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

[alsa-devel] [PATCH] ASoC: SOF: Intel: work around snd_hdac_aligned_read link failure
by Arnd Bergmann 10 Sep '19

10 Sep '19

When CONFIG_SND_HDA_ALIGNED_MMIO is selected by another driver (i.e. Tegra) that selects CONFIG_SND_HDA_CORE as a loadable module, but SND_SOC_SOF_HDA_COMMON is built-in, we get a link failure from some functions that access the hda register: sound/soc/sof/intel/hda.o: In function `hda_ipc_irq_dump': hda.c:(.text+0x784): undefined reference to `snd_hdac_aligned_read' sound/soc/sof/intel/hda-stream.o: In function `hda_dsp_stream_threaded_handler': hda-stream.c:(.text+0x12e4): undefined reference to `snd_hdac_aligned_read' hda-stream.c:(.text+0x12f8): undefined reference to `snd_hdac_aligned_write' Add an explicit 'select' statement as a workaround. This is not a great solution, but it's the easiest way I could come up with. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..9180184026e1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_SOF_HDA_COMMON tristate select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE + select SND_HDA_CORE if SND_HDA_ALIGNED_MMIO help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.20.0

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[alsa-devel] omap-mcbsp: TX Buffer Overflow
by Tomas Novotny 10 Sep '19

10 Sep '19

Hi, we have AM3703 based board similar to BeagleBoard. I'm hitting this error after upgrade to latest LTS 4.19.71 (upgraded from 4.1): omap-mcbsp 49022000.mcbsp: TX Buffer Overflow! This appears during or after playing of short (~2s) ding-dong wav. That error exists for longer time, because handling of tx buffer overflow irq was introduced in 2016: 4e85e7776eba ("ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts"). I've cherry-picked it to 4.1 and I see the error there also. The sound seems clear and ok to me, but we are using low quality speaker. There are two workarounds to get rid of the message: 1) Change 'dma_op_mode' sysfs attribute from 'element' to 'threshold'. I found that just by coincidence when checking sysfs attributes. 2) Compile kernel with CONFIG_VIDEO_OMAP3=y. Found on Logic PD forum [1]. Does anybody have any idea what's going wrong? Or why these (somehow) unrelated workarounds help? Thanks, Tomas [1] https://support.logicpd.com/TDGForum/tabid/124/aft/2277/Default.aspx

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[alsa-devel] [PATCH] ASoC: fsl_sai: Fix noise when using EDMA
by Daniel Baluta 09 Sep '19

09 Sep '19

From: Mihai Serban <mihai.serban(a)nxp.com> EDMA requires the period size to be multiple of maxburst. Otherwise the remaining bytes are not transferred and thus noise is produced. We can handle this issue by adding a constraint on SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 728307acab90..fe126029f4e3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -612,6 +612,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE); + /* + * some DMA controllers need period size to be a multiple of + * tx/rx maxburst + */ + if (sai->soc_data->use_constraint_period_size) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + tx ? sai->dma_params_tx.maxburst : + sai->dma_params_rx.maxburst); + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); @@ -1011,30 +1021,35 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .fifo_depth = 16, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .fifo_depth = 128, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .fifo_depth = 64, .reg_offset = 0, + .use_constraint_period_size = true, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b89b0ca26053..3a3f6f8e5595 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -157,6 +157,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; + bool use_constraint_period_size; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.17.1

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[alsa-devel] about pop chime on pcm api
by Yang, Xiaotao (X.) 09 Sep '19

09 Sep '19

Hi: Why should I call api<snd_pcm_close()> between the interval if I want to play a wav file two times repeatly, Otherwise pop chime occurs? Eg: (fake code as follows): Following codes works good. Pcm.open();//open pcm device Pcm.play(demo.wav);//set parameter and call snd_pcm_writei() to play Pcm.close();//close pcm device Pcm.open(); Pcm.play(demo.wav); Pcm.close(); But following codes works with pop chime Pcm.open(); Pcm.play(demo.wav); Pcm.play(demo.wav); Pcm.close();

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[alsa-devel] Applied "ASoC: hdmi-codec: Add an op to set callback function for plug event" to the asoc tree
by Mark Brown 09 Sep '19

