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

[alsa-devel] [RESEND PATCH v4 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 27 Aug '19

27 Aug '19

Resending this series due to a typo in yaml filename! Thanks for reviewing v3 patchset, here is v4 with addressing the comments in v3 This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v3: - updated slave bindings according to Rob's Suggestion. - moved bindings to yaml Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 44 + .../soundwire/soundwire-controller.yaml | 75 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1320 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] Applied "ASoC: sunxi: Revert initial A83t support" to the asoc tree
by Mark Brown 27 Aug '19

27 Aug '19

The patch ASoC: sunxi: Revert initial A83t support 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 455b1d42e82c6027b9763f0055b54e45ff6cd7fd Mon Sep 17 00:00:00 2001 From: Maxime Ripard <maxime.ripard(a)bootlin.com> Date: Tue, 27 Aug 2019 11:32:05 +0200 Subject: [PATCH] ASoC: sunxi: Revert initial A83t support This reverts commit 3e9acd7ac6933c (ASoC: sun4i-i2s: Remove duplicated quirks structure"). It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> Link: https://lore.kernel.org/r/20190827093206.17919-1-mripard@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: imx-audmix: register the card on a proper dev
by Shengjiu Wang 27 Aug '19

27 Aug '19

This platform device is registered from "fsl_audmix", which is its parent device. If use pdev->dev.parent for the priv->card.dev, the value set by dev_set_drvdata in parent device will be covered by the value in child device. Fixes: b86ef5367761 ("ASoC: fsl: Add Audio Mixer machine driver") Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-audmix.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 9e1cb18859ce..71590ca6394b 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -325,14 +325,14 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->card.num_configs = priv->num_dai_conf; priv->card.dapm_routes = priv->dapm_routes; priv->card.num_dapm_routes = priv->num_dapm_routes; - priv->card.dev = pdev->dev.parent; + priv->card.dev = &pdev->dev; priv->card.owner = THIS_MODULE; priv->card.name = "imx-audmix"; platform_set_drvdata(pdev, &priv->card); snd_soc_card_set_drvdata(&priv->card, priv); - ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed\n"); return ret; -- 2.21.0

3 2

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Make some functions static
by YueHaibing 27 Aug '19

27 Aug '19

Fix sparse warnings: sound/soc/sof/imx/imx8.c:104:6: warning: symbol 'imx8_dsp_handle_reply' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:115:6: warning: symbol 'imx8_dsp_handle_request' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:336:5: warning: symbol 'imx8_get_bar_index' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:341:6: warning: symbol 'imx8_ipc_msg_data' was not declared. Should it be static? sound/soc/sof/imx/imx8.c:348:5: warning: symbol 'imx8_ipc_pcm_params' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/imx/imx8.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f58..6404724 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -101,7 +101,7 @@ static int imx8_get_window_offset(struct snd_sof_dev *sdev, u32 id) return MBOX_OFFSET; } -void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) +static void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) { struct imx8_priv *priv = imx_dsp_get_data(ipc); unsigned long flags; @@ -112,7 +112,7 @@ void imx8_dsp_handle_reply(struct imx_dsp_ipc *ipc) spin_unlock_irqrestore(&priv->sdev->ipc_lock, flags); } -void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc) +static void imx8_dsp_handle_request(struct imx_dsp_ipc *ipc) { struct imx8_priv *priv = imx_dsp_get_data(ipc); @@ -333,21 +333,21 @@ static int imx8_remove(struct snd_sof_dev *sdev) } /* on i.MX8 there is 1 to 1 match between type and BAR idx */ -int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) +static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) { return type; } -void imx8_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static void imx8_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); } -int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - const struct sof_ipc_pcm_params_reply *reply) +static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + const struct sof_ipc_pcm_params_reply *reply) { return 0; } -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: SOF: imx8: Fix an is IS_ERR() vs NULL check
by Dan Carpenter 27 Aug '19

27 Aug '19

The device_link_add() function only returns NULL on error, it doesn't return error pointers. Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/imx/imx8.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f584207f..41ad3a310808 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -227,8 +227,8 @@ static int imx8_probe(struct snd_sof_dev *sdev) DL_FLAG_STATELESS | DL_FLAG_PM_RUNTIME | DL_FLAG_RPM_ACTIVE); - if (IS_ERR(priv->link[i])) { - ret = PTR_ERR(priv->link[i]); + if (!priv->link[i]) { + ret = -ENOMEM; dev_pm_domain_detach(priv->pd_dev[i], false); goto exit_unroll_pm; } -- 2.20.1

3 2

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Fix return value check in imx8_probe()
by Wei Yongjun 27 Aug '19

