December 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda: intel-dsp-config: add JasperLake support
by Brent Lu 13 Jan '22

13 Jan '22

Add rules to select SOF driver for Jasper Lake systems if digital microphone is present or the system is a Chromebook. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/hda/intel-dsp-config.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index b9ac9e9e45a4..be47a1ca12b6 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -299,6 +299,27 @@ static const struct config_entry config_table[] = { }, #endif +/* Jasper Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) + { + .flags = FLAG_SOF, + .device = 0x4dc8, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4dc8, + }, +#endif + /* Tigerlake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { -- 2.25.1

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[PATCH 0/3] firewire: assist unit driver to compute packet timestamp
by Takashi Sakamoto 12 Jan '22

12 Jan '22

Hi, In 1394 OHCI specification, each descriptor of IR/IT/AR/AT DMA context has timeStamp field. The value of timeStamp field express the time in which the controller accept packet. The resolution of value is isochronous cycle count (8,000 Hz) with second up to 7. I have a plan to use the value of timeStamp field for ALSA firewire stack so that userspace ALSA PCM/Rawmidi applications can get converted timestamp (ktime) for PCM frame/MIDI message. The timestamp can ideally express finer granularity than the time to invoke IRQ handler (and co). Current implementation of Linux FireWire subsystem delivers the value of timeStamp field to unit driver for IR/IT/AT DMA context, but not for AR DMA context. Additionally, the way to refer to Isochronous Cycle Timer Register in MMIO region of 1394 OHCI controller is transaction to local node. It includes overhead of transaction and it's preferable to add less-overhead way available in any type of IRQ context. This patchset adds two functions exposed in kernel space: * fw_card_read_cycle_time() * allow unit driver to access to CYCLE_TIME register in MMIO region without initiate transaction * fw_request_get_timestamp() * allow unit driver to get timestamp of request packet inner request handler I note that Hector Martin found kernel null pointer dereference during process to remove PCI card and has posted a patch: * https://lore.kernel.org/lkml/20211027113130.8802-1-marcan@marcan.st/ His patch is included in the series with my comment for relevant commit 20802224298c ("firewire: core: add forgotten dummy driver methods, remove unused ones"). The patch is required since unit driver can refer to dummy driver between removal callback of PCI subsystem and removal callback of FireWire subsystem. Hector Martin (1): firewire: Add dummy read_csr/write_csr functions Takashi Sakamoto (2): firewire: add kernel API to access CYCLE_TIME register firewire: add kernel API to access packet structure in request structure for AR context drivers/firewire/core-card.c | 39 +++++++++++++++++++++++++++++ drivers/firewire/core-cdev.c | 6 +++-- drivers/firewire/core-transaction.c | 18 +++++++++++++ include/linux/firewire.h | 3 +++ 4 files changed, 64 insertions(+), 2 deletions(-) -- 2.32.0

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[PATCH 0/7] ASoC/SoundWire: improve suspend flows and use set_stream() instead of set_tdm_slots() for HDAudio
by Bard Liao 10 Jan '22

10 Jan '22

This series contains three topics. 1. SoundWire: Intel: remove pdm support 2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire 3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines The topics are independent but the changes are dependent. So please allow me to send them in one series. Pierre-Louis Bossart (6): ASOC: SOF: Intel: use snd_soc_dai_get_widget() ASoC/soundwire: intel: simplify callbacks for params/hw_free ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio soundwire: intel: remove unnecessary init soundwire: intel: remove PDM support Ranjani Sridharan (1): soundwire: intel: improve suspend flows drivers/soundwire/cadence_master.c | 36 +--- drivers/soundwire/cadence_master.h | 14 +- drivers/soundwire/intel.c | 253 ++++++++++++++-------------- drivers/soundwire/qcom.c | 8 +- drivers/soundwire/stream.c | 4 +- include/linux/soundwire/sdw_intel.h | 4 +- include/sound/soc-dai.h | 32 ++-- sound/soc/codecs/hdac_hda.c | 22 +-- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca.c | 2 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/codecs/sdw-mockup.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 6 +- sound/soc/intel/skylake/skl-pcm.c | 7 +- sound/soc/qcom/sdm845.c | 4 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/sof/intel/hda-dai.c | 7 +- sound/soc/sof/intel/hda.c | 12 +- 26 files changed, 190 insertions(+), 247 deletions(-) -- 2.17.1

