July 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v0 0/3] ASoC: max9867: driver update
by Ladislav Michl 30 Jul '18

30 Jul '18

Hi there, after this patch serie, I'm finally able to use max9867 codec. This brings in a question how was this driver tested and whenever these tests can be repeated with this patchset applied. Anyway, this serie is intentionally marked as v0 as I'm unclear about few things described in comment section of respective patches. I'd be very happy for answers as they are missing in documentation and are not obvious from other drivers code. Thank you in advance, ladis

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Re: [alsa-devel] [PATCH 10/10] ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits
by Oleksandr Andrushchenko 30 Jul '18

30 Jul '18

Hello, Takashi! > Date: Wed, 25 Jul 2018 23:19:45 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > To: alsa-devel(a)alsa-project.org > Subject: [alsa-devel] [PATCH 10/10] ALSA: xen: Use standard > pcm_format_to_bits() for ALSA format bits > Message-ID: <20180725211945.19222-11-tiwai(a)suse.de> > > The open codes with the bit shift in xen_snd_front_alsa.c give sparse > warnings as the PCM format type is with __bitwise. > There is already a standard macro to get the format bits, so let's use > it instead. > > This fixes sparse warnings like: > sound/xen/xen_snd_front_alsa.c:191:47: warning: restricted snd_pcm_format_t degrades to integer > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > sound/xen/xen_snd_front_alsa.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c > index 5a2bd70a2fa1..129180e17db1 100644 > --- a/sound/xen/xen_snd_front_alsa.c > +++ b/sound/xen/xen_snd_front_alsa.c > @@ -188,7 +188,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats) > > mask = 0; > for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) > - if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) > + if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats) > mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; > > return mask; > @@ -202,7 +202,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats) > mask = 0; > for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) > if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) > - mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; > + mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa); > > return mask; > } Thank you for your patch, Reviewed-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Thank you, Oleksandr

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[alsa-devel] [PATCH v2] ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally
by Pierre-Louis Bossart 30 Jul '18

30 Jul '18

Make this helper inline function available for all platforms. This helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not defined. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: add <asm/page.h> to solve compilation issue on ALPHA reported by 0-day include/sound/memalloc.h | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; /* @@ -67,6 +69,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -90,14 +100,6 @@ struct snd_sg_buf { struct device *dev; }; -/* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - /* * return the physical address at the corresponding offset */ -- 2.14.1

3 2

[alsa-devel] [PATCH 2/2] ALSA: hda - Sleep for 10ms after entering D3 on Conexant codecs
by Park Ju Hyung 28 Jul '18

28 Jul '18

On rare occasions, we are still noticing that the internal speaker spitting out spurious noises even after adding the problematic codec to the list. Adding a 10ms artificial delay before rebooting fixes the issue entirely. Patch for Realtek codecs also adds the same amount of delay after entering D3. Signed-off-by: Park Ju Hyung <qkrwngud825(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 909a880f5e01..1a8a2d440fbd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -226,6 +226,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); } static void cx_auto_free(struct hda_codec *codec) -- 2.18.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Turn CX8200 into D3 as well upon reboot
by Park Ju Hyung 28 Jul '18

28 Jul '18

As an equivalent codec with CX20724, CX8200 is also subject to the reboot bug. Late 2017 and 2018 LG Gram and some HP Spectre laptops are known victims to this issue, causing extremely loud noises upon reboot. Now that we know that this bug is subject to multiple codecs, fix the comment as well. Signed-off-by: Park Ju Hyung <qkrwngud825(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f641c20095f7..909a880f5e01 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -211,6 +211,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x14f12008: /* CX8200 */ case 0x14f150f2: /* CX20722 */ case 0x14f150f4: /* CX20724 */ break; @@ -218,7 +219,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) return; } - /* Turn the CX20722 codec into D3 to avoid spurious noises + /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); -- 2.18.0

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[alsa-devel] Patch "ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()" has been added to the 4.17-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() to the 4.17-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch and it can be found in the queue-4.17 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:14:31 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:37 +0200 Subject: ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2019,13 +2019,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.17/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.17/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

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[alsa-devel] Patch "ASoC: topology: Add missing clock gating parameter when parsing hw_configs" has been added to the 4.17-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Add missing clock gating parameter when parsing hw_configs to the 4.17-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch and it can be found in the queue-4.17 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:14:31 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:38 +0200 Subject: ASoC: topology: Add missing clock gating parameter when parsing hw_configs From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2006,6 +2006,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.17/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.17/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