27 Aug '19

In case of error, the function devm_ioremap_wc() returns NULL pointer not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sof/imx/imx8.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index e502f584207f..263d4df35fe8 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -296,10 +296,10 @@ static int imx8_probe(struct snd_sof_dev *sdev) sdev->bar[SOF_FW_BLK_TYPE_SRAM] = devm_ioremap_wc(sdev->dev, res.start, res.end - res.start + 1); - if (IS_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM])) { + if (!sdev->bar[SOF_FW_BLK_TYPE_SRAM]) { dev_err(sdev->dev, "failed to ioremap mem 0x%x size 0x%x\n", base, size); - ret = PTR_ERR(sdev->bar[SOF_FW_BLK_TYPE_SRAM]); + ret = -ENOMEM; goto exit_pdev_unregister; } sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM;

3 2

[alsa-devel] [PATCH] ASoC: mediatek: mt8183: remove unused DAPM-related variables
by Tzung-Bi Shih 27 Aug '19

27 Aug '19

Remove unused variable mt8183_mt6358_ts3a227_max98357_dapm_widgets and mt8183_mt6358_ts3a227_max98357_dapm_routes. They are accidentially included when rebasing commits. Fixes: 6191cbde5ff0 ("ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index fe8a7461ace5..951706fa75fc 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -46,16 +46,6 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static const struct snd_soc_dapm_widget -mt8183_mt6358_ts3a227_max98357_dapm_widgets[] = { - SND_SOC_DAPM_OUTPUT("IT6505_8CH"), -}; - -static const struct snd_soc_dapm_route -mt8183_mt6358_ts3a227_max98357_dapm_routes[] = { - {"IT6505_8CH", NULL, "TDM"}, -}; - enum PINCTRL_PIN_STATE { PIN_STATE_DEFAULT = 0, PIN_TDM_OUT_ON, -- 2.23.0.187.g17f5b7556c-goog

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: es8316: fix headphone mixer volume table
by Katsuhiro Suzuki 27 Aug '19

27 Aug '19

This patch fix setting table of Headphone mixer volume. Current code uses 4 ... 7 values but these values are prohibited. Correct settings are the following: 0000 -12dB 0001 -10.5dB 0010 -9dB 0011 -7.5dB 0100 -6dB 1000 -4.5dB 1001 -3dB 1010 -1.5dB 1011 0dB Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8dfb5dbeebbf..f97d9c5210f0 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -53,7 +53,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -91,7 +94,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, -- 2.23.0.rc1

3 5

[alsa-devel] [PATCH 1/2] ASoC: rt1011: add mutex protection to set_fmt/set_tdm_slot
by shumingf＠realtek.com 27 Aug '19

27 Aug '19

From: Shuming Fan <shumingf(a)realtek.com> The calibration process at booting will reset registers and bypass cache to make sure the calibration is done. We add mutex protection to avoid unexpected settings while the registration process and calibration are interleaved. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 27 +++++++++++++++++++-------- 1 file changed, 19 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index fa34565a3938..f9235764ad32 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1627,14 +1627,18 @@ static int rt1011_hw_params(struct snd_pcm_substream *substream, static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int reg_val = 0, reg_bclk_inv = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: reg_val |= RT1011_I2S_TDM_MS_S; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -1644,7 +1648,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_bclk_inv |= RT1011_TDM_INV_BCLK; break; default: - return -EINVAL; + ret = -EINVAL; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -1660,7 +1664,7 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) reg_val |= RT1011_I2S_TDM_DF_PCM_B; break; default: - return -EINVAL; + ret = -EINVAL; } switch (dai->id) { @@ -1675,9 +1679,11 @@ static int rt1011_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; default: dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); - return -EINVAL; + ret = -EINVAL; } - return 0; + + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_set_component_sysclk(struct snd_soc_component *component, @@ -1796,8 +1802,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_component *component = dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); unsigned int val = 0, tdm_en = 0; + int ret = 0; + snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) tdm_en = RT1011_TDM_I2S_DOCK_EN_1; @@ -1817,7 +1827,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 2: break; default: - return -EINVAL; + ret = -EINVAL; } switch (slot_width) { @@ -1836,7 +1846,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, case 16: break; default: - return -EINVAL; + ret = -EINVAL; } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1853,7 +1863,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); - return 0; + snd_soc_dapm_mutex_unlock(dapm); + return ret; } static int rt1011_probe(struct snd_soc_component *component) -- 2.23.0

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: mediatek: mt8183: fix tdm out data align issue
by Jiaxin Yu 27 Aug '19

27 Aug '19

Mt8183 tdm out support S16_LE/S24_LE/S32_LE formats. When output S32_LE, we need set hd_align so that memif can output MSB 24bits. When output S24_LE, we need reset hd_align so that memif can output LSB 24bits. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index d16563408465..10ea4fdbeb1e 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -241,7 +241,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; int hd_audio = 0; - int hd_align = 1; + int hd_align = 0; /* set hd mode */ switch (substream->runtime->format) { @@ -254,7 +254,6 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, break; case SNDRV_PCM_FORMAT_S24_LE: hd_audio = 1; - hd_align = 0; break; default: dev_err(afe->dev, "%s() error: unsupported format %d\n", -- 2.18.0

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