June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 4.19 071/172] soundwire: slave: don't init debugfs on device registration error
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8893ab5e8ee5d7c12e0fc1dca4a309475064473d ] The error handling flow seems incorrect, there is no reason to try and add debugfs support if the device registration did not succeed. Return on error. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20200419185117.4233-2-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/slave.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index ac103bd0c176..b6330b6672d5 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -55,6 +55,8 @@ static int sdw_slave_add(struct sdw_bus *bus, list_del(&slave->node); mutex_unlock(&bus->bus_lock); put_device(&slave->dev); + + return ret; } return ret; -- 2.25.1

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[PATCH AUTOSEL 4.19 069/172] slimbus: ngd: get drvdata from correct device
by Sasha Levin 18 Jun '20

18 Jun '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit b58c663059b484f7ff547d076a34cf6d7a302e56 ] Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200417093618.7929-1-srinivas.kandagatla@linaro.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 9221ba7b7863..f40ac8dcb081 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1350,7 +1350,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1365,12 +1364,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.25.1

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[PATCH AUTOSEL 4.19 065/172] ALSA: usb-audio: Fix racy list management in output queue
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ] The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 36e255d88937..48611849b79b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -359,17 +359,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) ep->next_packet_read_pos %= MAX_URBS; /* take URB out of FIFO */ - if (!list_empty(&ep->ready_playback_urbs)) + if (!list_empty(&ep->ready_playback_urbs)) { ctx = list_first_entry(&ep->ready_playback_urbs, struct snd_urb_ctx, ready_list); + list_del_init(&ctx->ready_list); + } } spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) return; - list_del_init(&ctx->ready_list); - /* copy over the length information */ for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; -- 2.25.1

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[PATCH AUTOSEL 4.19 064/172] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 7f11655bde50..36198ddb4754 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -83,6 +83,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index aeb74cc6ceff..36e255d88937 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -887,10 +912,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -949,6 +981,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8b91be394407..42ae774f6db7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1558,6 +1558,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

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[PATCH AUTOSEL 4.19 015/172] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..13c8e6542a2f 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

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[PATCH AUTOSEL 4.19 014/172] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b06f7d52faad..9ab82ead1825 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,6 +94,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -3862,11 +3863,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -3877,7 +3876,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -3886,6 +3886,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5476,7 +5477,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

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[PATCH AUTOSEL 4.19 011/172] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 14ab16e1369f..1203de8aab99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1758,8 +1758,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

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[PATCH AUTOSEL 4.19 003/172] ASoC: tegra: tegra_wm8903: Support nvidia, headset property
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dmitry Osipenko <digetx(a)gmail.com> [ Upstream commit 3ef9d5073b552d56bd6daf2af1e89b7e8d4df183 ] The microphone-jack state needs to be masked in a case of a 4-pin jack when microphone and ground pins are shorted. Presence of nvidia,headset tells that WM8903 CODEC driver should mask microphone's status if short circuit is detected, i.e headphones are inserted. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/tegra/tegra_wm8903.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 69bc9461974b..301850df368d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -173,6 +173,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = codec_dai->component; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + int shrt = 0; if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -185,12 +186,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) &tegra_wm8903_hp_jack_gpio); } + if (of_property_read_bool(card->dev->of_node, "nvidia,headset")) + shrt = SND_JACK_MICROPHONE; + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, &tegra_wm8903_mic_jack, tegra_wm8903_mic_jack_pins, ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, - 0); + shrt); snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS"); -- 2.25.1

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[PATCH AUTOSEL 5.4 253/266] ASoC: rt5645: Add platform-data for Asus T101HA
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ] The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 19662ee330d6..c83f7f5da96b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3702,6 +3708,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { -- 2.25.1

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[PATCH AUTOSEL 5.4 252/266] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ] The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f222e1091bae..f7964d1ec486 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

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