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

[alsa-devel] [PATCH] ASoC: Intel: kbl_da7219_max98357a: Map BTN_0 to KEY_PLAYPAUSE
by mac.chiang＠intel.com 02 May '19

02 May '19

From: Mac Chiang <mac.chiang(a)intel.com> On kbl_rt5663_max98927, commit 38a5882e4292 ("ASoC: Intel: kbl_rt5663_max98927: Map BTN_0 to KEY_PLAYPAUSE") This key pair mapping to play/pause when playing Youtube The Android 3.5mm Headset jack specification mentions that BTN_0 should be mapped to KEY_MEDIA, but this is less logical than KEY_PLAYPAUSE, which has much broader userspace support. For example, the Chrome OS userspace now supports KEY_PLAYPAUSE to toggle play/pause of videos and audio, but does not handle KEY_MEDIA. Furthermore, Android itself now supports KEY_PLAYPAUSE equivalently, as the new USB headset spec requires KEY_PLAYPAUSE for BTN_0. https://source.android.com/devices/accessories/headset/usb-headset-spec The same fix is required on Chrome kbl_da7219_max98357a. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Reviewed-by: Benson Leung <bleung(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 38f6ab7..07491a0 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -188,7 +188,7 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); -- 2.7.4

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[alsa-devel] [PATCH v2 0/2] ASoC: CX2072X codec support (revised)
by Takashi Iwai 02 May '19

02 May '19

Hi, this is a revised patchset for ASoC CX2072X codec support on Intel platforms. v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback === It's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2197 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 317 +++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 286 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2838 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

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[alsa-devel] [PATCH] ASoC: rt5645: Headphone Jack sense inverts on the LattePanda board
by Hui Wang 02 May '19

02 May '19

The LattePanda board has a sound card chtrt5645, when there is nothing plugged in the headphone jack, the system thinks the headphone is plugged in, while we plug a headphone in the jack, the system thinks the headphone is unplugged. If adding quirk=0x21 in the module parameter, the headphone jack can work well. So let us fix it via platform_data. https://bugs.launchpad.net/bugs/182459 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9a0751978090..a6064606424a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3631,6 +3631,11 @@ static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; +static const struct rt5645_platform_data lattepanda_board_platform_data = { + .jd_mode = 2, + .inv_jd1_1 = true +}; + static const struct dmi_system_id dmi_platform_data[] = { { .ident = "Chrome Buddy", @@ -3728,6 +3733,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&intel_braswell_platform_data, }, + { + .ident = "LattePanda board", + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_BOARD_VERSION, "Default string"), + }, + .driver_data = (void *)&lattepanda_board_platform_data, + }, { } }; -- 2.17.1

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[alsa-devel] [PATCH v2] ASoC: da7219: Use clk_round_rate to handle enabled bclk/wclk case
by Adam Thomson 02 May '19

02 May '19

For some platforms where DA7219 is the DAI clock master, BCLK/WCLK will be set and enabled prior to the codec's hw_params() function being called. It is possible the platform requires a different BCLK configuration than would be chosen by hw_params(), for example S16_LE format needed with a 64-bit frame to satisfy certain devices using the clocks. To handle those kinds of scenarios, the use of clk_round_rate() is now employed as part of hw_params(). If BCLK is already enabled then this function will just return the currently set rate, if it is valid for the desired frame size, so the subsequent call to clk_set_rate() will succeed and nothing changes with regards to clocking. In addition the specific BCLK & WCLK recalc_rate() implementations needed updating to always give back a real value, as those functions are called as part of the clk init code and a real value is needed for the clk_round_rate() call to work as expected. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Changes in v2: - Removed clk_round_rate() call for WCLK/SR as this is not sensible. - Added checking after clk_round_rate() call for BCLK to verify returne rate is suitable for the hw_params() desired frame size. sound/soc/codecs/da7219.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5f5fa34..206d01c 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1621,6 +1621,21 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, if (bclk) { bclk_rate = frame_size * sr; + /* + * Rounding the rate here avoids failure trying to set a + * new rate on an already enabled bclk. In that + * instance this will just set the same rate as is + * currently in use, and so should continue without + * problem, as long as the BCLK rate is suitable for the + * desired frame size. + */ + bclk_rate = clk_round_rate(bclk, bclk_rate); + if ((bclk_rate / sr) < frame_size) { + dev_err(component->dev, + "BCLK rate mismatch against frame size"); + return -EINVAL; + } + ret = clk_set_rate(bclk, bclk_rate); if (ret) { dev_err(component->dev, @@ -1927,9 +1942,6 @@ static unsigned long da7219_wclk_recalc_rate(struct clk_hw *hw, struct snd_soc_component *component = da7219->component; u8 fs = snd_soc_component_read32(component, DA7219_SR); - if (!da7219->master) - return 0; - switch (fs & DA7219_SR_MASK) { case DA7219_SR_8000: return 8000; @@ -2016,9 +2028,6 @@ static unsigned long da7219_bclk_recalc_rate(struct clk_hw *hw, u8 bclks_per_wclk = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); - if (!da7219->master) - return 0; - switch (bclks_per_wclk & DA7219_DAI_BCLKS_PER_WCLK_MASK) { case DA7219_DAI_BCLKS_PER_WCLK_32: return parent_rate * 32; -- 1.9.1

