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

[alsa-devel] [PATCH AUTOSEL 5.5 333/542] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Sasha Levin 14 Feb '20

14 Feb '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit 73ac9f5e5b43a5dbadb61f27dae7a971f7ec0d22 ] Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9f743ebae615d..2f582ac7cf789 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -600,6 +600,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b5..0464df785f2ee 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd07364..df0355843a4c1 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 288/542] ASoC: Intel: sof_rt5682: Ignore the speaker amp when there isn't one.
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sam McNally <sammc(a)chromium.org> [ Upstream commit d4b74e218a8d0d6cf58e546627ab9d4d4f2645ab ] Some members of the Google_Hatch family include a rt5682 jack codec, but no speaker amplifier. This uses the same driver (sof_rt5682) as a combination of rt5682 jack codec and max98357a speaker amplifier. Within the sof_rt5682 driver, these cases are not currently distinguishable, relying on a DMI quirk to decide the configuration. This causes an incorrect configuration when only the rt5682 is present on a Google_Hatch device. For CML, the jack codec is used as the primary key when matching, with a possible speaker amplifier described in quirk_data. The two cases of interest are the second and third 10EC5682 entries in snd_soc_acpi_intel_cml_machines[]. The second entry matches the combination of rt5682 and max98357a, resulting in the quirk_data field in the snd_soc_acpi_mach being non-null, pointing at max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker amplifier. The third entry matches just the rt5682, resulting in a null quirk_data. The sof_rt5682 driver's DMI data matching identifies that a speaker amplifier is present for all Google_Hatch family devices. Detect cases where there is no speaker amplifier by checking for a null quirk_data in the snd_soc_acpi_mach and remove the speaker amplifier bit in that case. Signed-off-by: Sam McNally <sammc(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200103124921.v3.1.Ib87c4a7fbb3fc818ea12198e291b… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad8a2b4bc7092..8a13231dee15d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -603,6 +603,14 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -663,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 284/542] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index bf1fb0d8a9306..f69072d2564c8 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -101,10 +101,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 283/542] ALSA: hda/realtek - Apply mic mute LED quirk for Dell E7xx laptops, too
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5fab5829674c279839a7408ab30c71c6dfe726b9 ] Dell E7xx laptops have also mic mute LED that is driven by the dell-laptop platform driver. Bind it with the capture control as already done for other models. A caveat is that the fixup hook for the mic mute LED has to be applied at last, otherwise it results in the invalid override of the callback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205529 Link: https://lore.kernel.org/r/20200105081119.21396-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2ea3528bfb1d..f952df64d5302 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5848,6 +5848,7 @@ enum { ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, + ALC292_FIXUP_DELL_E7X_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, @@ -6543,12 +6544,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE }, - [ALC292_FIXUP_DELL_E7X] = { + [ALC292_FIXUP_DELL_E7X_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC292_FIXUP_DELL_E7X] = { + .type = HDA_FIXUP_FUNC, + .v.func = snd_hda_gen_fixup_micmute_led, + /* micmute fixup must be applied at last */ + .chained_before = true, + .chain_id = ALC292_FIXUP_DELL_E7X_AAMIX, + }, [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 280/542] ALSA: sh: Fix unused variable warnings
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5da116f164ce265e397b8f59af5c39e4a61d61a5 ] Remove unused variables that are left over after the conversion of new PCM ops: sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime' Fixes: 1cc2f8ba0b3e ("ALSA: sh: Convert to the new PCM ops") Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/sh_dac_audio.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index f9e36abc98ac7..725992937e8f1 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -175,7 +175,6 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; if (copy_from_user_toio(chip->data_buffer + pos, src, count)) return -EFAULT; @@ -195,7 +194,6 @@ static int snd_sh_dac_pcm_copy_kernel(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memcpy_toio(chip->data_buffer + pos, src, count); chip->buffer_end = chip->data_buffer + pos + count; @@ -214,7 +212,6 @@ static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memset_io(chip->data_buffer + pos, 0, count); chip->buffer_end = chip->data_buffer + pos + count; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 275/542] ASoC: soc-topology: fix endianness issues