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[alsa-devel] Patch "ASoC: topology: Add missing clock gating parameter when parsing hw_configs" has been added to the 4.14-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Add missing clock gating parameter when parsing hw_configs to the 4.14-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch and it can be found in the queue-4.14 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:25:26 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:38 +0200 Subject: ASoC: topology: Add missing clock gating parameter when parsing hw_configs From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1997,6 +1997,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.14/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.14/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

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[alsa-devel] Patch "ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()" has been added to the 4.14-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() to the 4.14-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch and it can be found in the queue-4.14 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:25:26 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:37 +0200 Subject: ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2010,13 +2010,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.14/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.14/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

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[alsa-devel] Support for Focusrite 18i20 audio interface
by debian＠callistix.net 27 Jul '18

27 Jul '18

Hi everyone, I own a Focusrite Scarlett 18i20 2nd Gen audio interface. The inputs and outputs are supported by alsa out of the box, with jack all of them are visible and patchable. But there are some functions which are not exposed by alsa, which include: - low latency monitoring, apparently you are supposed to be able to patch any input to any headphone output - selection of sample rates Is someone working on adding this functionality? As an owner of this hardware I would be able to support any dev who's willing to work on this. User manuals: https://us.focusrite.com/downloads?product=Scarlett+18i20 Best regards, vitaminx

1 0

[alsa-devel] [PATCH 00/12] mfd: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 27 Jul '18

27 Jul '18

Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback and capture paths. This patchset also depends on the SLIMbus Stream apis which is already merged via char-misc tree for 4.19. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Srinivas Kandagatla (12): mfd: dt-bindings: Add wcd9335 mfd bindings mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: dt-bindings: add dt bindings for wcd9335 audio codec ASoC: core: add support to snd_soc_dai_get_channel_map() ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: Add support to Analog audio via WCD9335 slim .../devicetree/bindings/mfd/qcom,wcd9335.txt | 130 + .../devicetree/bindings/sound/qcom,wcd9335.txt | 21 + drivers/mfd/Kconfig | 18 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 278 ++ drivers/mfd/wcd9335-irq.c | 172 + include/dt-bindings/mfd/wcd9335.h | 43 + include/linux/mfd/wcd9335/registers.h | 580 +++ include/linux/mfd/wcd9335/wcd9335.h | 43 + include/sound/soc-dai.h | 8 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh.c | 1010 ++++ sound/soc/codecs/wcd-clsh.h | 69 + sound/soc/codecs/wcd9335.c | 4981 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 68 + sound/soc/soc-core.c | 22 + 17 files changed, 7454 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/qcom,wcd9335.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 drivers/mfd/wcd9335-irq.c create mode 100644 include/dt-bindings/mfd/wcd9335.h create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.16.2

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[alsa-devel] [PATCH] sound: pci: cthw20k2: Replace mdelay() with msleep() and usleep_range()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

hw_pll_init(), hw_dac_stop(), hw_dac_start() and hw_adc_init() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ctxfi/cthw20k2.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index b866d6b2c923..3c966fafc754 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1316,12 +1316,12 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4); set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); pllctl = hw_read_20kx(hw, PLL_CTL); set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); for (i = 0; i < 1000; i++) { pllstat = hw_read_20kx(hw, PLL_STAT); @@ -1584,7 +1584,7 @@ static void hw_dac_stop(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data &= 0xFFFFFFFD; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(10); + usleep_range(10000, 11000); } static void hw_dac_start(struct hw *hw) @@ -1593,7 +1593,7 @@ static void hw_dac_start(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data |= 0x2; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); } static void hw_dac_reset(struct hw *hw) @@ -1864,11 +1864,11 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw_write_20kx(hw, GPIO_DATA, data); } - mdelay(10); + usleep_range(10000, 11000); /* Return the ADC to normal operation. */ data |= (0x1 << 15); hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); /* I2C write to register offset 0x0B to set ADC LRCLK polarity */ /* invert bit, interface format to I2S, word length to 24-bit, */ -- 2.17.0