09 Sep '19

The patch ASoC: hdmi-codec: Add an op to set callback function for plug event has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6fa5963c37a2e3335eba0b7455e35a01318ebc15 Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Wed, 17 Jul 2019 16:33:23 +0800 Subject: [PATCH] ASoC: hdmi-codec: Add an op to set callback function for plug event Add an op in hdmi_codec_ops so codec driver can register callback function to handle plug event. Driver in DRM can use this callback function to report connector status. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.org Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/hdmi-codec.h | 17 +++++++++++++ sound/soc/codecs/hdmi-codec.c | 46 +++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 7fea496f1f34..83b17682e01c 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -47,6 +47,9 @@ struct hdmi_codec_params { int channels; }; +typedef void (*hdmi_codec_plugged_cb)(struct device *dev, + bool plugged); + struct hdmi_codec_pdata; struct hdmi_codec_ops { /* @@ -88,6 +91,14 @@ struct hdmi_codec_ops { */ int (*get_dai_id)(struct snd_soc_component *comment, struct device_node *endpoint); + + /* + * Hook callback function to handle connector plug event. + * Optional + */ + int (*hook_plugged_cb)(struct device *dev, void *data, + hdmi_codec_plugged_cb fn, + struct device *codec_dev); }; /* HDMI codec initalization data */ @@ -99,6 +110,12 @@ struct hdmi_codec_pdata { void *data; }; +struct snd_soc_component; +struct snd_soc_jack; + +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + #define HDMI_CODEC_DRV_NAME "hdmi-audio-codec" #endif /* __HDMI_CODEC_H__ */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 0bf1c8cad108..b5fd8f08726e 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -7,6 +7,7 @@ #include <linux/module.h> #include <linux/string.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -274,6 +275,8 @@ struct hdmi_codec_priv { struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; struct mutex lock; + struct snd_soc_jack *jack; + unsigned int jack_status; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { @@ -663,6 +666,49 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) return 0; } +static void hdmi_codec_jack_report(struct hdmi_codec_priv *hcp, + unsigned int jack_status) +{ + if (hcp->jack && jack_status != hcp->jack_status) { + snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT); + hcp->jack_status = jack_status; + } +} + +static void plugged_cb(struct device *dev, bool plugged) +{ + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); + + if (plugged) + hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT); + else + hdmi_codec_jack_report(hcp, 0); +} + +/** + * hdmi_codec_set_jack_detect - register HDMI plugged callback + * @component: the hdmi-codec instance + * @jack: ASoC jack to report (dis)connection events on + */ +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -EOPNOTSUPP; + + if (hcp->hcd.ops->hook_plugged_cb) { + hcp->jack = jack; + ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent, + hcp->hcd.data, + plugged_cb, + component->dev); + if (ret) + hcp->jack = NULL; + } + return ret; +} +EXPORT_SYMBOL_GPL(hdmi_codec_set_jack_detect); + static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) { struct hdmi_codec_daifmt *cf = dai->playback_dma_data; -- 2.20.1

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[alsa-devel] Alsa conformance test - a new test for ALSA drivers
by Yu-Hsuan Hsu 09 Sep '19

09 Sep '19

Hi all, I have created an new tool "Alsa conformance test"( https://chromium.googlesource.com/chromiumos/platform/audiotest/+/master/al…), which is used for verifying ALSA drivers. Here are some main functions it can measure: 1. Correctness of hw_params 2. Stability of rate 3. Stability of step 4. Runtime of each ALSA API 5. Underrun and Overrun I recommend to use the script "alsa_conformance_test.py" inside. Just give it a device and then it will test the below functions automatically. 1. To make sure whether all the parameters provided by an ALSA driver can be used. 2. To make sure whether the measured rate is the same as the expected rate. We have used this tool to find out many issues in ChromeOS. I hope it can help development in Linux too. If you have any problems or want more features, please create a bug in crbug.com and assign to me ( yuhsuan(a)chromium.org). Thanks! Best, Yu-Hsuan

