June 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: intel_hdmi: Use strlcpy() instead of strncpy()
by Takashi Iwai 27 Jun '18

27 Jun '18

hdmi_lpe_audio_probe() copies the pcm name string via strncpy(), but as a gcc8 warning suggests, it misses a NUL terminator, and unlikely the expected result. Use the proper one, strlcpy() instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 4ed9d0c41843..edc9f5a34eff 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1854,7 +1854,7 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* setup private data which can be retrieved when required */ pcm->private_data = ctx; pcm->info_flags = 0; - strncpy(pcm->name, card->shortname, strlen(card->shortname)); + strlcpy(pcm->name, card->shortname, strlen(card->shortname)); /* setup the ops for playabck */ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops); -- 2.17.1

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[alsa-devel] [PATCH tinycompress] cplay: Stop playback stream at the end
by Daniel Baluta 27 Jun '18

27 Jun '18

Each compress_start should have a corresponding compress_stop at the end. Also, this gives user a change to cleanup resources that might have been allocated in compress_start. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- I'm implementing codec library dynamic loading at runtime and I'm exploring using: * SNDRV_COMPRESS_START -> to load the library * SNDRV_COMPRESS_STOP -> to unload the library This might not be the best idea becasue for example loading the library might introduce some delays, anyhow i still think this that compress_stop is correct thing to add at the end of playback. src/utils/cplay.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/utils/cplay.c b/src/utils/cplay.c index 87863a3..3c48d15 100644 --- a/src/utils/cplay.c +++ b/src/utils/cplay.c @@ -403,6 +403,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, printf("%s: exit success\n", __func__); /* issue drain if it supports */ compress_drain(compress); + compress_stop(compress); free(buffer); fclose(file); compress_close(compress); -- 2.7.4

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[alsa-devel] [[PATCH 0/9] ASoC: Intel: bytcr_rt5651: Cleanup (input mapping quirks)
by Hans de Goede 27 Jun '18

27 Jun '18

Hi Mark, This series is mainly a cleanup series. In the beginning of the rt5651 machine driver some wrong assumptions were made, such as the headset mic being attached to IN2 (it is on IN3 on the 7 machines I have access to and on all otherwise known machines). And also adding a quirk for a machine with the intmic on IN2, which later got fixed with a new quirk for machines with 2 internal mics on both IN1 and IN2 and moving the one machine with the IN2 quirk over to the new IN1_IN2 quirk, leaving the IN2 quirk as an orphan quirk for non existing hardware. Then I made a similar mistake adding the IN2_HS_IN3 quirk, while I should have used the IN1_HS_IN3 (which itself only exists because the original IN1 quirk has the headset input mapping wrong), TL;DR: it is a bit of a mess due to a number of wrong assumptions about how the inputs where actually routed in the past. This series cleans this all up in small commits / one bit a time and refers the original commit messages in its commit messages. Note patch 10/11 is not a cleanup patch, but is more or less the reason I took a second look at all the quirks. Regards, Hans

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[alsa-devel] [PATCH 0/3] ASoC: add simple-amplifier
by Jerome Brunet 27 Jun '18

27 Jun '18

As discuss here [0], this patchset rename the dio2125 driver to simple-amplifier in order to make it more discoverable. The last patch adds simple-audio-amplifier as new compatible [0]: https://lkml.kernel.org/r/20180625114445.GG9413@sirena.org.uk Jerome Brunet (3): ASoC: simple-amplifier: remame dio2125 documentation ASoC: simple-amplifier: rename dio2125 to simple-amplifer ASoC: simple-amplifer: add simple-amplifier compatible .../devicetree/bindings/sound/dioo,dio2125.txt | 12 ------- .../devicetree/bindings/sound/simple-amplifier.txt | 12 +++++++ sound/soc/codecs/Kconfig | 10 +++--- sound/soc/codecs/Makefile | 4 +-- sound/soc/codecs/{dio2125.c => simple-amplifier.c} | 42 +++++++++++----------- 5 files changed, 41 insertions(+), 39 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/dioo,dio2125.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt rename sound/soc/codecs/{dio2125.c => simple-amplifier.c} (68%) -- 2.14.4

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[alsa-devel] Applied "ASoC: simple-amplifier: rename dio2125 to simple-amplifer" to the asoc tree
by Mark Brown 27 Jun '18