2 1

[alsa-devel] [PATCH] sound: pci: cthw20k1: Replace mdelay() with msleep()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

hw_pll_init(), hw_reset_dac() and hw_card_init() are never called in atomic context. They calls mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ctxfi/cthw20k1.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 8e6eb9d7984b..6a051a1c3724 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1319,7 +1319,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) break; hw_write_20kx(hw, PLLCTL, pllctl); - mdelay(40); + msleep(40); } if (i >= 3) { dev_alert(hw->card->dev, "PLL initialization failed!!!\n"); @@ -1407,7 +1407,7 @@ static int hw_reset_dac(struct hw *hw) /* To be effective, need to reset the DAC twice. */ for (i = 0; i < 2; i++) { /* set gpio */ - mdelay(100); + msleep(100); gpioorg = (u16)hw_read_20kx(hw, GPIO); gpioorg &= 0xfffd; hw_write_20kx(hw, GPIO, gpioorg); @@ -2030,7 +2030,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, GIE, 0); /* Reset all SRC pending interrupts */ hw_write_20kx(hw, SRCIP, 0); - mdelay(30); + msleep(30); /* Detect the card ID and configure GPIO accordingly. */ switch (hw->model) { -- 2.17.0

2 1

[alsa-devel] [PATCH] sound: usb: quirks: Replace mdelay() with msleep() and usleep_range()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and snd_usb_ctl_msg_quirk() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep() and usleep_range(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/quirks.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acbeb52f6fd6..9eed650c54d4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1181,7 +1181,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, if (err < 0) return err; - mdelay(20); /* Delay needed after setting the interface */ + msleep(20); /* Delay needed after setting the interface */ /* Vendor mode switch cmd is required. */ if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) { @@ -1202,7 +1202,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, return err; } - mdelay(20); + msleep(20); } return 0; } @@ -1249,7 +1249,7 @@ void snd_usb_set_interface_quirk(struct usb_device *dev) switch (USB_ID_VENDOR(chip->usb_id)) { case 0x23ba: /* Playback Design */ case 0x0644: /* TEAC Corp. */ - mdelay(50); + msleep(50); break; } } @@ -1269,7 +1269,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* * "TEAC Corp." products need a 20ms delay after each @@ -1277,14 +1277,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x0644 && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ if (is_itf_usb_dsd_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, * otherwise requests like get/set frequency return as failed despite @@ -1294,7 +1294,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, chip->usb_id == USB_ID(0x046d, 0x0a46) || chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + usleep_range(1000, 2000); } /* -- 2.17.0

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[alsa-devel] BUG: soft lockup in snd_virmidi_output_trigger
by Dae R. Jeong 27 Jul '18

