to ssp interface */ - ctx->ssp_mask_hdmi_in = (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; - - ctx->ssp_bt = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ctx->ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> - SOF_RT5682_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; - - if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_rt5682, ctx); if (ret) @@ -733,11 +691,11 @@ static int sof_audio_probe(struct platform_device *pdev) case CODEC_RT1015P: sof_rt1015p_codec_conf(&sof_audio_card_rt5682); break; - case CODEC_NONE: case CODEC_MAX98357A: case CODEC_MAX98360A: case CODEC_RT1019P: case CODEC_RT5650: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -764,98 +722,93 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "cml_rt1015_rt5682", + .name = "cml_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "jsl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "tgl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "adl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1) | - SOF_RT5682_NUM_HDMIDEV(3) | + SOF_SSP_PORT_CODEC(1) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_HDMI_CAPTURE_SSP_MASK(0x5)), + SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, { .name = "rpl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "rpl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "rpl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1) | - SOF_RT5682_NUM_HDMIDEV(3) | + SOF_SSP_PORT_CODEC(1) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_HDMI_CAPTURE_SSP_MASK(0x5)), + SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, { .name = "mtl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "mtl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(3)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "mtl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -881,4 +834,3 @@ MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index ee2e813bf4c0..206c9b723805 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -20,34 +20,12 @@ #include "sof_board_helpers.h" #include "sof_realtek_common.h" #include "sof_cirrus_common.h" -#include "sof_ssp_common.h" - -/* SSP port ID for speaker amplifier */ -#define SOF_AMPLIFIER_SSP(quirk) ((quirk) & GENMASK(3, 0)) -#define SOF_AMPLIFIER_SSP_MASK (GENMASK(3, 0)) - -/* HDMI capture*/ -#define SOF_HDMI_CAPTURE_SSP_MASK_SHIFT 4 -#define SOF_HDMI_CAPTURE_SSP_MASK_MASK (GENMASK(9, 4)) -#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \ - (((quirk) << SOF_HDMI_CAPTURE_SSP_MASK_SHIFT) & SOF_HDMI_CAPTURE_SSP_MASK_MASK) - -/* HDMI playback */ -#define SOF_HDMI_PLAYBACK_PRESENT BIT(13) -#define SOF_NO_OF_HDMI_PLAYBACK_SHIFT 14 -#define SOF_NO_OF_HDMI_PLAYBACK_MASK (GENMASK(16, 14)) -#define SOF_NO_OF_HDMI_PLAYBACK(quirk) \ - (((quirk) << SOF_NO_OF_HDMI_PLAYBACK_SHIFT) & SOF_NO_OF_HDMI_PLAYBACK_MASK) - -/* BT audio offload */ -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(17) -#define SOF_BT_OFFLOAD_SSP_SHIFT 18 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(20, 18)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) + +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_HDMI_PLAYBACK_PRESENT BIT(0) /* Default: SSP2 */ -static unsigned long sof_ssp_amp_quirk = SOF_AMPLIFIER_SSP(2); +static unsigned long sof_ssp_amp_quirk = SOF_SSP_PORT_AMP(2); static const struct dmi_system_id chromebook_platforms[] = { { @@ -75,197 +53,107 @@ static struct snd_soc_card sof_ssp_amp_card = { #define HDMI_IN_BE_ID 0 #define SPK_BE_ID 2 #define DMIC01_BE_ID 3 -#define DMIC16K_BE_ID 4 #define INTEL_HDMI_BE_ID 5 - -static struct snd_soc_dai_link * -sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, - int ssp_amp, int dmic_be_num, int hdmi_num, - bool idisp_codec) +/* extra BE links to support no-hdmi-in boards */ +#define DMIC16K_BE_ID 4 +#define BT_OFFLOAD_BE_ID 8 + +#define SSP_AMP_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_HDMI_IN, \ + SOF_LINK_AMP, \ + SOF_LINK_DMIC01, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_BT_OFFLOAD, \ + SOF_LINK_NONE) + +#define SSP_AMP_LINK_IDS SOF_LINK_ORDER(HDMI_IN_BE_ID, \ + SPK_BE_ID, \ + DMIC01_BE_ID, \ + DMIC16K_BE_ID, \ + INTEL_HDMI_BE_ID, \ + BT_OFFLOAD_BE_ID, \ + 0) + +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) { - struct snd_soc_dai_link *links; - int i; - int id = 0; int ret; - bool fixed_be = false; - int be_id; - unsigned long ssp_mask_hdmi_in; - - links = devm_kcalloc(dev, sof_ssp_amp_card.num_links, - sizeof(struct snd_soc_dai_link), GFP_KERNEL); - if (!links) - return NULL; - - /* HDMI-In SSP */ - ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> - SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; - - if (ssp_mask_hdmi_in) { - int port = 0; - - /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ - fixed_be = true; - - be_id = HDMI_IN_BE_ID; - for_each_set_bit(port, &ssp_mask_hdmi_in, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, &links[id], - be_id, port); - if (ret) - return NULL; - - id++; - be_id++; - } - } - - /* codec SSP */ - if (amp_type != CODEC_NONE) { - be_id = fixed_be ? SPK_BE_ID : id; - ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], be_id, - amp_type, ssp_amp); - if (ret) - return NULL; - - /* codec-specific fields */ - switch (amp_type) { - case CODEC_CS35L41: - cs35l41_set_dai_link(&links[id]); - break; - case CODEC_RT1308: - sof_rt1308_dai_link(&links[id]); - break; - default: - dev_err(dev, "invalid amp type %d\n", amp_type); - return NULL; - } - id++; - } - - /* dmic */ - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - be_id = fixed_be ? DMIC01_BE_ID : id; - ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id, - SOF_DMIC_01); - if (ret) - return NULL; - - id++; - } - - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - be_id = fixed_be ? DMIC16K_BE_ID : id; - ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id, - SOF_DMIC_16K); - if (ret) - return NULL; - - id++; - } + ret = sof_intel_board_set_dai_link(dev, card, ctx); + if (ret) + return ret; - /* HDMI playback */ - for (i = 1; i <= hdmi_num; i++) { - be_id = fixed_be ? (INTEL_HDMI_BE_ID + i - 1) : id; - ret = sof_intel_board_set_intel_hdmi_link(dev, &links[id], be_id, - i, idisp_codec); - if (ret) - return NULL; + if (ctx->amp_type == CODEC_NONE) + return 0; - id++; + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; } - /* BT audio offload */ - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - int port = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); - if (ret) - return NULL; - - id++; + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_CS35L41: + cs35l41_set_dai_link(ctx->amp_link); + break; + case CODEC_RT1308: + sof_rt1308_dai_link(ctx->amp_link); + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; } - return links; + return 0; } static int sof_ssp_amp_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_ssp_amp_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + dev_dbg(&pdev->dev, "sof_ssp_amp_quirk = %lx\n", sof_ssp_amp_quirk); - if (dmi_check_system(chromebook_platforms) || mach->mach_params.dmic_num > 0) - ctx->dmic_be_num = 2; - else - ctx->dmic_be_num = 0; - - /* port number/mask of peripherals attached to ssp interface */ - ctx->ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> - SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; - - ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; - - /* set number of dai links */ - sof_ssp_amp_card.num_links = ctx->dmic_be_num; - - if (ctx->amp_type != CODEC_NONE) - sof_ssp_amp_card.num_links++; + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_ssp_amp_quirk); + if (!ctx) + return -ENOMEM; - if (ctx->ssp_mask_hdmi_in) - sof_ssp_amp_card.num_links += hweight32(ctx->ssp_mask_hdmi_in); + if (!dmi_check_system(chromebook_platforms) && + (mach->mach_params.dmic_num == 0)) + ctx->dmic_be_num = 0; if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) { - ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >> - SOF_NO_OF_HDMI_PLAYBACK_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; - if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - - sof_ssp_amp_card.num_links += ctx->hdmi_num; } else { ctx->hdmi_num = 0; } - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - ctx->bt_offload_present = true; - sof_ssp_amp_card.num_links++; - } + ctx->link_order_overwrite = SSP_AMP_LINK_ORDER; - dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_amp, ctx->dmic_be_num, - ctx->hdmi_num, - ctx->hdmi.idisp_codec); - if (!dai_links) - return -ENOMEM; + if (ctx->ssp_mask_hdmi_in) { + /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ + ctx->link_id_overwrite = SSP_AMP_LINK_IDS; + } - sof_ssp_amp_card.dai_link = dai_links; + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &sof_ssp_amp_card, ctx); + if (ret) + return ret; /* update codec_conf */ switch (ctx->amp_type) { case CODEC_CS35L41: cs35l41_set_codec_conf(&sof_ssp_amp_card); break; - case CODEC_NONE: case CODEC_RT1308: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -292,38 +180,35 @@ static const struct platform_device_id board_ids[] = { }, { .name = "tgl_rt1308_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(2) | - SOF_HDMI_CAPTURE_SSP_MASK(0x22)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(2) | + SOF_SSP_MASK_HDMI_CAPTURE(0x22)), /* SSP 1 and SSP 5 are used for HDMI IN */ }, { .name = "adl_cs35l41", - .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(1) | - SOF_NO_OF_HDMI_PLAYBACK(4) | + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | SOF_HDMI_PLAYBACK_PRESENT | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "rpl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "mtl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | - /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | - SOF_HDMI_PLAYBACK_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ + SOF_HDMI_PLAYBACK_PRESENT), }, { } }; @@ -346,4 +231,3 @@ MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_CIRRUS_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 0da79a3ba1f0..7ce8aade07d7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -563,7 +563,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "sof_nau8825", + .drv_name = "adl_nau8825_def", .sof_tplg_filename = "sof-adl-nau8825.tplg", }, { @@ -616,7 +616,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "adl_mx98360_da7219", + .drv_name = "adl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-da7219.tplg", diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 5eab17820532..d47a548959ea 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -49,21 +49,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "10EC5682", - .drv_name = "cml_rt1015_rt5682", + .drv_name = "cml_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt1015_spk_codecs, .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", }, { .id = "10EC5682", - .drv_name = "sof_rt5682", + .drv_name = "cml_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98357a_spk_codecs, .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", }, { .id = "10EC5682", - .drv_name = "sof_rt5682", + .drv_name = "cml_rt5682_def", .sof_tplg_filename = "sof-cml-rt5682.tplg", }, { diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index a6ac2525df17..d4b397c53bcc 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -52,14 +52,14 @@ static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { { .id = "DLGS7219", - .drv_name = "jsl_mx98373_da7219", + .drv_name = "jsl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98373_spk, .sof_tplg_filename = "sof-jsl-da7219.tplg", }, { .id = "DLGS7219", - .drv_name = "jsl_mx98360_da7219", + .drv_name = "jsl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 00a21af210fa..77c917897c8d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -455,6 +455,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .drv_name = "rpl_lt6911_hdmi_ssp", .sof_tplg_filename = "sof-rpl-nocodec-hdmi-ssp02.tplg" }, + { + .id = "DLGS7219", + .drv_name = "rpl_da7219_def", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rpl_max98360a_amp, + .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", + }, + { + .id = "10EC5650", + .drv_name = "rpl_rt5682_def", + .sof_tplg_filename = "sof-rpl-rt5650.