by Sasha Levin 14 Feb '20

14 Feb '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 72bbeda0222bcd382ee33b3aff71346074410c21 ] Sparse complains about a series of easy warnings, fix. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200102195952.9465-3-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-topology.c | 42 +++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4e1fe623c3908..0119f07cece6f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -604,9 +604,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { - if (!sbe->put && ext_ops[i].id == be->ext_ops.put) + if (!sbe->put && + ext_ops[i].id == le32_to_cpu(be->ext_ops.put)) sbe->put = ext_ops[i].put; - if (!sbe->get && ext_ops[i].id == be->ext_ops.get) + if (!sbe->get && + ext_ops[i].id == le32_to_cpu(be->ext_ops.get)) sbe->get = ext_ops[i].get; } @@ -621,11 +623,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = tplg->io_ops_count; for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -638,11 +640,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = ARRAY_SIZE(io_ops); for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -931,7 +933,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, if (se->dobj.control.dtexts == NULL) return -ENOMEM; - for (i = 0; i < ec->items; i++) { + for (i = 0; i < le32_to_cpu(ec->items); i++) { if (strnlen(ec->texts[i], SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) { @@ -1325,7 +1327,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (kc[i].name == NULL) goto err_sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = mc->hdr.access; + kc[i].access = le32_to_cpu(mc->hdr.access); /* we only support FL/FR channel mapping atm */ sm->reg = tplc_chan_get_reg(tplg, mc->channel, @@ -1337,10 +1339,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( sm->rshift = tplc_chan_get_shift(tplg, mc->channel, SNDRV_CHMAP_FR); - sm->max = mc->max; - sm->min = mc->min; - sm->invert = mc->invert; - sm->platform_max = mc->platform_max; + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); sm->dobj.index = tplg->index; INIT_LIST_HEAD(&sm->dobj.list); @@ -1401,7 +1403,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( goto err_se; tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); + le32_to_cpu(ec->priv.size)); dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); @@ -1411,7 +1413,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( if (kc[i].name == NULL) goto err_se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = ec->hdr.access; + kc[i].access = le32_to_cpu(ec->hdr.access); /* we only support FL/FR channel mapping atm */ se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); @@ -1420,8 +1422,8 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); se->dobj.index = tplg->index; switch (le32_to_cpu(ec->hdr.ops.info)) { @@ -1523,9 +1525,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( if (kc[i].name == NULL) goto err_sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = be->hdr.access; + kc[i].access = le32_to_cpu(be->hdr.access); - sbe->max = be->max; + sbe->max = le32_to_cpu(be->max); INIT_LIST_HEAD(&sbe->dobj.list); /* map standard io handlers and check for external handlers */ -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 261/542] ASoC: Intel: kbl_da7219_max98357a: remove unused variable 'constraints_16000' and 'ch_mono'
by Sasha Levin 14 Feb '20

14 Feb '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit c5614fb8e3d13be7bba79f71b798468a3a6224f7 ] sound/soc/intel/boards/kbl_da7219_max98357a.c:343:48: warning: constraints_16000 defined but not used [-Wunused-const-variable=] sound/soc/intel/boards/kbl_da7219_max98357a.c:348:27: warning: ch_mono defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20191224140237.36732-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb69..0d55319a0773c 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 221/542] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d1caa8ed9e681..9985fc139487b 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -119,7 +119,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 175/542] ASoC: SOF: Intel: hda: solve MSI issues by merging ipc and stream irq handlers
by Sasha Levin 14 Feb '20