27 Jul '18

Reporting the crash: BUG: soft lockup in snd_virmidi_output_trigger This crash has been found in v4.18-rc3 using RaceFuzzer (a modified version of Syzkaller), which we describe more at the end of this report. Note that this bug is previously reported by Syzkaller a few month ago. (https://syzkaller.appspot.com/bug?id=642c927972318775e0ea1b4779ccd8624ce9e6…) Nonetheless, the crash still can be detected. We guess that the crash has not fixed yet. We report the crash log and the attached reproducer with a hope that they are helpful to diagnose the crash and to fix a bug. C Reproducer: https://kiwi.cs.purdue.edu/static/race-fuzzer/repro-dab082.c Kernel config: https://kiwi.cs.purdue.edu/static/race-fuzzer/config-dab082 (Please disregard all code related to "should_hypercall" in the C repro, as this is only for our debugging purposes using our own hypervisor.) Crash log: ================================================================== watchdog: BUG: soft lockup - CPU#1 stuck for 22s! [syz-executor0:24261] Modules linked in: irq event stamp: 6692 hardirqs last enabled at (6691): [<ffffffff84eb1f12>] __raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] hardirqs last enabled at (6691): [<ffffffff84eb1f12>] _raw_spin_unlock_irqrestore+0x62/0x90 kernel/locking/spinlock.c:184 hardirqs last disabled at (6692): [<ffffffff85000905>] interrupt_entry+0xb5/0xf0 arch/x86/entry/entry_64.S:625 softirqs last enabled at (1468): [<ffffffff852005ff>] __do_softirq+0x5ff/0x7f4 kernel/softirq.c:314 softirqs last disabled at (1455): [<ffffffff8130de56>] invoke_softirq kernel/softirq.c:368 [inline] softirqs last disabled at (1455): [<ffffffff8130de56>] irq_exit+0x126/0x130 kernel/softirq.c:408 CPU: 1 PID: 24261 Comm: syz-executor0 Not tainted 4.18.0-rc3 #1 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:783 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0x7d/0x90 kernel/locking/spinlock.c:184 Code: 0d c8 ce 16 7b 5b 41 5c 5d c3 e8 3e 0f 52 fc 48 c7 c7 68 b0 71 86 e8 62 4c 8d fc 48 83 3d 42 91 86 01 00 74 0e 48 89 df 57 9d <0f> 1f 44 00 00 eb cd 0f 0b 0f 0b 0f 1f 84 00 00 00 00 00 55 48 89 RSP: 0018:ffff8801b8e4f7e8 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: 0000000000000000 RBX: 0000000000000282 RCX: ffffffff84eb1f1e RDX: dffffc0000000000 RSI: dffffc0000000000 RDI: 0000000000000282 RBP: ffff8801b8e4f7f8 R08: ffffed002573a06a R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff88012b9d0348 R13: ffff8801e6417580 R14: ffff8801b8e4f868 R15: ffff8801e64175a8 FS: 00007f3447a49700(0000) GS:ffff8801f6d00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f3447a48db8 CR3: 00000001ee58d000 CR4: 00000000000006e0 Call Trace: spin_unlock_irqrestore include/linux/spinlock.h:365 [inline] snd_virmidi_output_trigger+0x37d/0x420 sound/core/seq/seq_virmidi.c:205 snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [inline] snd_rawmidi_kernel_write1+0x383/0x410 sound/core/rawmidi.c:1288 snd_rawmidi_write+0x249/0xa50 sound/core/rawmidi.c:1338 __vfs_write+0xf9/0x5d0 fs/read_write.c:485 vfs_write+0x195/0x380 fs/read_write.c:549 ksys_write+0xdb/0x1d0 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x43/0x50 fs/read_write.c:607 do_syscall_64+0x182/0x540 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x44b939 Code: 8d 6b fc ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 5b 6b fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f3447a48b48 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 000000000071bfa0 RCX: 000000000044b939 RDX: 000000002000010b RSI: 0000000020000100 RDI: 0000000000000013 RBP: 000000000000b7a8 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f3447a496d4 R13: 00000000ffffffff R14: 00000000006f7848 R15: 0000000000000000 = About RaceFuzzer RaceFuzzer is a customized version of Syzkaller, specifically tailored to find race condition bugs in the Linux kernel. While we leverage many different technique, the notable feature of RaceFuzzer is in leveraging a custom hypervisor (QEMU/KVM) to interleave the scheduling. In particular, we modified the hypervisor to intentionally stall a per-core execution, which is similar to supporting per-core breakpoint functionality. This allows RaceFuzzer to force the kernel to deterministically trigger racy condition (which may rarely happen in practice due to randomness in scheduling).

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Re: [alsa-devel] 答复: [PATCH 4/4] drm/amdgpu: Add audio component support
by Takashi Iwai 27 Jul '18

