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

[PATCH AUTOSEL 5.4 251/266] ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ] With additional checks on dailinks, we see errors such as [ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd NoCodec-6 does not support playback It's not clear why we set the dpcm_playback and dpcm_capture flags unconditionally, add a check on number of channels for each direction to avoid invalid configurations. Fixes: 8017b8fd37bf5e ('ASoC: SOF: Add Nocodec machine driver support') 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/20200608194415.4663-5-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/nocodec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 3d128e5a132c..ea0fe9a09f3f 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 250/266] 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 0215e2c94bf0..9df20768a8f2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1895,9 +1895,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->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 5.4 209/266] 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 01052a0808b0..5aee6b8366d2 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 5.4 190/266] 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 fdd2c73fd2fa..869fe0068cbd 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -482,6 +482,7 @@ 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 5.4 157/266] 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 e62e1d7815aa..f222e1091bae 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -742,6 +742,18 @@ 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

[PATCH AUTOSEL 5.4 145/266] ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback
by Sasha Levin 18 Jun '20

18 Jun '20

From: Erwin Burema <e.burema(a)gmail.com> [ Upstream commit 10ce77e4817fef99e1166be7e6685a80c63bf77f ] For USB sound devices using implicit feedback the endpoint used for this feedback should be able to be opened twice, once for required feedback and second time for audio data. This way these devices can be put in duplex audio mode. Since this only works if the settings of the endpoint don't change a check is included for this. This fixes bug 207023 ("MOTU M2 regression on duplex audio") and should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device will not operate full-duplex") Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series") Signed-off-by: Erwin Burema <e.burema(a)gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751 Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++- sound/usb/pcm.c | 5 ++ 3 files changed, 197 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 820e564656ed..d6219fba9699 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -108,6 +108,7 @@ struct snd_usb_endpoint { int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ + bool is_implicit_feedback; /* This endpoint is used as implicit feedback */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 50104f658ed4..9bea7d3f99f8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_add_tail(&ep->list, &chip->ep_list); + ep->is_implicit_feedback = 0; + __exit_unlock: mutex_unlock(&chip->mutex); @@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * Check data endpoint for format differences + */ +static bool check_ep_params(struct snd_usb_endpoint *ep, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs; + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); + bool ret = 1; + + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->stride == frame_bits >> 3); + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ret = ret && (ep->silence_value == 0x80); + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ret = ret && (ep->silence_value == 0x69); + break; + default: + ret = ret && (ep->silence_value == 0); + } + + /* assume max. frequency is 50% higher than nominal */ + ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1)); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { + /* whatever fits into a max. size packet */ + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval)); + } + + if (ep->fill_max) + ret = ret && (ep->curpacksize == ep->maxpacksize); + else + ret = ret && (ep->curpacksize == maxsize); + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { + packs_per_ms = 8 >> ep->datainterval; + max_packs_per_urb = MAX_PACKS_HS; + } else { + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; + } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, urb_packs); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ret = ret && (ep->nurbs == MAX_URBS); + + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ret = ret && (ep->max_urb_frames == + DIV_ROUND_UP(frames_per_period, urbs_per_period)); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ret = ret && (ep->nurbs == min(max_urbs, + urbs_per_period * periods_per_buffer)); + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->maxpacksize == fmt->maxpacksize); + ret = ret && + (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)); + + return ret; +} + /* * configure a data endpoint */ @@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int err; if (ep->use_count != 0) { - usb_audio_warn(ep->chip, - "Unable to change format on ep #%x: already in use\n", - ep->ep_num); - return -EBUSY; + bool check = ep->is_implicit_feedback && + check_ep_params(ep, pcm_format, + channels, period_bytes, + period_frames, buffer_periods, + fmt, sync_ep); + + if (!check) { + usb_audio_warn(ep->chip, + "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + usb_audio_dbg(ep->chip, + "Ep #%x already in use as implicit feedback but format not changed\n", + ep->ep_num); + return 0; } /* release old buffers, if any */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e52d129085c0..6c391e5fad2a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -386,6 +386,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, if (!subs->sync_endpoint) return -EINVAL; + subs->sync_endpoint->is_implicit_feedback = 1; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 1; @@ -484,12 +486,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb ? SND_USB_ENDPOINT_TYPE_DATA : SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) { if (is_playback && attr == USB_ENDPOINT_SYNC_NONE) return 0; return -EINVAL; } + subs->sync_endpoint->is_implicit_feedback = implicit_fb; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 136/266] ASoC: ti: omap-mcbsp: Fix an error handling path in 'asoc_mcbsp_probe()'
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 03990fd58d2b7c8f7d53e514ba9b8749fac260f9 ] If an error occurs after the call to 'omap_mcbsp_init()', the reference to 'mcbsp->fclk' must be decremented, as already done in the remove function. This can be achieved easily by using the devm_ variant of 'clk_get()' when the reference is taken in 'omap_mcbsp_init()' This fixes the leak in the probe and has the side effect to simplify both the error handling path of 'omap_mcbsp_init()' and the remove function. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Acked-by: Peter Ujfalusi <peter.ujflausi(a)ti.com> Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 26b503bbdb5f..3273b317fa3b 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -686,7 +686,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_CAPTURE); - mcbsp->fclk = clk_get(&pdev->dev, "fck"); + mcbsp->fclk = devm_clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); @@ -711,7 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); - goto err_thres; + return ret; } } @@ -724,8 +724,6 @@ static int omap_mcbsp_init(struct platform_device *pdev) err_st: if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); return ret; } @@ -1442,8 +1440,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) omap_mcbsp_st_cleanup(pdev); - clk_put(mcbsp->fclk); - return 0; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 135/266] ASoC: ux500: mop500: Fix some refcounted resources issues
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 4e8748fcaeec073e3ba794871ce86c545e4f961f ] There are 2 issues here: - if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()' is a no-op because the 'mop500_dai_links' structure has not been initialized yet, so the referenced are not decremented - The reference stored in 'mop500_dai_links[i].codecs' is refcounted only once in the probe and must be decremented only once. Fixes: 39013bd60e79 ("ASoC: Ux500: Dispose of device nodes correctly") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ux500/mop500.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 2873e8e6f02b..cdae1190b930 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,11 @@ static void mop500_of_node_put(void) { int i; - for (i = 0; i < 2; i++) { + for (i = 0; i < 2; i++) of_node_put(mop500_dai_links[i].cpus->of_node); - of_node_put(mop500_dai_links[i].codecs->of_node); - } + + /* Both links use the same codec, which is refcounted only once */ + of_node_put(mop500_dai_links[0].codecs->of_node); } static int mop500_of_probe(struct platform_device *pdev, @@ -81,7 +82,9 @@ static int mop500_of_probe(struct platform_device *pdev, if (!(msp_np[0] && msp_np[1] && codec_np)) { dev_err(&pdev->dev, "Phandle missing or invalid\n"); - mop500_of_node_put(); + for (i = 0; i < 2; i++) + of_node_put(msp_np[i]); + of_node_put(codec_np); return -EINVAL; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 134/266] SoC: rsnd: add interrupt support for SSI BUSIF buffer
by Sasha Levin 18 Jun '20

