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

[PATCH AUTOSEL 5.7 307/388] ASoC: SOF: Intel: hda: fix generic hda codec support
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 89d73ccab20a684d8446cea4d8ac6a2608c8d390 ] Add support for using generic codec driver with SOF. Generic driver is used if: - snd_sof_intel_hda_common.hda_model="generic" is set, or - fallback if no other codec driver is found The implementation is aligned with snd-hda-intel driver, and fixes audio support for systems like Acer Swift 3 SF314-57G, on which this issue was originally reported. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> BugLink: https://github.com/thesofproject/linux/issues/1807 BugLink: https://bugs.launchpad.net/bugs/1877757 Link: https://lore.kernel.org/r/20200529160358.12134-1-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 51 +++++++++++++++++++++++++++++---- 1 file changed, 45 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 3041fbbb010a..ea021db697b8 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -24,19 +24,44 @@ #define IDISP_VID_INTEL 0x80860000 /* load the legacy HDA codec driver */ -static int hda_codec_load_module(struct hda_codec *codec) +static int request_codec_module(struct hda_codec *codec) { #ifdef MODULE char alias[MODULE_NAME_LEN]; - const char *module = alias; + const char *mod = NULL; - snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); - dev_dbg(&codec->core.dev, "loading codec module: %s\n", module); - request_module(module); + switch (codec->probe_id) { + case HDA_CODEC_ID_GENERIC: +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + mod = "snd-hda-codec-generic"; #endif + break; + default: + snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); + mod = alias; + break; + } + + if (mod) { + dev_dbg(&codec->core.dev, "loading codec module: %s\n", mod); + request_module(mod); + } +#endif /* MODULE */ return device_attach(hda_codec_dev(codec)); } +static int hda_codec_load_module(struct hda_codec *codec) +{ + int ret = request_codec_module(codec); + + if (ret <= 0) { + codec->probe_id = HDA_CODEC_ID_GENERIC; + ret = request_codec_module(codec); + } + + return ret; +} + /* enable controller wake up event for all codecs with jack connectors */ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) { @@ -78,6 +103,13 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) {} EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(bus) \ + ((bus)->modelname && !strcmp((bus)->modelname, "generic")) +#else +#define is_generic_config(x) 0 +#endif + /* probe individual codec */ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, bool hda_codec_use_common_hdmi) @@ -87,6 +119,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; + struct hda_codec *codec; u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; @@ -108,6 +141,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->codec.bus = hbus; hdev = &hda_priv->codec.core; + codec = &hda_priv->codec; ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); if (ret < 0) @@ -122,6 +156,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->need_display_power = true; } + if (is_generic_config(hbus)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + /* * if common HDMI codec driver is not used, codec load * is skipped here and hdac_hdmi is used instead @@ -129,7 +168,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (hda_codec_use_common_hdmi || (resp & 0xFFFF0000) != IDISP_VID_INTEL) { hdev->type = HDA_DEV_LEGACY; - ret = hda_codec_load_module(&hda_priv->codec); + ret = hda_codec_load_module(codec); /* * handle ret==0 (no driver bound) as an error, but pass * other return codes without modification -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 306/388] 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 e7178817d7a7..1ee10eafe3e6 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -252,6 +252,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, 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 5.7 294/388] ASoC: dapm: Move dai_link widgets to runtime to fix use after free
by Sasha Levin 18 Jun '20