27 Jun '18

The patch ASoC: simple-amplifier: rename dio2125 to simple-amplifer 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 8d881bb6216d28896112bdc0b42f1f21eb6cd4ee Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Tue, 26 Jun 2018 14:11:27 +0200 Subject: [PATCH] ASoC: simple-amplifier: rename dio2125 to simple-amplifer MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The dio2125 is simple enough that we can make it a generic component. Just rename and sed the dio2125 amplifier driver to simple_amplifier. Suggested-by: Nicolò Veronese <nicveronese(a)gmail.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 10 ++--- sound/soc/codecs/Makefile | 4 +- .../codecs/{dio2125.c => simple-amplifier.c} | 41 ++++++++++--------- 3 files changed, 28 insertions(+), 27 deletions(-) rename sound/soc/codecs/{dio2125.c => simple-amplifier.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f6b8d4bf8796..53c2d726bedf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -74,7 +74,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DIO2125 select SND_SOC_DMIC if GPIOLIB select SND_SOC_ES8316 if I2C select SND_SOC_ES8328_SPI if SPI_MASTER @@ -144,6 +143,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5682 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIMPLE_AMPLIFIER select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SPDIF select SND_SOC_SSM2305 @@ -573,10 +573,6 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DIO2125 - tristate "Dioo DIO2125 Amplifier" - select GPIOLIB - config SND_SOC_DMIC tristate @@ -897,6 +893,10 @@ config SND_SOC_SIGMADSP_REGMAP tristate select SND_SOC_SIGMADSP +config SND_SOC_SIMPLE_AMPLIFIER + tristate "Simple Audio Amplifier" + select GPIOLIB + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e43d99a039d3..f26ded89a1e5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -250,9 +250,9 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-zx-aud96p22-objs := zx_aud96p22.o # Amp -snd-soc-dio2125-objs := dio2125.o snd-soc-max9877-objs := max9877.o snd-soc-max98504-objs := max98504.o +snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o @@ -509,7 +509,7 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o obj-$(CONFIG_SND_SOC_ZX_AUD96P22) += snd-soc-zx-aud96p22.o # Amp -obj-$(CONFIG_SND_SOC_DIO2125) += snd-soc-dio2125.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o +obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/dio2125.c b/sound/soc/codecs/simple-amplifier.c similarity index 69% rename from sound/soc/codecs/dio2125.c rename to sound/soc/codecs/simple-amplifier.c index 09451cd44f9b..6c27d4afaf3a 100644 --- a/sound/soc/codecs/dio2125.c +++ b/sound/soc/codecs/simple-amplifier.c @@ -21,9 +21,9 @@ #include <linux/module.h> #include <sound/soc.h> -#define DRV_NAME "dio2125" +#define DRV_NAME "simple-amplifier" -struct dio2125 { +struct simple_amp { struct gpio_desc *gpiod_enable; }; @@ -31,7 +31,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) { struct snd_soc_component *c = snd_soc_dapm_to_component(w->dapm); - struct dio2125 *priv = snd_soc_component_get_drvdata(c); + struct simple_amp *priv = snd_soc_component_get_drvdata(c); int val; switch (event) { @@ -51,7 +51,7 @@ static int drv_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { +static const struct snd_soc_dapm_widget simple_amp_dapm_widgets[] = { SND_SOC_DAPM_INPUT("INL"), SND_SOC_DAPM_INPUT("INR"), SND_SOC_DAPM_OUT_DRV_E("DRV", SND_SOC_NOPM, 0, 0, NULL, 0, drv_event, @@ -60,24 +60,24 @@ static const struct snd_soc_dapm_widget dio2125_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTR"), }; -static const struct snd_soc_dapm_route dio2125_dapm_routes[] = { +static const struct snd_soc_dapm_route simple_amp_dapm_routes[] = { { "DRV", NULL, "INL" }, { "DRV", NULL, "INR" }, { "OUTL", NULL, "DRV" }, { "OUTR", NULL, "DRV" }, }; -static const struct snd_soc_component_driver dio2125_component_driver = { - .dapm_widgets = dio2125_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(dio2125_dapm_widgets), - .dapm_routes = dio2125_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(dio2125_dapm_routes), +static const struct snd_soc_component_driver simple_amp_component_driver = { + .dapm_widgets = simple_amp_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(simple_amp_dapm_widgets), + .dapm_routes = simple_amp_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(simple_amp_dapm_routes), }; -static int dio2125_probe(struct platform_device *pdev) +static int simple_amp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; - struct dio2125 *priv; + struct simple_amp *priv; int err; priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); @@ -93,28 +93,29 @@ static int dio2125_probe(struct platform_device *pdev) return err; } - return devm_snd_soc_register_component(dev, &dio2125_component_driver, + return devm_snd_soc_register_component(dev, + &simple_amp_component_driver, NULL, 0); } #ifdef CONFIG_OF -static const struct of_device_id dio2125_ids[] = { +static const struct of_device_id simple_amp_ids[] = { { .compatible = "dioo,dio2125", }, { } }; -MODULE_DEVICE_TABLE(of, dio2125_ids); +MODULE_DEVICE_TABLE(of, simple_amp_ids); #endif -static struct platform_driver dio2125_driver = { +static struct platform_driver simple_amp_driver = { .driver = { .name = DRV_NAME, - .of_match_table = of_match_ptr(dio2125_ids), + .of_match_table = of_match_ptr(simple_amp_ids), }, - .probe = dio2125_probe, + .probe = simple_amp_probe, }; -module_platform_driver(dio2125_driver); +module_platform_driver(simple_amp_driver); -MODULE_DESCRIPTION("ASoC DIO2125 output driver"); +MODULE_DESCRIPTION("ASoC Simple Audio Amplifier driver"); MODULE_AUTHOR("Jerome Brunet <jbrunet(a)baylibre.com>"); MODULE_LICENSE("GPL"); -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: add tas5707 support
by Jerome Brunet 27 Jun '18

