April 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 0/2] Fix apq8016 compat string
by Bryan O'Donoghue 28 Jun '22

28 Jun '22

V3: - Marks qcom,lpass-cpu-apq8016 as deprecated instead of removing - Bjorn V2: - Adds Reviewed-by: - Srini - Adds Fixes - Srini V1: Reusing the apq8016 on msm8939 I found running checkpatch that the compat string for the LPASS was throwing a warning. This is easily fixed by alinging the YAML, DTS and driver to the documented compat string - compatible = "qcom,lpass-cpu-apq8016"; + compatible = "qcom,apq8016-lpass-cpu"; Bryan O'Donoghue (2): ASoC: qcom: lpass: Fix apq8016 compat string to match yaml arm64: dts: qcom: Fix apq8016 compat string to match yaml arch/arm64/boot/dts/qcom/msm8916.dtsi | 2 +- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 5 +++++ 3 files changed, 7 insertions(+), 1 deletion(-) -- 2.35.1

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[PATCH 1/4] ASoC: fsl-asoc-card: add WM8904 support
by Alifer Moraes 21 Jun '22

21 Jun '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> The codec WM8904 can use internal FLL as PLL source. Whenever the PLL source is not an external MCLK, this source must be setup during hw_params callback otherwise the BCLK could be wrongly evaluated. The SND_SOC_BIAS_PREPARE event is raised after the hw_params callback, so there is no need to set again PLL and SYSCLK and actually there's no need at all the set_bias_level function. Also, when esai is used, a dedicated snd_soc_dai_set_tdm_slot call is required. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/fsl/fsl-asoc-card.c | 47 +++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ee945505281..817dbc1ec635 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "imx-audmux.h" #include "../codecs/sgtl5000.h" +#include "../codecs/wm8904.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" @@ -257,6 +258,38 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) } } + if (of_device_is_compatible(dev->of_node, "fsl,imx-audio-wm8904")) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0, 0, 2, + params_physical_width(params)); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->pll_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + return 0; } @@ -651,6 +684,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->card.set_bias_level = NULL; + priv->codec_priv.mclk_id = WM8904_CLK_FLL; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + } } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -900,6 +946,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8904", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, -- 2.25.1

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[PATCH] ALSA: pcm: Test for "silence" field in struct "pcm_format_data"
by Fabio M. De Francesco 14 Jun '22

14 Jun '22

Syzbot reports "KASAN: null-ptr-deref Write in snd_pcm_format_set_silence".[1] It is due to missing validation of the "silence" field of struct "pcm_format_data" in "pcm_formats" array. Add a test for valid "pat" and, if it is not so, return -EINVAL. [1] https://lore.kernel.org/lkml/000000000000d188ef05dc2c7279@google.com/ Reported-and-tested-by: syzbot+205eb15961852c2c5974(a)syzkaller.appspotmail.com Signed-off-by: Fabio M. De Francesco <fmdefrancesco(a)gmail.com> --- I wasn't able to figure out the commit for the "Fixes:" tag. If this patch is good, can someone please help with providing this missing information? sound/core/pcm_misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4866aed97aac..5588b6a1ee8b 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -433,7 +433,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; - if (! width) + if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { -- 2.34.1

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[PATCH v2 00/14] ASoC: Intel: avs: Topology and path management
by Cezary Rojewski 10 Jun '22