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[PATCH 1/5] ASoC: rt5640: Change jack_work to a delayed_work
by Hans de Goede 06 Jan '22

06 Jan '22

Change jack_work from a struct work_struct to a struct delayed_work, this is a preparation patch for adding support for boards where an external GPIO is used for jack-detect, rather then one of the JD pins of the codec. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5640.c | 12 ++++++------ sound/soc/codecs/rt5640.h | 2 +- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index d01fe73ab9c8..36c00ad17182 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2297,7 +2297,7 @@ EXPORT_SYMBOL_GPL(rt5640_detect_headset); static void rt5640_jack_work(struct work_struct *work) { struct rt5640_priv *rt5640 = - container_of(work, struct rt5640_priv, jack_work); + container_of(work, struct rt5640_priv, jack_work.work); struct snd_soc_component *component = rt5640->component; int status; @@ -2348,7 +2348,7 @@ static void rt5640_jack_work(struct work_struct *work) * disabled the OVCD IRQ, the IRQ pin will stay high and as * we react to edges, we miss the unplug event -> recheck. */ - queue_work(system_long_wq, &rt5640->jack_work); + queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } } @@ -2357,7 +2357,7 @@ static irqreturn_t rt5640_irq(int irq, void *data) struct rt5640_priv *rt5640 = data; if (rt5640->jack) - queue_work(system_long_wq, &rt5640->jack_work); + queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); return IRQ_HANDLED; } @@ -2366,7 +2366,7 @@ static void rt5640_cancel_work(void *data) { struct rt5640_priv *rt5640 = data; - cancel_work_sync(&rt5640->jack_work); + cancel_delayed_work_sync(&rt5640->jack_work); cancel_delayed_work_sync(&rt5640->bp_work); } @@ -2475,7 +2475,7 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, } /* sync initial jack state */ - queue_work(system_long_wq, &rt5640->jack_work); + queue_delayed_work(system_long_wq, &rt5640->jack_work, 0); } static int rt5640_set_jack(struct snd_soc_component *component, @@ -2856,7 +2856,7 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = true; rt5640->irq = i2c->irq; INIT_DELAYED_WORK(&rt5640->bp_work, rt5640_button_press_work); - INIT_WORK(&rt5640->jack_work, rt5640_jack_work); + INIT_DELAYED_WORK(&rt5640->jack_work, rt5640_jack_work); /* Make sure work is stopped on probe-error / remove */ ret = devm_add_action_or_reset(&i2c->dev, rt5640_cancel_work, rt5640); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 2c28f83e338a..7ab930def8dd 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2145,7 +2145,7 @@ struct rt5640_priv { int release_count; int poll_count; struct delayed_work bp_work; - struct work_struct jack_work; + struct delayed_work jack_work; struct snd_soc_jack *jack; unsigned int jd_src; bool jd_inverted; -- 2.33.1

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[PATCH] ALSA: hda: ALC287: Add Lenovo IdeaPad Slim 9i 14ITL5 speaker quirk
by Bart Kroon 05 Jan '22

05 Jan '22

Hello, The speaker fixup that is used for the Yoga 7 14ITL5 also applies to the IdeaPad Slim 9i 14ITL5. The attached patch applies the quirk to initialise the amplifier on the IdeaPad Slim 9i as well. This is validated to work on my laptop. Signed-off-by: Bart Kroon <bart(a)tarmack.eu> Regards, Bart

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Correct stopping capture and playback substreams?
by Pavel Hofman 05 Jan '22