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index 42e636c10c1e..363cc258a3d5 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -1,6 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o +snd-soc-mtk-common-objs += mtk-dai-adda-common.o + obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.c b/sound/soc/mediatek/common/mtk-dai-adda-common.c new file mode 100644 index 000000000000..4dc1412489d6 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dai-adda-common.c @@ -0,0 +1,70 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek ALSA SoC Audio DAI ADDA Common + * + * Copyright (c) 2021 MediaTek Inc. + * Copyright (c) 2024 Collabora Ltd. + * AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> + */ + +#include <linux/delay.h> +#include <linux/dev_printk.h> + +#include "mtk-base-afe.h" +#include "mtk-dai-adda-common.h" + +unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_DL_RATE_8K; + case 11025: + return MTK_AFE_ADDA_DL_RATE_11K; + case 12000: + return MTK_AFE_ADDA_DL_RATE_12K; + case 16000: + return MTK_AFE_ADDA_DL_RATE_16K; + case 22050: + return MTK_AFE_ADDA_DL_RATE_22K; + case 24000: + return MTK_AFE_ADDA_DL_RATE_24K; + case 32000: + return MTK_AFE_ADDA_DL_RATE_32K; + case 44100: + return MTK_AFE_ADDA_DL_RATE_44K; + case 48000: + return MTK_AFE_ADDA_DL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_DL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_DL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + return MTK_AFE_ADDA_DL_RATE_48K; + } +} +EXPORT_SYMBOL_GPL(mtk_adda_dl_rate_transform); + +unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_UL_RATE_8K; + case 16000: + return MTK_AFE_ADDA_UL_RATE_16K; + case 32000: + return MTK_AFE_ADDA_UL_RATE_32K; + case 48000: + return MTK_AFE_ADDA_UL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_UL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_UL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + return MTK_AFE_ADDA_UL_RATE_48K; + } +} +EXPORT_SYMBOL_GPL(mtk_adda_ul_rate_transform); diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.h b/sound/soc/mediatek/common/mtk-dai-adda-common.h new file mode 100644 index 000000000000..208b0dd89f57 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dai-adda-common.h @@ -0,0 +1,45 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2021 MediaTek Inc. + * Copyright (c) 2024 Collabora Ltd. + * AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com> + */ + +#ifndef _MTK_DAI_ADDA_COMMON_H_ +#define _MTK_DAI_ADDA_COMMON_H_ + +struct mtk_base_afe; + +enum adda_input_mode_rate { + MTK_AFE_ADDA_DL_RATE_8K = 0, + MTK_AFE_ADDA_DL_RATE_11K = 1, + MTK_AFE_ADDA_DL_RATE_12K = 2, + MTK_AFE_ADDA_DL_RATE_16K = 3, + MTK_AFE_ADDA_DL_RATE_22K = 4, + MTK_AFE_ADDA_DL_RATE_24K = 5, + MTK_AFE_ADDA_DL_RATE_32K = 6, + MTK_AFE_ADDA_DL_RATE_44K = 7, + MTK_AFE_ADDA_DL_RATE_48K = 8, + MTK_AFE_ADDA_DL_RATE_96K = 9, + MTK_AFE_ADDA_DL_RATE_192K = 10, +}; + +enum adda_voice_mode_rate { + MTK_AFE_ADDA_UL_RATE_8K = 0, + MTK_AFE_ADDA_UL_RATE_16K = 1, + MTK_AFE_ADDA_UL_RATE_32K = 2, + MTK_AFE_ADDA_UL_RATE_48K = 3, + MTK_AFE_ADDA_UL_RATE_96K = 4, + MTK_AFE_ADDA_UL_RATE_192K = 5, + MTK_AFE_ADDA_UL_RATE_48K_HD = 6, +}; + +enum adda_rxif_delay_data { + DELAY_DATA_MISO1 = 0, + DELAY_DATA_MISO0 = 1, + DELAY_DATA_MISO2 = 1, +}; + +unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate); +unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate); +#endif diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c index a58e1e3674de..000a086a8cf4 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.c +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c @@ -22,7 +22,11 @@ static int set_card_codec_info(struct snd_soc_card *card, codec_node = of_get_child_by_name(sub_node, "codec"); if (!codec_node) { - dev_dbg(dev, "%s no specified codec\n", dai_link->name); + dev_dbg(dev, "%s no specified codec: setting dummy.