10 Jun '22

A continuation of avs-driver initial series [1]. This chapter covers path management and topology parsing part which was ealier path of the main series. The two patches that represented these two subjects in the initial series, have been split into many to allow for easier review and discussion. AVS topology is split into two major parts: dictionaries - found within ASoC topology manifest - and path templates. Dictionaries job is to reduce the total amount of memory occupied by topology elements. Rather than having every pipeline and module carry its own information, each refers to specific entry in specific dictionary by provided (from topology file) indexes. In consequence, most struct avs_tplg_xxx are made out of pointers. Path templates are similar to path descriptions found in skylake-driver and they describe how given path shall look like in runtime - number of modules and pipelines that shape it and how they are laid out. A single path template is tied either to FE or BE and thus at most to a single, user-visible endpoint when speaking of FE. Path is a software representation of its ADSP firmware equivalent. It's a logical container for pipelines which are themselves containers - this time for modules i.e. processing units. Depending on the number of audio formats supported, each path template may carry one or more descriptions of given path. During runtime, when audio format is known, description matching said format is selected and used when instantiating path on ADSP firmware side through IPCs. Changes v1 -> v2: - fixed 'set but not used' warning in patch 10/14 as mentioned by ikp builder Changes RFC -> v1: - Plenty of rewordings and spelling fixes as requested by Pierre - any sysfs-functionality-related left in RFC dropped, will be part of separate subject in the future - separated parsing of formatted strings into: "Support link_mask formatted string" patch, increasing series size to 14 patches [1]: https://lore.kernel.org/alsa-devel/20220207122108.3780926-1-cezary.rojewski… Cezary Rojewski (14): ASoC: Intel: avs: Declare vendor tokens ASoC: Intel: avs: Add topology parsing infrastructure ASoC: Intel: avs: Parse module-extension tuples ASoC: Intel: avs: Parse pplcfg and binding tuples ASoC: Intel: avs: Parse pipeline and module tuples ASoC: Intel: avs: Parse path and path templates tuples ASoC: Intel: avs: Add topology loading operations ASoC: Intel: avs: Support link_mask formatted string ASoC: Intel: avs: Declare path and its components ASoC: Intel: avs: Path creation and freeing ASoC: Intel: avs: Path state management ASoC: Intel: avs: Arm paths after creating them ASoC: Intel: avs: Prepare modules before bindings them ASoC: Intel: avs: Configure modules according to their type include/uapi/sound/intel/avs/tokens.h | 126 ++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/avs/Makefile | 3 +- sound/soc/intel/avs/avs.h | 23 + sound/soc/intel/avs/path.c | 1005 ++++++++++++++++ sound/soc/intel/avs/path.h | 72 ++ sound/soc/intel/avs/topology.c | 1600 +++++++++++++++++++++++++ sound/soc/intel/avs/topology.h | 194 +++ 8 files changed, 3024 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h base-commit: bc55cfd5718c7c23e5524582e9fa70b4d10f2433 -- 2.25.1

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[PATCH 0/6] ASoC: ak4613: add TDM256 test support
by Kuninori Morimoto 08 Jun '22

08 Jun '22

Renesas is the only user of ak4613 driver on upstream for now. It has STEREO/TDM512/TDM256/TDM128 mode, but STEREO only is used, because of Renesas board connection. I noticed that I can test first 2ch out of TDM256 mode 8ch Playback even in such a situation. [PATCH 1/6] - [PATCH 5/6] are prepare for TDM mode support. [PATCH 6/6] uses "ifdef style" to enable TDM because I can't test full TDM256 and/or other TDM mode. "ifdef style" has no effect to current supported STEREO mode. Kuninori Morimoto (6): 1) ASoC: ak4613: add missing mutex_lock() 2) ASoC: ak4613: tidyup ak4613_interface 3) ASoC: ak4613: return error if it was setup as clock provider 4) ASoC: ak4613: priv has ctrl1 instead of iface 5) ASoC: ak4613: rename constraint to constraint_rates 6) ASoC: ak4613: add TDM256 support sound/soc/codecs/ak4613.c | 367 +++++++++++++++++++++++++++++++------- 1 file changed, 300 insertions(+), 67 deletions(-) -- 2.25.1

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[PATCH] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/G2UL SoC
by Biju Das 07 Jun '22

07 Jun '22

Document RZ/G2U2L SSI bindings. RZ/G2UL SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. While at it add a '.' at the end of dmas description for the first cell. Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 7e8d252f7bca..0d9840375132 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -13,6 +13,7 @@ properties: compatible: items: - enum: + - renesas,r9a07g043-ssi # RZ/G2UL - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi @@ -50,7 +51,7 @@ properties: minItems: 1 maxItems: 2 description: - The first cell represents a phandle to dmac + The first cell represents a phandle to dmac. The second cell specifies the encoded MID/RID values of the SSI port connected to the DMA client and the slave channel configuration parameters. -- 2.25.1

