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

[PATCH AUTOSEL 4.14 086/108] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-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/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e1b97e59275a..15d7e6da0555 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 041/108] 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 ed87cc83eb47..9dbcbb27c28e 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -81,6 +81,10 @@ 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 8caf0b57f9c6..841219560872 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); @@ -889,10 +914,17 @@ 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; @@ -951,6 +983,7 @@ 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 ff38fca1781b..fd73186d6003 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1484,6 +1484,8 @@ 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

1 0

[PATCH AUTOSEL 4.14 012/108] 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

1 0

[PATCH AUTOSEL 4.14 009/108] 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 e10e03800cce..6991718d7c8a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1747,8 +1747,10 @@ 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

1 0

[PATCH AUTOSEL 4.19 166/172] 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 7e3b47eeea04..9185bd7c5a6d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3656,6 +3656,12 @@ 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, @@ -3733,6 +3739,14 @@ 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

1 0

[PATCH AUTOSEL 4.19 165/172] 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 5dfc17987b2b..0dcd249877c5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -724,6 +724,18 @@ 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

1 0

[PATCH AUTOSEL 4.19 164/172] ASoC: core: only convert non DPCM link to DPCM link
by Sasha Levin 18 Jun '20

18 Jun '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ] Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e45dfcb62b6a..595fe20bbc6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1920,9 +1920,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platform_name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 133/172] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-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/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 1033ac6631b0..b9ac448989ed 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -241,6 +241,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 119/172] ASoC: fix incomplete error-handling in img_i2s_in_probe.
by Sasha Levin 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> [ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ] Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()" if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding "pm_runtime_put()" into this error path. Fixes: f65bb92ca12e ("ASoC: img-i2s-in: Add runtime PM") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-in.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 388cefd7340a..c22880aea82a 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -485,6 +485,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (IS_ERR(rst)) { if (PTR_ERR(rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; + pm_runtime_put(&pdev->dev); goto err_suspend; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 098/172] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ] The Toshiba Encore WT8-A tablet almost fully works with the default settings for non-CR Bay Trail devices. The only problem is that its jack-detect switch is not inverted (it is active high instead of the normal active low). Add a quirk for this model using the default settings + BYT_RT5640_JD_NOT_INV. 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/20200518072416.5348-1-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 f29014a7d672..5dfc17987b2b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -712,6 +712,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT8-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0