\n", dai_link->name); + + dai_link->codecs = &snd_soc_dummy_dlc; + dai_link->num_codecs = 1; + dai_link->dynamic = 1; return 0; } diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index 0ac6409c6d61..78f3ad758c12 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -10,86 +10,7 @@ #include "mt6797-afe-common.h" #include "mt6797-interconnection.h" #include "mt6797-reg.h" - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} +#include "../common/mtk-dai-adda-common.h" /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { @@ -246,7 +167,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0); /* set input sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28; + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28; /* set output mode */ switch (rate) { @@ -296,7 +217,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, 0x1 << 0, 0x0 << 0); - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 5b8a274419ed..be69bcea2a78 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -10,6 +10,7 @@ #include "mt8183-afe-common.h" #include "mt8183-interconnection.h" #include "mt8183-reg.h" +#include "../common/mtk-dai-adda-common.h" enum { AUDIO_SDM_LEVEL_MUTE = 0, @@ -18,91 +19,6 @@ enum { /* you need to change formula of hp impedance and dc trim too */ }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -369,7 +285,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0); /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28; + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28; /* set output mode */ switch (rate) { @@ -420,7 +336,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, 0x1 << 0, 0x0 << 0); - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index ad6d4b5cf697..dbd157d1a1ea 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -11,6 +11,7 @@ #include "mt8186-afe-common.h" #include "mt8186-afe-gpio.h" #include "mt8186-interconnection.h" +#include "../common/mtk-dai-adda-common.h" enum { UL_IIR_SW = 0, @@ -33,35 +34,6 @@ enum { AUDIO_SDM_3RD, }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - #define SDM_AUTO_RESET_THRESHOLD 0x190000 struct mtk_afe_adda_priv { @@ -83,64 +55,6 @@ static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, return afe_priv->dai_priv[dai_id]; } -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - } - - return MTK_AFE_ADDA_DL_RATE_48K; -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - } - - return MTK_AFE_ADDA_UL_RATE_48K; -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -658,7 +572,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, adda_priv->dl_rate = rate; /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << DL_2_INPUT_MODE_CTL_SFT; /* set output mode, UP_SAMPLING_RATE_X8 */ @@ -721,7 +635,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, } } else { unsigned int ul_src_con0 = 0; - unsigned int voice_mode = adda_ul_rate_transform(afe, rate); + unsigned int voice_mode = mtk_adda_ul_rate_transform(afe, rate); adda_priv->ul_rate = rate; ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c index 7dc029f2b428..8a17d1935c48 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c @@ -14,6 +14,7 @@ #include "mt8188-afe-clk.h" #include "mt8188-afe-common.h" #include "mt8188-reg.h" +#include "../common/mtk-dai-adda-common.h" #define ADDA_HIRES_THRES 48000 @@ -24,94 +25,10 @@ enum { SUPPLY_SEQ_ADDA_AFE_ON, }; -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, -}; - -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO0 = 1, -}; - struct mtk_dai_adda_priv { bool hires_required; }; -static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - static int mt8188_adda_mtkaif_init(struct mtk_base_afe *afe) { struct mt8188_afe_private *afe_priv = afe->platform_priv; @@ -440,7 +357,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe, /* set sampling rate */ mask |= DL_2_INPUT_MODE_CTL_MASK; val |= FIELD_PREP(DL_2_INPUT_MODE_CTL_MASK, - afe_adda_dl_rate_transform(afe, rate)); + mtk_adda_dl_rate_transform(afe, rate)); /* turn off saturation */ mask |= DL_2_CH1_SATURATION_EN_CTL; @@ -474,7 +391,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe, mask = UL_VOICE_MODE_CTL_MASK; val = FIELD_PREP(UL_VOICE_MODE_CTL_MASK, - afe_adda_ul_rate_transform(afe, rate)); + mtk_adda_ul_rate_transform(afe, rate)); regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0, mask, val); diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index bdd1e91824d9..aed22baef9fb 