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[PATCH 00/26] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ALSA: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.32.0

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[PATCH v7 0/1] firmware: mtk: add adsp ipc protocol for SOF
by Allen-KH Cheng 25 May '22

25 May '22

This patch provides mediatek adsp ipc support for SOF. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This patch was tested and confirmed to with SOF fw on MT8195 cherry board. Based on matthias.bgg/linux.git, v5.18-next/soc changes since v6: - rebase to matthias.bgg/linux.git, v5.18-next/soc - Prefer "GPL" over "GPL v2" for MODULE_LICENSE changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): firmware: mediatek: add adsp ipc protocol interface drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 161 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 6 files changed, 239 insertions(+) create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

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[PATCH] ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min
by Marek Vasut 19 May '22

19 May '22

While the $val/$val2 values passed in from userspace are always >= 0 integers, the limits of the control can be signed integers and the $min can be non-zero and less than zero. To correctly validate $val/$val2 against platform_max, add the $min offset to val first. Fixes: 817f7c9335ec0 ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index f24f7354f46fe..6389a512c4dc6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -317,7 +317,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, mask = BIT(sign_bit + 1) - 1; val = ucontrol->value.integer.value[0]; - if (mc->platform_max && val > mc->platform_max) + if (mc->platform_max && ((int)val + min) > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; @@ -330,7 +330,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { val2 = ucontrol->value.integer.value[1]; - if (mc->platform_max && val2 > mc->platform_max) + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; -- 2.34.1

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[PATCH v2] ASoC: hdmi-codec: Add option for ELD bypass
by Sugar Zhang 14 May '22

14 May '22

This patch allow users to enable "ELD Bypass" who don't care anything from EDID Link Data. Currently, this driver gets ELD(from EDID) to constraint channels and rates. Unfortunately, EDID is not always valid, maybe caused by the fragile HDMI port or cable, in this situation, the max features are limited to 48kHz stereo. So, add this option to allow user to select the manual way to output audio as expected. such as multi-channels LPCM(7.1), or HBR bitstream for these sink devices. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - Use MACRO SOC_SINGLE_BOOL_EXT to simplify code. Fix event_missing checked by mixer-test. Add suffix "Switch" for "ELD Bypass". sound/soc/codecs/hdmi-codec.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b07607a..be46fbd 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -275,6 +275,7 @@ struct hdmi_codec_priv { unsigned int chmap_idx; struct mutex lock; bool busy; + bool eld_bypass; struct snd_soc_jack *jack; unsigned int jack_status; u8 iec_status[AES_IEC958_STATUS_SIZE]; @@ -427,6 +428,31 @@ static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } +static int hdmi_codec_eld_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = hcp->eld_bypass; + + return 0; +} + +static int hdmi_codec_eld_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + if (hcp->eld_bypass == ucontrol->value.integer.value[0]) + return 0; + + hcp->eld_bypass = ucontrol->value.integer.value[0]; + + return 1; +} + static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -447,7 +473,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (tx && hcp->hcd.ops->get_eld) { + if (tx && !hcp->eld_bypass && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -770,6 +796,8 @@ static struct snd_kcontrol_new hdmi_codec_controls[] = { .info = hdmi_eld_ctl_info, .get = hdmi_eld_ctl_get, }, + SOC_SINGLE_BOOL_EXT("ELD Bypass Switch", 0, + hdmi_codec_eld_bypass_get, hdmi_codec_eld_bypass_put), }; static int hdmi_codec_pcm_new(struct snd_soc_pcm_runtime *rtd, @@ -854,7 +882,7 @@ static void plugged_cb(struct device *dev, bool plugged) struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); if (plugged) { - if (hcp->hcd.ops->get_eld) { + if (!hcp->eld_bypass && hcp->hcd.ops->get_eld) { hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); } -- 2.7.4

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