27 Jul '18

On Wed, 25 Jul 2018 07:38:37 +0200, Qu, Jim wrote: > > Jim: Just like Alex said, we want driver can get eld info when hotplug in new device. amdgpu driver is a bit difference from radeon driver, it is not a suitable place to call notify() function in *_audio_enable() , since they are not in the hotplug process context like them in radeon driver, but the mode setting. > > IMO, the right place to call notify() function is also in the amdgpu_connector_xxxx_detect() in amdgpu_connector.c. Hm, but by the modesetting, it actually enables / disables the audio as well, no? If so, the notifier is exactly for that purpose. The audio driver needs to know not only about the physical connection but whether the audio can be actually driven. That is, even if the monitor is connected, the audio won't come out if the mode is off. That is equivalent with the unplugged state for the audio driver. The i915 driver deals with the notifier just like the above, so the behavior is intentional. thanks, Takashi > With None-DC driver, the detect() of connectors will to be preformed in drm_helper_hpd_irq_event() when there is device hotplug. However, amdgpu_connector_xxxx_detect() maybe need to be modified to fit your change. > > Thanks > JimQu > ************************************************************************************************************ > --- > drivers/gpu/drm/Kconfig | 1 + > drivers/gpu/drm/amd/amdgpu/Makefile | 2 +- > drivers/gpu/drm/amd/amdgpu/amdgpu.h | 4 + > drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c | 97 +++++++++++++++++++++++ > drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h | 3 + > drivers/gpu/drm/amd/amdgpu/dce_v10_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v11_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v6_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v8_0.c | 6 ++ > 9 files changed, 130 insertions(+), 1 deletion(-) > create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > > diff --git a/drivers/gpu/drm/Kconfig b/drivers/gpu/drm/Kconfig > index 2c7112ddfed4..fbe7216c5c56 100644 > --- a/drivers/gpu/drm/Kconfig > +++ b/drivers/gpu/drm/Kconfig > @@ -193,6 +193,7 @@ config DRM_AMDGPU > select BACKLIGHT_LCD_SUPPORT > select INTERVAL_TREE > select CHASH > + select SND_HDA_COMPONENT if SND_HDA_CORE > help > Choose this option if you have a recent AMD Radeon graphics card. > > diff --git a/drivers/gpu/drm/amd/amdgpu/Makefile b/drivers/gpu/drm/amd/amdgpu/Makefile > index bfd332c95b61..9c26facddb17 100644 > --- a/drivers/gpu/drm/amd/amdgpu/Makefile > +++ b/drivers/gpu/drm/amd/amdgpu/Makefile > @@ -52,7 +52,7 @@ amdgpu-y += amdgpu_device.o amdgpu_kms.o \ > amdgpu_ucode.o amdgpu_bo_list.o amdgpu_ctx.o amdgpu_sync.o \ > amdgpu_gtt_mgr.o amdgpu_vram_mgr.o amdgpu_virt.o amdgpu_atomfirmware.o \ > amdgpu_queue_mgr.o amdgpu_vf_error.o amdgpu_sched.o amdgpu_debugfs.o \ > - amdgpu_ids.o > + amdgpu_ids.o amdgpu_audio.o > > # add asic specific block > amdgpu-$(CONFIG_DRM_AMDGPU_CIK)+= cik.o cik_ih.o kv_smc.o kv_dpm.o \ > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu.h b/drivers/gpu/drm/amd/amdgpu/amdgpu.h > index a59c07590cee..203d2584c989 100644 > --- a/drivers/gpu/drm/amd/amdgpu/amdgpu.h > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu.h > @@ -1957,5 +1957,9 @@ int amdgpu_dm_display_resume(struct amdgpu_device *adev ); > static inline int amdgpu_dm_display_resume(struct amdgpu_device *adev) { return 0; } > #endif > > +int amdgpu_audio_component_init(struct amdgpu_device *adev); > +void amdgpu_audio_component_fini(struct amdgpu_device *adev); > +void amdgpu_audio_eld_notify(struct amdgpu_device *adev, int pin); > + > #include "amdgpu_object.h" > #endif > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > new file mode 100644 > index 000000000000..39256e2f84b3 > --- /dev/null > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > @@ -0,0 +1,97 @@ > +// SPDX-License-Identifier: MIT > + > +#include <linux/component.h> > +#include "amdgpu.h" > + > +static int amdgpu_audio_component_get_eld(struct device *kdev, int port, > + int pipe, bool *enabled, > + unsigned char *buf, int max_bytes) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct drm_encoder *encoder; > + struct amdgpu_encoder *amdgpu_encoder; > + struct amdgpu_encoder_atom_dig *dig; > + struct drm_connector *connector; > + int ret = 0; > + > + *enabled = 0; > + list_for_each_entry(encoder, &dev->mode_config.encoder_list, head) { > + amdgpu_encoder = to_amdgpu_encoder(encoder); > + dig = amdgpu_encoder->enc_priv; > + if (!dig || !dig->afmt || !dig->afmt->enabled) > + continue; > + if (!dig->afmt->pin || dig->afmt->pin->id != port) > + continue; > + connector = amdgpu_get_connector_for_encoder(encoder); > + *enabled = !!connector; > + if (connector) { > + ret = drm_eld_size(connector->eld); > + memcpy(buf, connector->eld, min(max_bytes, ret)); > + break; > + } > + } > + > + return ret; > +} > + > +static const struct drm_audio_component_ops amdgpu_audio_component_ops = { > + .get_eld = amdgpu_audio_component_get_eld, > +}; > + > +static int amdgpu_audio_component_bind(struct device *kdev, > + struct device *hda_kdev, void *data) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct amdgpu_device *adev = dev->dev_private; > + struct drm_audio_component *acomp = data; > + > + acomp->ops = &amdgpu_audio_component_ops; > + acomp->dev = kdev; > + adev->mode_info.audio.component = acomp; > + return 0; > +} > + > +static void amdgpu_audio_component_unbind(struct device *kdev, > + struct device *hda_kdev, void *data) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct amdgpu_device *adev = dev->dev_private; > + struct drm_audio_component *acomp = data; > + > + acomp->ops = NULL; > + acomp->dev = NULL; > + adev->mode_info.audio.component = NULL; > +} > + > +static const struct component_ops amdgpu_audio_component_bind_ops = { > + .bind = amdgpu_audio_component_bind, > + .unbind = amdgpu_audio_component_unbind, > +}; > + > +int amdgpu_audio_component_init(struct amdgpu_device *adev) > +{ > + int err; > + > + err = component_add(adev->dev, &amdgpu_audio_component_bind_ops); > + if (err < 0) > + return err; > + adev->mode_info.audio.component_registered = true; > + return 0; > +} > + > +void amdgpu_audio_component_fini(struct amdgpu_device *adev) > +{ > + if (adev->mode_info.audio.component_registered) { > + component_del(adev->dev, &amdgpu_audio_component_bind_ops); > + adev->mode_info.audio.component_registered = false; > + } > +} > + > +void amdgpu_audio_eld_notify(struct amdgpu_device *adev, int pin) > +{ > + struct drm_audio_component *acomp = adev->mode_info.audio.component; > + > + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > + pin, -1); > +} > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h b/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > index b9e9e8b02fb7..7cb1aa97e522 100644 > --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > @@ -39,6 +39,7 @@ > #include <drm/drm_fb_helper.h> > #include <drm/drm_plane_helper.h> > #include <drm/drm_fb_helper.h> > +#include <drm/drm_audio_component.h> > #include <linux/i2c.h> > #include <linux/i2c-algo-bit.h> > #include <linux/hrtimer.h> > @@ -260,6 +261,8 @@ struct amdgpu_audio { > bool enabled; > struct amdgpu_audio_pin pin[AMDGPU_MAX_AFMT_BLOCKS]; > int num_pins; > + bool component_registered; > + struct drm_audio_component *component; > }; > > struct amdgpu_display_funcs { > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > index ada241bfeee9..c8471a59930f 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > @@ -1392,6 +1392,8 @@ static void dce_v10_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[] = > @@ -1430,6 +1432,8 @@ static int dce_v10_0_audio_init(struct amdgpu_device *adev) > dce_v10_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1443,6 +1447,8 @@ static void dce_v10_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v10_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > index a5b96eac3033..49edb74725a7 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > @@ -1418,6 +1418,8 @@ static void dce_v11_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[] = > @@ -1472,6 +1474,8 @@ static int dce_v11_0_audio_init(struct amdgpu_device *adev) > dce_v11_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1485,6 +1489,8 @@ static void dce_v11_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v11_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > index 394cc1e8fe20..2b7468eaacca 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > @@ -1297,6 +1297,8 @@ static void dce_v6_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[7] = > @@ -1343,6 +1345,8 @@ static int dce_v6_0_audio_init(struct amdgpu_device *adev) > dce_v6_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1356,6 +1360,8 @@ static void dce_v6_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v6_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > index c9b9ab8f1b05..c2370ea1f64f 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > @@ -1348,6 +1348,8 @@ static void dce_v8_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[7] = > @@ -1395,6 +1397,8 @@ static int dce_v8_0_audio_init(struct amdgpu_device *adev) > dce_v8_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1408,6 +1412,8 @@ static void dce_v8_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v8_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > -- > 2.18.0 > > _______________________________________________ > amd-gfx mailing list > amd-gfx(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/amd-gfx