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2205,44 +2205,34 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* reset controller to reset audio regs before regmap cache */ rstc = devm_reset_control_get_exclusive(dev, "audiosys"); - if (IS_ERR(rstc)) { - ret = PTR_ERR(rstc); - dev_err(dev, "could not get audiosys reset:%d\n", ret); - return ret; - } + if (IS_ERR(rstc)) + return dev_err_probe(dev, PTR_ERR(rstc), "could not get audiosys reset\n"); ret = reset_control_reset(rstc); - if (ret) { - dev_err(dev, "failed to trigger audio reset:%d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "failed to trigger audio reset\n"); - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) - goto err_pm_disable; + ret = devm_pm_runtime_enable(&pdev->dev); + if (ret) + return ret; /* regmap init */ afe->regmap = syscon_node_to_regmap(dev->parent->of_node); - if (IS_ERR(afe->regmap)) { - dev_err(dev, "could not get regmap from parent\n"); - ret = PTR_ERR(afe->regmap); - goto err_pm_disable; - } + if (IS_ERR(afe->regmap)) + return dev_err_probe(dev, PTR_ERR(afe->regmap), + "could not get regmap from parent"); + ret = regmap_attach_dev(dev, afe->regmap, &mt8192_afe_regmap_config); - if (ret) { - dev_warn(dev, "regmap_attach_dev fail, ret %d\n", ret); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(dev, ret, "regmap_attach_dev fail\n"); /* enable clock for regcache get default value from hw */ afe_priv->pm_runtime_bypass_reg_ctl = true; pm_runtime_get_sync(&pdev->dev); ret = regmap_reinit_cache(afe->regmap, &mt8192_afe_regmap_config); - if (ret) { - dev_err(dev, "regmap_reinit_cache fail, ret %d\n", ret); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(dev, ret, "regmap_reinit_cache fail\n"); pm_runtime_put_sync(&pdev->dev); afe_priv->pm_runtime_bypass_reg_ctl = false; @@ -2254,10 +2244,8 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) afe->memif_size = MT8192_MEMIF_NUM; afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif), GFP_KERNEL); - if (!afe->memif) { - ret = -ENOMEM; - goto err_pm_disable; - } + if (!afe->memif) + return -ENOMEM; for (i = 0; i < afe->memif_size; i++) { afe->memif[i].data = &memif_data[i]; @@ -2271,47 +2259,35 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) afe->irqs_size = MT8192_IRQ_NUM; afe->irqs = devm_kcalloc(dev, afe->irqs_size, sizeof(*afe->irqs), GFP_KERNEL); - if (!afe->irqs) { - ret = -ENOMEM; - goto err_pm_disable; - } + if (!afe->irqs) + return -ENOMEM; for (i = 0; i < afe->irqs_size; i++) afe->irqs[i].irq_data = &irq_data[i]; /* request irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id < 0) { - ret = irq_id; - goto err_pm_disable; - } + if (irq_id < 0) + return irq_id; ret = devm_request_irq(dev, irq_id, mt8192_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); - if (ret) { - dev_err(dev, "could not request_irq for Afe_ISR_Handle\n"); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(dev, ret, "could not request_irq for Afe_ISR_Handle\n"); /* init sub_dais */ INIT_LIST_HEAD(&afe->sub_dais); for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { ret = dai_register_cbs[i](afe); - if (ret) { - dev_warn(afe->dev, "dai register i %d fail, ret %d\n", - i, ret); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(afe->dev, ret, "dai %d register fail", i); } /* init dai_driver and component_driver */ ret = mtk_afe_combine_sub_dai(afe); - if (ret) { - dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", - ret); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(afe->dev, ret, "mtk_afe_combine_sub_dai fail\n"); /* others */ afe->mtk_afe_hardware = &mt8192_afe_hardware; @@ -2327,26 +2303,17 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* register platform */ ret = devm_snd_soc_register_component(&pdev->dev, &mt8192_afe_component, NULL, 0); - if (ret) { - dev_warn(dev, "err_platform\n"); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(dev, ret, "Couldn't register AFE component\n"); ret = devm_snd_soc_register_component(&pdev->dev, &mt8192_afe_pcm_component, afe->dai_drivers, afe->num_dai_drivers); - if (ret) { - dev_warn(dev, "err_dai_component\n"); - goto err_pm_disable; - } + if (ret) + return dev_err_probe(dev, ret, "Couldn't register AFE-PCM component\n"); return 0; - -err_pm_disable: - pm_runtime_disable(&pdev->dev); - - return ret; } static void mt8192_afe_pcm_dev_remove(struct platform_device *pdev) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c index 36d33032a37a..99de85b87643 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c @@ -13,6 +13,7 @@ #include "mt8192-afe-common.h" #include "mt8192-afe-gpio.h" #include "mt8192-interconnection.h" +#include "../common/mtk-dai-adda-common.h" enum { UL_IIR_SW = 0, @@ -35,93 +36,8 @@ enum { AUDIO_SDM_3RD, }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - #define SDM_AUTO_RESET_THRESHOLD 0x190000 -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -1156,7 +1072,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, unsigned int dl_src2_con1 = 0; /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << DL_2_INPUT_MODE_CTL_SFT; /* set output mode, UP_SAMPLING_RATE_X8 */ @@ -1246,7 +1162,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, unsigned int voice_mode = 0; unsigned int ul_src_con0 = 0; /* default value */ - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c index 0dd35255066b..8da1587128cc 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c @@ -12,6 +12,7 @@ #include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" #include "mt8195-reg.h" +#include "../common/mtk-dai-adda-common.h" #define ADDA_DL_GAIN_LOOPBACK 0x1800 #define ADDA_HIRES_THRES 48000 @@ -26,35 +27,6 @@ enum { }; enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, -}; - -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO0 = 1, - DELAY_DATA_MISO2 = 1, -}; - -enum { MTK_AFE_ADDA, MTK_AFE_ADDA6, }; @@ -63,62 +35,6 @@ struct mtk_dai_adda_priv { bool hires_required; }; -static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - static int mt8195_adda_mtkaif_init(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; @@ -644,7 +560,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe, /* set sampling rate */ mask |= DL_2_INPUT_MODE_CTL_MASK; - val |= DL_2_INPUT_MODE_CTL(afe_adda_dl_rate_transform(afe, rate)); + val |= DL_2_INPUT_MODE_CTL(mtk_adda_dl_rate_transform(afe, rate)); /* turn off saturation */ mask |= DL_2_CH1_SATURATION_EN_CTL; @@ -681,7 +597,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe, unsigned int mask = 0; mask |= UL_VOICE_MODE_CTL_MASK; - val |= UL_VOICE_MODE_CTL(afe_adda_ul_rate_transform(afe, rate)); + val |= UL_VOICE_MODE_CTL(mtk_adda_ul_rate_transform(afe, rate)); switch (id) { case MT8195_AFE_IO_UL_SRC1: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2ec13d1634b6..3ab6626ad680 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2796,10 +2796,12 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); if (!component->name) { - dev_err(dev, "ASoC: Failed to allocate name\n"); - return -ENOMEM; + component->name = fmt_single_name(dev, &component->id); + if (!component->name) { + dev_err(dev, "ASoC: Failed to allocate name\n"); + return -ENOMEM; + } } component->dev = dev; 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. */ diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 092ca09f3631..83db1a83d8ba 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -441,6 +441,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; + if (config->name) + pcm->component.name = config->name; + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) goto err_free_dma; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index ba4890991f0d..fad9432a10f1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1083,8 +1083,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, break; } - /* add route, but keep going if some fail */ - snd_soc_dapm_add_routes(dapm, route, 1); + ret = snd_soc_dapm_add_routes(dapm, route, 1); + if (ret) { + if (!dapm->card->disable_route_checks) { + dev_err(tplg->dev, "ASoC: dapm_add_routes failed: %d\n", ret); + break; + } + dev_info(tplg->dev, + "ASoC: disable_route_checks set, ignoring dapm_add_routes errors\n"); + } } return ret; |