18 Jun '20

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f4aa5e214eeaf7f1c7f157526a5aa29784cb6a1f ] The newly added CODEC to CODEC DAI link widget pointers in snd_soc_dai_link are better placed in snd_soc_pcm_runtime. snd_soc_dai_link is really intended for static configuration of the DAI, and the runtime for dynamic data. The snd_soc_dai_link structures are not destroyed if the card is unbound. The widgets are cleared up on unbind, however if the card is rebound as the snd_soc_dai_link structures are reused these pointers will be left at their old values, causing access to freed memory. Fixes: 595571cca4de ("ASoC: dapm: Fix regression introducing multiple copies of DAI widgets") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200526161930.30759-1-ckeepax@opensource.cirrus.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 6 +++--- sound/soc/soc-dapm.c | 12 ++++++------ 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0371e70242d..8e480efeda2a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,9 +790,6 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; - struct snd_soc_dapm_widget *playback_widget; - struct snd_soc_dapm_widget *capture_widget; - unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ @@ -1156,6 +1153,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai **cpu_dais; unsigned int num_cpus; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + struct delayed_work delayed_work; void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e2632841b321..c0aa64ff8e32 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4340,16 +4340,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (dai_link->params && !dai_link->playback_widget) { + if (dai_link->params && !rtd->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; dai = snd_soc_dapm_new_dai(card, substream, "playback"); if (IS_ERR(dai)) goto capture; - dai_link->playback_widget = dai; + rtd->playback_widget = dai; } dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, - dai_link->playback_widget, + rtd->playback_widget, codec_dai, codec); } @@ -4358,16 +4358,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (dai_link->params && !dai_link->capture_widget) { + if (dai_link->params && !rtd->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; dai = snd_soc_dapm_new_dai(card, substream, "capture"); if (IS_ERR(dai)) return; - dai_link->capture_widget = dai; + rtd->capture_widget = dai; } dapm_connect_dai_routes(&card->dapm, codec_dai, codec, - dai_link->capture_widget, + rtd->capture_widget, cpu_dai, capture_cpu); } } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 279/388] 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 a495d1050d49..e30b66b94bf6 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 5.7 232/388] 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 08f4ae964b02..fbfd53874b47 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 5.7 217/388] ASoC: codecs: rt*-sdw: fix memory leak in set_sdw_stream()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 07b542fe831cbefce163ad1b3aa7292c8a6332b8 ] Now that the sdw_stream is allocated in machine driver, set_sdw_stream() is also called with a NULL argument during the dailink shutdown. In this case, the drivers should not allocate any memory, and just return. Detected with KASAN/kmemleak. 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: Rander Wang <rander.wang(a)intel.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Shuming Fan <shumingf(a)realtek.com> Cc: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/20200515211531.11416-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1308-sdw.c | 3 +++ sound/soc/codecs/rt5682.c | 3 +++ sound/soc/codecs/rt700.c | 3 +++ sound/soc/codecs/rt711.c | 3 +++ sound/soc/codecs/rt715.c | 3 +++ 5 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index a5a7e46de246..a7f45191364d 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -482,6 +482,9 @@ static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d36f560ad7a8..c4892af14850 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2958,6 +2958,9 @@ static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index ff68f0e4f629..687ac2153666 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -860,6 +860,9 @@ static int rt700_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 2daed7692a3b..65b59dbfb43c 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -906,6 +906,9 @@ static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 2cbc57b16b13..099c8bd20006 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -530,6 +530,9 @@ static int rt715_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 210/388] 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 b50965ab3b3a..d61c2f1095b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -404,6 +404,8 @@ 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; @@ -502,12 +504,15 @@ 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

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[PATCH AUTOSEL 5.7 209/388] ALSA: usb-audio: fixing upper volume limit for RME Babyface Pro routing crosspoints
by Sasha Levin 18 Jun '20

18 Jun '20

From: Thomas Ebeling <penguins(a)bollie.de> [ Upstream commit 47b4f5f5b65680fbef7a7a9a4796b35f38a6e43e ] In my initial patch, these were set too low. Fixes: 3e8f3bd04716 ("ALSA: usb-audio: RME Babyface Pro mixer patch") Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> Link: https://lore.kernel.org/r/20200515114556.vtspnonzvp4xp44m@bollie.ca9.eu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index bdff8674942e..aad2683ff793 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2191,7 +2191,7 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) * These devices exposes a couple of DSP functions via request to EP0. * Switches are available via control registers, while routing is controlled * by controlling the volume on each possible crossing point. - * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 65536) with * 0dB being at dec. 32768. */ enum { @@ -2220,7 +2220,7 @@ enum { #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf -#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB +#define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 -- 2.25.1

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[PATCH AUTOSEL 5.7 195/388] 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 3d41ca2238d4..4f33ddb7b441 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

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[PATCH AUTOSEL 5.7 194/388] 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

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