05 Jan '22

Hi Takashi, I am working on stopping alsa streams of audio USB gadget when USB host stops capture/playback/USB cable unplugged. For capture I used code from AK4114 SPDIF receiver https://elixir.bootlin.com/linux/latest/source/sound/i2c/other/ak4114.c#L590: static void stop_substream(struct uac_rtd_params *prm) { unsigned long _flags; struct snd_pcm_substream *substream; substream = prm->ss; if (substream) { snd_pcm_stream_lock_irqsave(substream, _flags); if (snd_pcm_running(substream)) // TODO - correct handling for playback substream? snd_pcm_stop(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_stream_unlock_irqrestore(substream, _flags); } } For setup I found calling snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP) (https://elixir.bootlin.com/linux/latest/source/drivers/staging/vc04_service…) Or for both capture and playback using SNDRV_PCM_STATE_DISCONNECTED (https://elixir.bootlin.com/linux/latest/source/sound/core/pcm.c#L1103). Or perhaps using snd_pcm_dev_disconnect(dev) or snd_pcm_drop(substream)? Please what is the recommended way? Thanks a lot, Pavel.

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[PATCH] ASoC: qdsp6: fix a use after free bug in open()
by Dan Carpenter 05 Jan '22

05 Jan '22

This code frees "graph" and then dereferences to save the error code. Save the error code first and then use gotos to unwind the allocation. Fixes: 59716aa3f976 ("ASoC: qdsp6: Fix an IS_ERR() vs NULL bug") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/qcom/qdsp6/q6apm.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 3e007d609a9b..f424d7aa389a 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -615,7 +615,7 @@ struct q6apm_graph *q6apm_graph_open(struct device *dev, q6apm_cb cb, graph = kzalloc(sizeof(*graph), GFP_KERNEL); if (!graph) { ret = -ENOMEM; - goto err; + goto put_ar_graph; } graph->apm = apm; @@ -631,13 +631,15 @@ struct q6apm_graph *q6apm_graph_open(struct device *dev, q6apm_cb cb, graph->port = gpr_alloc_port(apm->gdev, dev, graph_callback, graph); if (IS_ERR(graph->port)) { - kfree(graph); ret = PTR_ERR(graph->port); - goto err; + goto free_graph; } return graph; -err: + +free_graph: + kfree(graph); +put_ar_graph: kref_put(&ar_graph->refcount, q6apm_put_audioreach_graph); return ERR_PTR(ret); } -- 2.20.1

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[PATCH] ASoC: nau8821: Add DMIC selections.
by Seven Lee 05 Jan '22