27 Jun '18

This patchset extends the tas571x driver to support the tas5707 audio power amplifier. Jerome Brunet (2): ASoC: tas571x: add tas5707 compatible ASoC: tas517x: add tas5707 support .../devicetree/bindings/sound/tas571x.txt | 1 + sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/tas571x.c | 110 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 16 +++ 4 files changed, 131 insertions(+), 1 deletion(-) -- 2.14.4

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[alsa-devel] [PATCH v2 0/2] ASoC: add tas5707 support
by Jerome Brunet 27 Jun '18

27 Jun '18

This patchset extends the tas571x driver to support the tas5707 audio power amplifier. Jerome Brunet (2): ASoC: tas571x: add tas5707 compatible ASoC: tas517x: add tas5707 support Changes since v1 [0]: * Fixed double const as reported by kbuild robot [1] [0]: https://lkml.kernel.org/r/20180626174056.6400-1-jbrunet@baylibre.com [1]: https://lore.kernel.org/lkml/201806271000.j9AWAfLp%25fengguang.wu@intel.com/ .../devicetree/bindings/sound/tas571x.txt | 1 + sound/soc/codecs/Kconfig | 5 +- sound/soc/codecs/tas571x.c | 110 +++++++++++++++++++++ sound/soc/codecs/tas571x.h | 16 +++ 4 files changed, 131 insertions(+), 1 deletion(-) -- 2.14.4

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[alsa-devel] [PATCH] ASoC: Intel: Power down links before turning off display audio power
by Sriram Periyasamy 27 Jun '18

27 Jun '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++------- sound/soc/intel/skylake/skl.c | 11 ++++++----- 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 84f7a7a36e4b..b78fd84bf591 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2127,12 +2127,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); @@ -2141,7 +2135,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(ebus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 670ff9aaca55..c62e474717ff 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -762,6 +762,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -771,11 +777,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); -- 2.7.4

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[alsa-devel] ASoC: new ac97 bus and codec clock
by Robert Jarzmik 27 Jun '18