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[alsa-devel] [PATCH] spi: AD ASoC: declare missing of table
by Daniel Gomez 02 May '19

02 May '19

Add missing <of_device_id> table for SPI driver relying on SPI device match since compatible is in a DT binding or in a DTS. Before this patch: modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias alias: spi:adau1979 alias: spi:adau1978 alias: spi:adau1977 After this patch: modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias alias: of:N*T*Cadi,adau1979C* alias: of:N*T*Cadi,adau1979 alias: of:N*T*Cadi,adau1978C* alias: of:N*T*Cadi,adau1978 alias: of:N*T*Cadi,adau1977C* alias: of:N*T*Cadi,adau1977 alias: spi:adau1979 alias: spi:adau1978 alias: spi:adau1977 Reported-by: Javier Martinez Canillas <javier(a)dowhile0.org> Signed-off-by: Daniel Gomez <dagmcr(a)gmail.com> --- sound/soc/codecs/adau1977-spi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c index 84ffbde..2baf615 100644 --- a/sound/soc/codecs/adau1977-spi.c +++ b/sound/soc/codecs/adau1977-spi.c @@ -10,6 +10,8 @@ #include <linux/mod_devicetable.h> #include <linux/module.h> #include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <linux/spi/spi.h> #include <sound/soc.h> @@ -54,9 +56,18 @@ static const struct spi_device_id adau1977_spi_ids[] = { }; MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); +static const struct of_device_id adau1977_spi_of_match[] = { + { .compatible = "adi,adau1977" }, + { .compatible = "adi,adau1978" }, + { .compatible = "adi,adau1979" }, + { }, +}; +MODULE_DEVICE_TABLE(of, adau1977_spi_of_match); + static struct spi_driver adau1977_spi_driver = { .driver = { .name = "adau1977", + .of_match_table = of_match_ptr(adau1977_spi_of_match), }, .probe = adau1977_spi_probe, .id_table = adau1977_spi_ids, -- 2.7.4

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[alsa-devel] Applied "ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform"" to the asoc tree
by Mark Brown 02 May '19

02 May '19

The patch ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" 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 7b6531c5054e7804ccce25f389a2d4810357f5c9 Mon Sep 17 00:00:00 2001 From: Colin Ian King <colin.king(a)canonical.com> Date: Wed, 17 Apr 2019 17:26:28 +0100 Subject: [PATCH] ASoC: hdmi-codec: fix spelling mistake "plalform" -> "platform" There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdmi-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 35df73e42cbc..b9d9dde9fbaf 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -757,7 +757,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) dev_dbg(dev, "%s()\n", __func__); if (!hcd) { - dev_err(dev, "%s: No plalform data\n", __func__); + dev_err(dev, "%s: No platform data\n", __func__); return -EINVAL; } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: Fix build err while CONFIG_I2C set to module
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> During randconfig builds, I occasionally run into an invalid configuration WARNING: unmet direct dependencies detected for SND_SOC_TS3A227E Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=m] Selected by [y]: - SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8183 [=y] sound/soc/codecs/ts3a227e.o: In function `ts3a227e_i2c_probe': ts3a227e.c:(.text+0x684): undefined reference to `__devm_regmap_init_i2c' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_init': ts3a227e.c:(.init.text+0x18): undefined reference to `i2c_register_driver' sound/soc/codecs/ts3a227e.o: In function `ts3a227e_driver_exit': ts3a227e.c:(.exit.text+0x14): undefined reference to `i2c_del_driver' This patch add I2C dependency to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ebbddc75bbe8 ("ASoC: Mediatek: MT8183: Add machine driver with DA7219") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 874404b..f70b710 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -118,6 +118,7 @@ config SND_SOC_MT8183 config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A codec" + depends on I2C depends on SND_SOC_MT8183 select SND_SOC_MT6358 select SND_SOC_MAX98357A -- 2.7.4

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[alsa-devel] [v3] ASoC: AMD: Configure wclk and bclk of master codec
by Agrawal, Akshu 02 May '19