05 Jan '22

Add the following two control options: 1. DMIC pin slew rate selection. 2. DMIC clock speed selection. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- sound/soc/codecs/nau8821.c | 79 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8821.h | 6 ++- 2 files changed, 84 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 2de818377484..52cdbf263e42 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -305,6 +305,66 @@ static int nau8821_biq_coeff_put(struct snd_kcontrol *kcontrol, return 0; } +static int nau8821_slew_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int value; + + regmap_read(nau8821->regmap, NAU8821_R13_DMIC_CTRL, &value); + nau8821->def_mclk_src = (value & NAU8821_DMIC_SLEW_MASK) + >> NAU8821_DMIC_SLEW_SFT; + ucontrol->value.bytes.data[0] = nau8821->def_mclk_src; + + return 0; +} + +static int nau8821_slew_rate_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + nau8821->def_mclk_src = ucontrol->value.integer.value[0]; + + regmap_update_bits(component->regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SLEW_MASK, + nau8821->def_mclk_src << NAU8821_DMIC_SLEW_SFT); + + return 0; +} + +static int nau8821_dmic_clock_speed_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int value; + + regmap_read(nau8821->regmap, NAU8821_R13_DMIC_CTRL, &value); + nau8821->def_dmic_clock = (value & NAU8821_DMIC_SRC_MASK) + >> NAU8821_DMIC_SRC_SFT; + ucontrol->value.bytes.data[0] = nau8821->def_dmic_clock; + + return 0; +} + +static int nau8821_dmic_clock_speed_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + nau8821->def_dmic_clock = ucontrol->value.integer.value[0]; + + regmap_update_bits(component->regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SRC_MASK, + nau8821->def_dmic_clock << NAU8821_DMIC_SRC_SFT); + + return 0; +} + static const char * const nau8821_adc_decimation[] = { "32", "64", "128", "256" }; @@ -319,6 +379,20 @@ static const struct soc_enum nau8821_dac_oversampl_enum = SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); +static const char *const dmic_slew_text[] = {"0", "1", "2", "3", "4", "5", + "6", "7"}; + +static const struct soc_enum nau8821_slew_rate_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_slew_text), + dmic_slew_text); + +static const char *const dmic_clock_speed_text[] = {"3.072", "1.536", "0.768", + "0.384"}; + +static const struct soc_enum nau8821_dmic_clock_speed_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_clock_speed_text), + dmic_clock_speed_text); + static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); @@ -350,6 +424,11 @@ static const struct snd_kcontrol_new nau8821_controls[] = { nau8821_biq_coeff_get, nau8821_biq_coeff_put), SOC_SINGLE("ADC Phase Switch", NAU8821_R1B_TDM_CTRL, NAU8821_ADCPHS_SFT, 1, 0), + SOC_ENUM_EXT("Slew Rate Selection", nau8821_slew_rate_enum, + nau8821_slew_rate_get, nau8821_slew_rate_put), + SOC_ENUM_EXT("DMIC Clock Speed Selection", + nau8821_dmic_clock_speed_enum, nau8821_dmic_clock_speed_get, + nau8821_dmic_clock_speed_put), }; static const struct snd_kcontrol_new nau8821_dmic_mode_switch = diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index a92edfeb9d3a..0ca099fa98bb 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -228,7 +228,7 @@ #define NAU8821_IRQ_INSERT_DIS 0x1 /* DMIC_CTRL (0x13) */ -#define NAU8821_DMIC_DS_SFT 7 +#define NAU8821_DMIC_DS_SFT 11 #define NAU8821_DMIC_DS_MASK (0x1 << NAU8821_DMIC_DS_SFT) #define NAU8821_DMIC_DS_HIGH (0x1 << NAU8821_DMIC_DS_SFT) #define NAU8821_DMIC_DS_LOW (0x0 << NAU8821_DMIC_DS_SFT) @@ -236,6 +236,8 @@ #define NAU8821_DMIC_SRC_MASK (0x3 << NAU8821_DMIC_SRC_SFT) #define NAU8821_CLK_DMIC_SRC (0x2 << NAU8821_DMIC_SRC_SFT) #define NAU8821_DMIC_EN_SFT 0 +#define NAU8821_DMIC_SLEW_SFT 8 +#define NAU8821_DMIC_SLEW_MASK (0x7 << NAU8821_DMIC_SLEW_SFT) /* GPIO12_CTRL (0x1a) */ #define NAU8821_JKDET_PULL_UP (0x1 << 11) /* 0 - pull down, 1 - pull up */ @@ -525,6 +527,8 @@ struct nau8821 { int jack_eject_debounce; int fs; int dmic_clk_threshold; + int def_mclk_src; + int def_dmic_clock; }; int nau8821_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

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[PATCH v2 1/3] ASoC: cs4265: Fix part number ID error message
by Fabio Estevam 04 Jan '22

04 Jan '22

From: Fabio Estevam <festevam(a)denx.de> The Chip ID - Register 01h contains the following description as per the CS4265 datasheet: "Bits 7 through 4 are the part number ID, which is 1101b (0Dh)" The current error message is incorrect as it prints CS4265_CHIP_ID, which is the register number, instead of printing the expected part number ID value. To make it clearer, also do a shift by 4, so that the error message would become: [ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd Signed-off-by: Fabio Estevam <festevam(a)denx.de> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - None sound/soc/codecs/cs4265.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index cffd6111afac..b89002189a2b 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -611,8 +611,8 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, if (devid != CS4265_CHIP_ID_VAL) { ret = -ENODEV; dev_err(&i2c_client->dev, - "CS4265 Device ID (%X). Expected %X\n", - devid, CS4265_CHIP_ID); + "CS4265 Part Number ID: 0x%x Expected: 0x%x\n", + devid >> 4, CS4265_CHIP_ID_VAL >> 4); return ret; } dev_info(&i2c_client->dev, -- 2.25.1

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hda/cs8904: Fix jack detection after resume
by Christian A. Ehrhardt 04 Jan '22