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[alsa-devel] [PATCH] SoC: simple-card-utils: set 0Hz to sysclk when shutdown
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch set 0Hz to sysclk when shutdown the card. Some codecs set rate constraints that derives from sysclk. This mechanism works correctly if machine drivers give fixed frequency. But simple-audio and audio-graph card set variable clock rate if 'mclk-fs' property exists. In this case, rate constraints will go bad scenario. For example a codec accepts three limited rates (mclk / 256, mclk / 384, mclk / 512). Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - Codec set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 24kHz (1/512) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256) - Codec set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512) - Play 48kHz again, but it's NOT acceptable because constraints do not allow 48kHz So codecs treat 0Hz sysclk as signal of applying no constraints to avoid this problem. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/generic/simple-card-utils.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 556b1a789629..9b794775df53 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -213,10 +213,17 @@ EXPORT_SYMBOL_GPL(asoc_simple_startup); void asoc_simple_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = simple_priv_to_props(priv, rtd->num); + if (dai_props->mclk_fs) { + snd_soc_dai_set_sysclk(codec_dai, 0, 0, SND_SOC_CLOCK_IN); + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } + asoc_simple_clk_disable(dai_props->cpu_dai); asoc_simple_clk_disable(dai_props->codec_dai); -- 2.23.0.rc1

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[alsa-devel] [PATCH] ASoC: rockchip: ignore 0Hz sysclk
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch ignores sysclk setting if it is 0Hz. Some codecs treat 0Hz sysclk as signal of applying no constraints. This driver does not have such feature but current implementation outputs 'Failed to set mclk' error message if machine driver sets 0Hz sysclk to this driver. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/rockchip/rockchip_i2s.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 88ebaf6e1880..af2d5a6124c8 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -419,6 +419,9 @@ static int rockchip_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, struct rk_i2s_dev *i2s = to_info(cpu_dai); int ret; + if (freq == 0) + return 0; + ret = clk_set_rate(i2s->mclk, freq); if (ret) dev_err(i2s->dev, "Fail to set mclk %d\n", ret); -- 2.23.0.rc1

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[alsa-devel] [PATCH 1/2] ASoC: es8316: fix redundant codes of clock
by Katsuhiro Suzuki 09 Sep '19

09 Sep '19

This patch removes redundant null checks for optional MCLK clock. And fix DT binding document for changing clock property to optional from required. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 3 ++ sound/soc/codecs/es8316.c | 31 ++++++++----------- 2 files changed, 16 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt index aefcff9c48a2..1bf03c5f2af4 100644 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ b/Documentation/devicetree/bindings/sound/everest,es8316.txt @@ -6,6 +6,9 @@ Required properties: - compatible : should be "everest,es8316" - reg : the I2C address of the device for I2C + +Optional properties: + - clocks : a list of phandle, should contain entries for clock-names - clock-names : should include as follows: "mclk" : master clock (MCLK) of the device diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index d07d50f51b28..1424ecd6952c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -373,11 +373,9 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq == 0) return 0; - if (es8316->mclk) { - ret = clk_set_rate(es8316->mclk, freq); - if (ret) - return ret; - } + ret = clk_set_rate(es8316->mclk, freq); + if (ret) + return ret; /* Limit supported sample rates to ones that can be autodetected * by the codec running in slave mode. @@ -712,20 +710,18 @@ static int es8316_probe(struct snd_soc_component *component) es8316->component = component; - es8316->mclk = devm_clk_get(component->dev, "mclk"); - if (PTR_ERR(es8316->mclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + es8316->mclk = devm_clk_get_optional(component->dev, "mclk"); if (IS_ERR(es8316->mclk)) { - dev_err(component->dev, "clock is invalid, ignored\n"); - es8316->mclk = NULL; + dev_err(component->dev, "unable to get mclk\n"); + return PTR_ERR(es8316->mclk); } + if (!es8316->mclk) + dev_warn(component->dev, "assuming static mclk\n"); - if (es8316->mclk) { - ret = clk_prepare_enable(es8316->mclk); - if (ret) { - dev_err(component->dev, "unable to enable clock\n"); - return ret; - } + ret = clk_prepare_enable(es8316->mclk); + if (ret) { + dev_err(component->dev, "unable to enable mclk\n"); + return ret; } /* Reset codec and enable current state machine */ @@ -754,8 +750,7 @@ static void es8316_remove(struct snd_soc_component *component) { struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - if (es8316->mclk) - clk_disable_unprepare(es8316->mclk); + clk_disable_unprepare(es8316->mclk); } static const struct snd_soc_component_driver soc_component_dev_es8316 = { -- 2.23.0.rc1