02 May '19

With CCF support in da7219, we can now set the correct rate of wclk and bclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Removed clk set rate and enable/disable from da7219 ops as da7219 codec takes care of them internally. Clock configuration kept for those codecs where da7219 acts as master of clock. As suggested by Adam Thomson. v3: As per spec, adau7002 bclk to be minimum of 64x of wclk. Hence, limiting bclk for all codecs. Also, because of this changes done in v2 of this patch have to be brought back. sound/soc/amd/acp-da7219-max98357a.c | 66 ++++++++++++++++++---------- 1 file changed, 42 insertions(+), 24 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 8a619a75b3a9..16b0ea3a3d72 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -46,7 +46,8 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -static struct clk *da7219_dai_clk; +static struct clk *da7219_dai_wclk; +static struct clk *da7219_dai_bclk; extern bool bt_uart_enable; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) @@ -72,7 +73,8 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); + da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | @@ -94,12 +96,15 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream) +static int da7219_clk_enable(struct snd_pcm_substream *substream, + int wclk_rate, int bclk_rate) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - ret = clk_prepare_enable(da7219_dai_clk); + clk_set_rate(da7219_dai_wclk, wclk_rate); + clk_set_rate(da7219_dai_bclk, bclk_rate); + ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); return ret; @@ -110,7 +115,7 @@ static int da7219_clk_enable(struct snd_pcm_substream *substream) static void da7219_clk_disable(void) { - clk_disable_unprepare(da7219_dai_clk); + clk_disable_unprepare(da7219_dai_bclk); } static const unsigned int channels[] = { @@ -151,7 +156,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -173,12 +178,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return da7219_clk_enable(substream); -} - -static void cz_da7219_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,12 +199,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); -} - -static void cz_max_shutdown(struct snd_pcm_substream *substream) -{ - da7219_clk_disable(); + return 0; } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -225,7 +220,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return da7219_clk_enable(substream); + return 0; } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -247,10 +242,28 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return da7219_clk_enable(substream); + return 0; } -static void cz_dmic_shutdown(struct snd_pcm_substream *substream) +static int cz_da7219_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int wclk, bclk; + + wclk = params_rate(params); + bclk = wclk * params_channels(params) * + snd_pcm_format_width(params_format(params)); + /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate + * that is minimum of 64x the LRCLK sample rate." + * DA7219 is the only clk source so for all codecs + * we have to limit bclk to 64X lrclk. + */ + if (bclk < (wclk * 64)) + bclk = wclk * 64; + return da7219_clk_enable(substream, wclk, bclk); +} + +static void cz_da7219_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); } @@ -258,26 +271,31 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, - .shutdown = cz_max_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, - .shutdown = cz_dmic_shutdown, + .shutdown = cz_da7219_shutdown, + .hw_params = cz_da7219_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

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[alsa-devel] [PATCH] ASoC: pcm3168a: Enable TDM support for DSP_A/B modes
by Peter Ujfalusi 02 May '19

02 May '19

The 24-bit TDM mode also applies to DSP_A and DSP_B modes. Most dais on the SoC side can not interpret I2S/Left_j with other than 2 channels of audio. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 08d3fe192e65..e0d5839fe1a7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -457,13 +457,16 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, if (chan > 2) { switch (fmt) { case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: fmt = PCM3168A_FMT_I2S_TDM; break; case PCM3168A_FMT_LEFT_J: + case PCM3168A_FMT_DSP_B: fmt = PCM3168A_FMT_LEFT_J_TDM; break; default: - dev_err(component->dev, "TDM is supported under I2S/Left_J only\n"); + dev_err(component->dev, + "TDM is supported under DSP/I2S/Left_J only\n"); return -EINVAL; } } @@ -526,6 +529,8 @@ static int pcm3168a_startup(struct snd_pcm_substream *substream, break; case PCM3168A_FMT_LEFT_J: case PCM3168A_FMT_I2S: + case PCM3168A_FMT_DSP_A: + case PCM3168A_FMT_DSP_B: sample_min = 24; channel_max = channel_maxs[tx]; break; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH -next] ASoC: fsl_micfil: Remove set but not used variable 'osr'
by Yue Haibing 02 May '19

02 May '19

From: YueHaibing <yuehaibing(a)huawei.com> Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_micfil.c: In function 'get_clk_div': sound/soc/fsl/fsl_micfil.c:154:6: warning: variable 'osr' set but not used [-Wunused-but-set-variable] It is never used since introduction in commit 47a70e6fc9a8 ("ASoC: Add MICFIL SoC Digital Audio Interface driver.") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_micfil.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 40c07e7..f7f2d29 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -151,12 +151,9 @@ static inline int get_clk_div(struct fsl_micfil *micfil, { u32 ctrl2_reg; long mclk_rate; - int osr; int clk_div; regmap_read(micfil->regmap, REG_MICFIL_CTRL2, &ctrl2_reg); - osr = 16 - ((ctrl2_reg & MICFIL_CTRL2_CICOSR_MASK) - >> MICFIL_CTRL2_CICOSR_SHIFT); mclk_rate = clk_get_rate(micfil->mclk); -- 2.7.4

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