27 Jun '18

Hi Mark, The new ac97 is working well in a platform data build. But in the device tree case, there is an issue, triggered by our review in [1]. The main issue is the lack of one clock. The device hierarchy is : - platform device sound@40500000 (the AC97 controller) \-> ac97 device sound@40500000:0 (the discovered ac97 IP for a wm9713) Now the issue is here : - sound/ac97/bus.c:419, function ac97_get_enable_clk() - the clk_get() fails The sound@40500000:0 device was automatically created by the AC97 bus code, but I don't see any way to provide this device the "ac97_clk" ... How do you think this is supposed to work, and how can my devicetree description provide this clock ? -- Robert [1] Discussion reference https://groups.google.com/forum/#!msg/linux.kernel/6_zIXK_maA4/RMYsmLlRMgAJ Look for "You probably mean the BITCLK clock." [2] Device tree extract ac97: sound@40500000 { compatible = "marvell,pxa270-ac97"; reg = < 0x40500000 0x1000 >; interrupts = <14>; reset-gpios = <&gpio 95 GPIO_ACTIVE_HIGH>; #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = < &pinctrl_ac97_default >; clocks = <&clks CLK_AC97>, <&clks CLK_AC97CONF>; clock-names = "AC97CLK", "AC97CONFCLK"; };

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[alsa-devel] [PATCH] ALSA: hda - Kill snd_hda_codec_update_cache()
by Takashi Iwai 27 Jun '18

27 Jun '18

snd_hda_codec_update_cache() used to serve for a slightly different purpose from snd_hdac_write_cache(), but now both of them became identical. Let's unify and replace with the latter one consistently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/hda_codec.h | 3 --- sound/pci/hda/hda_generic.c | 8 ++++---- sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- 5 files changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d91c87e41756..f42af88bd2de 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3842,7 +3842,7 @@ EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); * This function is a helper to set a pin ctl value more safely. * It corrects the pin ctl value via snd_hda_correct_pin_ctl(), stores the * value in pin target array via snd_hda_codec_set_pin_target(), then - * actually writes the value via either snd_hda_codec_update_cache() or + * actually writes the value via either snd_hda_codec_write_cache() or * snd_hda_codec_write() depending on @cached flag. */ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, @@ -3851,7 +3851,7 @@ int _snd_hda_set_pin_ctl(struct hda_codec *codec, hda_nid_t pin, val = snd_hda_correct_pin_ctl(codec, pin, val); snd_hda_codec_set_pin_target(codec, pin, val); if (cached) - return snd_hda_codec_update_cache(codec, pin, 0, + return snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); else return snd_hda_codec_write(codec, pin, 0, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 681c360f29f9..8a095c16ee65 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -381,9 +381,6 @@ snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, return snd_hdac_regmap_write(&codec->core, nid, verb, parm); } -#define snd_hda_codec_update_cache(codec, nid, flags, verb, parm) \ - snd_hda_codec_write_cache(codec, nid, flags, verb, parm) - /* the struct for codec->pin_configs */ struct hda_pincfg { hda_nid_t nid; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 942f96e184b6..579984ecdec3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -209,7 +209,7 @@ static void parse_user_hints(struct hda_codec *codec) */ #define update_pin_ctl(codec, pin, val) \ - snd_hda_codec_update_cache(codec, pin, 0, \ + snd_hda_codec_write_cache(codec, pin, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) /* restore the pinctl based on the cached value */ @@ -898,7 +898,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, hda_nid_t nid = path->path[i]; if (enable && path->multi[i]) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, path->idx[i]); if (has_amp_in(codec, path, i)) @@ -930,7 +930,7 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) return; if (codec->inv_eapd) enable = !enable; - snd_hda_codec_update_cache(codec, pin, 0, + snd_hda_codec_write_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); } @@ -5973,7 +5973,7 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec) hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) - snd_hda_codec_update_cache(codec, nid, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, 0); } } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 757857313426..fd476fb40e1b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -148,7 +148,7 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) return; if (codec->inv_eapd) enabled = !enabled; - snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, + snd_hda_codec_write_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); } @@ -991,7 +991,7 @@ static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) if (spec->eapd_nid) ad_vmaster_eapd_hook(private_data, enabled); - snd_hda_codec_update_cache(codec, 0x01, 0, + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, enabled ? 0x00 : 0x02); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75ba66eb4ccd..8b48352689b4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -341,7 +341,7 @@ static void cxt_fixup_headset_mic(struct hda_codec *codec, * control. */ #define update_mic_pin(codec, nid, val) \ - snd_hda_codec_update_cache(codec, nid, 0, \ + snd_hda_codec_write_cache(codec, nid, 0, \ AC_VERB_SET_PIN_WIDGET_CONTROL, val) static const struct hda_input_mux olpc_xo_dc_bias = { -- 2.17.1

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