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[alsa-devel] [RFC PATCH] ASoC: ams-delta: Take control over audio mute GPIO pins
by Janusz Krzysztofik 09 Sep '19

09 Sep '19

Since commit 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs"), on-board audio has appeared muted. It has been discovered that believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be set low for audible sound in handsfree and handset mode respectively. According to Amstrad E3 wiki, the purpose of both pins hasn't been clearly identified. Original Amstrad software used to produce a high pulse on them when the phone was taken off hook or recall was pressed. With the current findings, we can assume the pins provide a kind of audio mute function, separately for handset and handsfree operation modes. Commit 2afdb4c41d78 ("ARM: OMAP1: ams-delta: Fix audio permanently muted") attempted to fix the issue temporarily by hogging the GPIO pin "hookflash1" renamed to "audio_mute", however the fix occurred incomplete as it restored audible sound only for handsfree mode. Stop hogging that pin, rename the pins to "handsfree_mute" and "handset_mute" respectively and implement appropriate DAPM event callbacks for "Speaker" and "Earpiece" DAPM widgets. Fixes: 1137ceee76ba ("ARM: OMAP1: ams-delta: Don't request unused GPIOs") Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- arch/arm/mach-omap1/board-ams-delta.c | 10 +++++---- sound/soc/ti/ams-delta.c | 31 +++++++++++++++++++++++++-- 2 files changed, 35 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index e47a6fbcfd6e..a2aa7a12b374 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -246,8 +246,8 @@ static struct platform_device latch2_gpio_device = { #define LATCH2_PIN_SCARD_CMDVCC 11 #define LATCH2_PIN_MODEM_NRESET 12 #define LATCH2_PIN_MODEM_CODEC 13 -#define LATCH2_PIN_AUDIO_MUTE 14 -#define LATCH2_PIN_HOOKFLASH 15 +#define LATCH2_PIN_HANDSFREE_MUTE 14 +#define LATCH2_PIN_HANDSET_MUTE 15 static struct regulator_consumer_supply modem_nreset_consumers[] = { REGULATOR_SUPPLY("RESET#", "serial8250.1"), @@ -476,6 +476,10 @@ static struct gpiod_lookup_table ams_delta_audio_gpio_table = { "hook_switch", 0), GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_MODEM_CODEC, "modem_codec", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSFREE_MUTE, + "handsfree_mute", 0), + GPIO_LOOKUP(LATCH2_LABEL, LATCH2_PIN_HANDSET_MUTE, + "handset_mute", 0), { }, }, }; @@ -590,8 +594,6 @@ static int gpiochip_match_by_label(struct gpio_chip *chip, void *data) static struct gpiod_hog ams_delta_gpio_hogs[] = { GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_KEYBRD_DATAOUT, "keybrd_dataout", GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), - GPIO_HOG(LATCH2_LABEL, LATCH2_PIN_AUDIO_MUTE, "audio_mute", - GPIO_ACTIVE_HIGH, GPIOD_OUT_LOW), {}, }; diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index dee8fc70a64f..8e2fb81ad05c 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -23,14 +23,31 @@ #include "omap-mcbsp.h" #include "../codecs/cx20442.h" +static struct gpio_desc *handset_mute; +static struct gpio_desc *handsfree_mute; + +static int ams_delta_event_handset(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + /* Board specific DAPM widgets */ static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset), /* Handsfree/Speakerphone */ SND_SOC_DAPM_MIC("Microphone", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree), }; /* How they are connected to codec pins */ @@ -542,6 +559,16 @@ static int ams_delta_probe(struct platform_device *pdev) card->dev = &pdev->dev; + handset_mute = devm_gpiod_get(card->dev, "handset_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handset_mute)) + return PTR_ERR(handset_mute); + + handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute", + GPIOD_OUT_HIGH); + if (IS_ERR(handsfree_mute)) + return PTR_ERR(handsfree_mute); + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); -- 2.21.0

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