04 Jan '22

Hi, while debugging something different, I found the following jack detection related bug in the cs8904 driver on my DELL Inspirion 3501: - Plug in a headset, play some sound to make sure that the headset is detected. - Stop the sound, wait for the codec to go into suspend (30s should be plenty enough time) - Unplug the headset while the codec is in suspend. - Try to play sound on the internal speakers. Notice that the the driver still thinks the headset is connected. As a result there is no sound on the internal speakers. AFAICS the bug was introduced by the following commit: | commit 424e531b47f83da87490464c5bf633dfb624fe6a | Author: Stefan Binding <sbinding(a)opensource.cirrus.com> | Date: Fri Aug 27 12:02:51 2021 +0100 | | ALSA: hda/cs8409: Ensure Type Detection is only run on startup when | necessary The bug is with the hp_jack_in and mic_jack_in flags that are used to determine if jack detection should be run etc. These flags are cleared in the suspend path. This prevents a status change from being detected after resume as hp_jack_in is cleared by suspend and there is no jack due to the unplug of the headset. This status change was never reported to the HDA core, though. On suspend this is not done, only hd_jack_in is cleared and on resume the driver thinks that there is no change the needs reporting. Proposed fix below, please consider inclusion. regards Christian >From 1dcf34c1d1c6c4852a86ec3ae189afa5d90ea09c Mon Sep 17 00:00:00 2001 From: "Christian A. Ehrhardt" <lk(a)c--e.de> Date: Fri, 31 Dec 2021 12:13:48 +0100 Subject: [PATCH 2/2] ALSA: hda/cs8409: Fix Jack detection after resume The suspend code unconditionally sets ->hp_jack_in and ->mic_jack_in to zero but without reporting this status change to the HDA core. To compensate for this, always assume a status change on the first unsol event after boot or resume. Fixes: 424e531b47f8 ("ALSA: hda/cs8409: Ensure Type Detection is only run on startup when necessary") Signed-off-by: Christian A. Ehrhardt <lk(a)c--e.de> --- sound/pci/hda/patch_cs8409-tables.c | 3 +++ sound/pci/hda/patch_cs8409.c | 5 ++++- sound/pci/hda/patch_cs8409.h | 1 + 3 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 0fb0a428428b..df0b4522babf 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -252,6 +252,7 @@ struct sub_codec cs8409_cs42l42_codec = { .init_seq_num = ARRAY_SIZE(cs42l42_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 0, @@ -443,6 +444,7 @@ struct sub_codec dolphin_cs42l42_0 = { .init_seq_num = ARRAY_SIZE(dolphin_c0_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 0, @@ -456,6 +458,7 @@ struct sub_codec dolphin_cs42l42_1 = { .init_seq_num = ARRAY_SIZE(dolphin_c1_init_reg_seq), .hp_jack_in = 0, .mic_jack_in = 0, + .force_status_change = 1, .paged = 1, .suspended = 1, .no_type_dect = 1, diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index bf5d7f0c6ba5..aff2b5abb81e 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -636,7 +636,9 @@ static void cs42l42_run_jack_detect(struct sub_codec *cs42l42) static int cs42l42_handle_tip_sense(struct sub_codec *cs42l42, unsigned int reg_ts_status) { - int status_changed = 0; + int status_changed = cs42l42->force_status_change; + + cs42l42->force_status_change = 0; /* TIP_SENSE INSERT/REMOVE */ switch (reg_ts_status) { @@ -791,6 +793,7 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) cs42l42->last_page = 0; cs42l42->hp_jack_in = 0; cs42l42->mic_jack_in = 0; + cs42l42->force_status_change = 1; /* Put CS42L42 into Reset */ gpio_data = snd_hda_codec_read(codec, CS8409_PIN_AFG, 0, AC_VERB_GET_GPIO_DATA, 0); diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index ade2b838590c..d0b725c7285b 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -305,6 +305,7 @@ struct sub_codec { unsigned int hp_jack_in:1; unsigned int mic_jack_in:1; + unsigned int force_status_change:1; unsigned int suspended:1; unsigned int paged:1; unsigned int last_page; -- 2.32.0

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