14 Feb '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 7c11af9fcdc425b80f140a218d4fef9f17734bfc ] The existing code uses two handlers for a shared edge-based MSI interrupts. In corner cases, interrupts are lost, leading to IPC timeouts. Those timeouts do not appear in legacy mode. This patch merges the two handlers and threads into a single one, and simplifies the mask/unmask operations by using a single top-level mask (Global Interrupt Enable). The handler only checks for interrupt sources using the Global Interrupt Status (GIS) field, and all the actual work happens in the thread. This also enables us to remove the use of spin locks. Stream events are prioritized over IPC ones. This patch was tested with HDaudio and SoundWire platforms, and all known IPC timeout issues are solved in MSI mode. The SoundWire-specific patches will be provided in follow-up patches, where the SoundWire interrupts are handled in the same thread as IPC and stream interrupts. 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> Link: https://lore.kernel.org/r/20191204212859.13239-1-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/intel/apl.c | 1 - sound/soc/sof/intel/cnl.c | 5 --- sound/soc/sof/intel/hda-ipc.c | 23 +++-------- sound/soc/sof/intel/hda-stream.c | 20 ++++----- sound/soc/sof/intel/hda.c | 69 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda.h | 11 ++--- 6 files changed, 70 insertions(+), 59 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 7daa8eb456c8d..6f45e14f2b2e3 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -41,7 +41,6 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = hda_dsp_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 0e1e265f3f3b3..9bd169e2691e2 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -106,10 +106,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } @@ -231,7 +227,6 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = cnl_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 0fd2153c17695..1837f66e361fd 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -230,22 +230,15 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } -/* is this IRQ for ADSP ? - we only care about IPC here */ -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) +/* Check if an IPC IRQ occurred */ +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev) { - struct snd_sof_dev *sdev = context; - int ret = IRQ_NONE; + bool ret = false; u32 irq_status; - spin_lock(&sdev->hw_lock); - /* store status */ irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); dev_vdbg(sdev->dev, "irq handler: irq_status:0x%x\n", irq_status); @@ -255,16 +248,10 @@ irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) goto out; /* IPC message ? */ - if (irq_status & HDA_DSP_ADSPIS_IPC) { - /* disable IPC interrupt */ - snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, - HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, 0); - ret = IRQ_WAKE_THREAD; - } + if (irq_status & HDA_DSP_ADSPIS_IPC) + ret = true; out: - spin_unlock(&sdev->hw_lock); return ret; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 29ab432816701..927a36f92c242 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -549,22 +549,23 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, return 0; } -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) { - struct hdac_bus *bus = context; - int ret = IRQ_WAKE_THREAD; + struct hdac_bus *bus = sof_to_bus(sdev); + bool ret = false; u32 status; - spin_lock(&bus->reg_lock); + /* The function can be called at irq thread, so use spin_lock_irq */ + spin_lock_irq(&bus->reg_lock); status = snd_hdac_chip_readl(bus, INTSTS); dev_vdbg(bus->dev, "stream irq, INTSTS status: 0x%x\n", status); - /* Register inaccessible, ignore it.*/ - if (status == 0xffffffff) - ret = IRQ_NONE; + /* if Register inaccessible, ignore it.*/ + if (status != 0xffffffff) + ret = true; - spin_unlock(&bus->reg_lock); + spin_unlock_irq(&bus->reg_lock); return ret; } @@ -602,7 +603,8 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) { - struct hdac_bus *bus = context; + struct snd_sof_dev *sdev = context; + struct hdac_bus *bus = sof_to_bus(sdev); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) u32 rirb_status; #endif diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fb17b87b684bf..82ecadda886c6 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -499,6 +499,49 @@ static const struct sof_intel_dsp_desc return chip_info; } +static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* + * Get global interrupt status. It includes all hardware interrupt + * sources in the Intel HD Audio controller. + */ + if (snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS) & + SOF_HDA_INTSTS_GIS) { + + /* disable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + 0); + + return IRQ_WAKE_THREAD; + } + + return IRQ_NONE; +} + +static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* deal with streams and controller first */ + if (hda_dsp_check_stream_irq(sdev)) + hda_dsp_stream_threaded_handler(irq, sdev); + + if (hda_dsp_check_ipc_irq(sdev)) + sof_ops(sdev)->irq_thread(irq, sdev); + + /* enable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_GLOBAL_EN); + + return IRQ_HANDLED; +} + int hda_dsp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); @@ -603,9 +646,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) */ if (hda_use_msi && pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI) > 0) { dev_info(sdev->dev, "use msi interrupt mode\n"); - hdev->irq = pci_irq_vector(pci, 0); - /* ipc irq number is the same of hda irq */ - sdev->ipc_irq = hdev->irq; + sdev->ipc_irq = pci_irq_vector(pci, 0); /* initialised to "false" by kzalloc() */ sdev->msi_enabled = true; } @@ -616,28 +657,17 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) * in IO-APIC mode, hda->irq and ipc_irq are using the same * irq number of pci->irq */ - hdev->irq = pci->irq; sdev->ipc_irq = pci->irq; } - dev_dbg(sdev->dev, "using HDA IRQ %d\n", hdev->irq); - ret = request_threaded_irq(hdev->irq, hda_dsp_stream_interrupt, - hda_dsp_stream_threaded_handler, - IRQF_SHARED, "AudioHDA", bus); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to register HDA IRQ %d\n", - hdev->irq); - goto free_irq_vector; - } - dev_dbg(sdev->dev, "using IPC IRQ %d\n", sdev->ipc_irq); - ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_ipc_irq_handler, - sof_ops(sdev)->irq_thread, IRQF_SHARED, - "AudioDSP", sdev); + ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_interrupt_handler, + hda_dsp_interrupt_thread, + IRQF_SHARED, "AudioDSP", sdev); if (ret < 0) { dev_err(sdev->dev, "error: failed to register IPC IRQ %d\n", sdev->ipc_irq); - goto free_hda_irq; + goto free_irq_vector; } pci_set_master(pci); @@ -668,8 +698,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) free_ipc_irq: free_irq(sdev->ipc_irq, sdev); -free_hda_irq: - free_irq(hdev->irq, bus); free_irq_vector: if (sdev->msi_enabled) pci_free_irq_vectors(pci); @@ -715,7 +743,6 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) SOF_HDA_PPCTL_GPROCEN, 0); free_irq(sdev->ipc_irq, sdev); - free_irq(hda->irq, bus); if (sdev->msi_enabled) pci_free_irq_vectors(pci); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 18d7e72bf9b72..63df888dddb6c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -43,11 +43,14 @@ /* SOF_HDA_GCTL register bist */ #define SOF_HDA_GCTL_RESET BIT(0) -/* SOF_HDA_INCTL and SOF_HDA_INTSTS regs */ +/* SOF_HDA_INCTL regs */ #define SOF_HDA_INT_GLOBAL_EN BIT(31) #define SOF_HDA_INT_CTRL_EN BIT(30) #define SOF_HDA_INT_ALL_STREAM 0xff +/* SOF_HDA_INTSTS regs */ +#define SOF_HDA_INTSTS_GIS BIT(31) + #define SOF_HDA_MAX_CAPS 10 #define SOF_HDA_CAP_ID_OFF 16 #define SOF_HDA_CAP_ID_MASK GENMASK(SOF_HDA_CAP_ID_OFF + 11,\ @@ -406,8 +409,6 @@ struct sof_intel_hda_dev { /* the maximum number of streams (playback + capture) supported */ u32 stream_max; - int irq; - /* PM related */ bool l1_support_changed;/* during suspend, is L1SEN changed or not */ @@ -511,11 +512,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params); int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, int cmd); -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context); irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context); int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct hdac_stream *stream); +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction); @@ -540,7 +542,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context); irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 174/542] ASoC: intel: sof_rt5682: Add support for tgl-max98357a-rt5682
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> [ Upstream commit 6605f0ca3af3b964635287ec7c9dadc812b78eb0 ] This patch adds the driver data and updates quirk info for tgl with max98357a speaker amp and ALC5682 headset codec. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191126143205.21987-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 57adadacbf436..ad8a2b4bc7092 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -598,6 +598,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + dmi_check_system(sof_rt5682_quirk_table); if (soc_intel_is_byt() || soc_intel_is_cht()) { @@ -691,6 +694,21 @@ static int sof_rt5682_remove(struct platform_device *pdev) return 0; } +static const struct platform_device_id board_ids[] = { + { + .name = "sof_rt5682", + }, + { + .name = "tgl_max98357a_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { } +}; + static struct platform_driver sof_audio = { .probe = sof_audio_probe, .remove = sof_rt5682_remove, @@ -698,6 +716,7 @@ static struct platform_driver sof_audio = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, }, + .id_table = board_ids, }; module_platform_driver(sof_audio) @@ -707,3 +726,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); +MODULE_ALIAS("platform:tgl_max98357a_rt5682"); -- 2.20.1

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