18 Jun '20

From: Yongbo Zhang <giraffesnn123(a)gmail.com> [ Upstream commit 66c705d07d784fb6b4622c6e47b6acae357472db ] SSI BUSIF buffer is possible to overflow or underflow, especially in a hypervisor environment. If there is no interrupt support, it will eventually lead to errors in pcm data. This patch adds overflow and underflow interrupt support for SSI BUSIF buffer. Reported-by: Chen Li <licheng0822(a)thundersoft.com> Signed-off-by: Yongbo Zhang <giraffesnn123(a)gmail.com> Tested-by: Chen Li <licheng0822(a)thundersoft.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/gen.c | 8 +++ sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/ssi.c | 145 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 162 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index af19010b9d88..8bd49c8a9517 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ea6cbaa9743e..d47608ff5fac 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -189,6 +189,14 @@ enum rsnd_reg { SSI_SYS_STATUS5, SSI_SYS_STATUS6, SSI_SYS_STATUS7, + SSI_SYS_INT_ENABLE0, + SSI_SYS_INT_ENABLE1, + SSI_SYS_INT_ENABLE2, + SSI_SYS_INT_ENABLE3, + SSI_SYS_INT_ENABLE4, + SSI_SYS_INT_ENABLE5, + SSI_SYS_INT_ENABLE6, + SSI_SYS_INT_ENABLE7, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, @@ -237,6 +245,7 @@ enum rsnd_reg { #define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) #define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) +#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i)) struct rsnd_priv; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4a7d3413917f..47d5ddb526f2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, u32 wsr = ssi->wsr; int width; int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); @@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } + /* enable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable |= 0xf << (id * 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable |= 0xf << 4; + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (!rsnd_ssi_is_run_mods(mod, io)) return 0; @@ -517,6 +559,38 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, ssi->wsr = 0; } + /* disable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable &= ~(0xf << (id * 4)); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable &= ~(0xf << 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + return 0; } @@ -622,6 +696,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, int enable) { u32 val = 0; + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (rsnd_is_gen1(priv)) return 0; @@ -635,6 +714,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (enable) val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000; + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 9: + val |= 0x0000ff00; + break; + } + } + rsnd_mod_write(mod, SSI_INT_ENABLE, val); return 0; @@ -651,6 +743,12 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, u32 status; bool elapsed = false; bool stop = false; + int id = rsnd_mod_id(mod); + int i; + int is_tdm, is_tdm_split; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); spin_lock(&priv->lock); @@ -672,6 +770,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, stop = true; } + status = 0; + + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS(i * 2)); + status &= 0xf << (id * 4); + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS(i * 2), + 0xf << (id * 4)); + stop = true; + break; + } + } + break; + case 9: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS((i * 2) + 1)); + status &= 0xf << 4; + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1), + 0xf << 4); + stop = true; + break; + } + } + break; + } + } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 130/266] ASoC: SOF: core: fix error return code in sof_probe_continue()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Wei Yongjun <weiyongjun1(a)huawei.com> [ Upstream commit 7d8785bc7adbb4dc5ba8ee06994107637848ded8 ] Fix to return negative error code -ENOMEM from the IPC init error handling case instead of 0, as done elsewhere in this function. Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 12aec140819a..2a6b84d2781e 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -372,6 +372,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) /* init the IPC */ sdev->ipc = snd_sof_ipc_init(sdev); if (!sdev->ipc) { + ret = -ENOMEM; dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); goto ipc_err; } -- 2.25.1

1 0