2 3

[alsa-devel] [PATCH v7 00/10] soundwire: Add multi link support
by Shreyas NC 27 Jul '18

27 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Fix to avoid bus lock acquired twice - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch Additionally, as suggested by Vinod, the generic fixes are arranged at the top of the series followed by the Documentation patch for multi link support and then the multi link patches. changes in v7: - Fix the git bisect issue pointed by Pierre. Now, the initialization of the m_rt_count and its modification/usage is split into 2 patches. While the declaration and initialization is in one patch, modification of the refcount and usage is in the multilink bankswitch patch. - Re-arrange the patches as suggested by Vinod changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes Sanyog Kale (3): soundwire: Fix acquiring bus lock twice during master release Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (6): soundwire: Fix duplicate stream state assignment soundwire: Fix incorrect exit after configuring stream soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.7.4

2 12

[alsa-devel] [PATCH v4 0/8] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 26 Jul '18

26 Jul '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v4 is based on Takashi's topic/drm_audio_component merged on top on Marks' for-next branch - the merge is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (6): ASoC: Intel: Skylake: fix widget handling ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 469 ++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/intel/Kconfig | 18 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 133 ++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 174 ++++++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 +++- sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 100 ++++- sound/soc/intel/skylake/skl.h | 12 +- 18 files changed, 1093 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.14.1

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[alsa-devel] [PATCH] ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally
by Pierre-Louis Bossart 26 Jul '18

26 Jul '18

Make this helper inline function available for all platforms. This helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not defined. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/memalloc.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..c669900e6cbe 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -67,6 +67,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -90,14 +98,6 @@ struct snd_sg_buf { struct device *dev; }; -/* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - /* * return the physical address at the corresponding offset */ -- 2.14.1

4 7

[alsa-devel] [PATCH 0/6] ASoC: PCM format type conversion fixes
by Takashi Iwai 26 Jul '18

26 Jul '18

Hi, the PCM format is a fixed type with __bitwise, and it needs cast or other trick to deal with integer. This is a series of such fixes for ASoC. Takashi === Takashi Iwai (6): ALSA: pcm: Add snd_mask_set_format() helper for standard usages ASoC: doc: Replace open code with params_set_format() ASoC: intel: Fix snd_pcm_format_t handling ASoC: fsl: Use snd_mask_set_format() ASoC: pcm186x: Declare PCM format with snd_pcm_format_t ASoC: dmaengine: Use standard pcm_format_to_bits() macro Documentation/sound/soc/dpcm.rst | 4 +--- include/sound/pcm_params.h | 10 ++++++++-- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 14 files changed, 23 insertions(+), 21 deletions(-) -- 2.18.0

2 14

[alsa-devel] [PATCH 0/3] ASoC: meson-axg: fifo and card fixup
by Jerome Brunet 26 Jul '18

26 Jul '18

The purpose of this patchset is to solve the problems which have been raised since the initial axg sound support has been accepted. Jerome Brunet (3): ASoC: meson: use IRQ_RETVAL in the fifo irq handler ASoC: meson: update axg sound card bindings ASoC: meson: align axg card driver with DT bindings documentation .../bindings/sound/amlogic,axg-sound-card.txt | 68 +++++++++---------- sound/soc/meson/axg-card.c | 12 ++-- sound/soc/meson/axg-fifo.c | 2 +- 3 files changed, 41 insertions(+), 41 deletions(-) -- 2.17.1

2 6

[alsa-devel] Applied "ASoC: rsnd: Document R-Car M3-N support" to the asoc tree
by Mark Brown 26 Jul '18

26 Jul '18

The patch ASoC: rsnd: Document R-Car M3-N support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 150a6dc8fc764c1912fa2a46e18d1505dfd3536f Mon Sep 17 00:00:00 2001 From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Date: Thu, 26 Jul 2018 05:40:15 +0900 Subject: [PATCH] ASoC: rsnd: Document R-Car M3-N support Document support for the sound modules in the Renesas M3-N (r8a77965) SoC. No driver update is needed. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Simon Horman <horms+renesas(a)verge.net.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790f630f..9e764270c36b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.18.0

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[alsa-devel] [PATCH] ASoC: zte: Fix incorrect PCM format bit usages
by Takashi Iwai 26 Jul '18

26 Jul '18

zx-tdm driver sets the DAI driver definitions with the format bits wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*. This patch corrects the definitions. Spotted by a sparse warning: sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer Fixes: 870e0ddc4345 ("ASoC: zx-tdm: add zte's tdm controller driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/zte/zx-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { -- 2.18.0

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[alsa-devel] [PATCH] ASoC: dmaengine: Fix missing __user prefix in copy_user callback
by Takashi Iwai 26 Jul '18

26 Jul '18

It seems that __user prefix was forgotten to be added to dmaengine_copy_user callback while we refactored the user-copy PCM core. This patch adds the missing prefix, remove the superfluous cast, and add the needed cast (__force is needed for downgrading from user pointer to kernel pointer), too. Spotted by a sparse warning like: sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces)) Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..61027dd0ef91 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -343,7 +343,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -359,18 +359,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; -- 2.18.0

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[alsa-devel] [PATCH] ASoC: dapm: Use int for format bit position
by Takashi Iwai 26 Jul '18

26 Jul '18

fmt in snd_soc_dai_link_event() contains the format bit position, not the format bit itself. Hence it can be a simple integer instead of the explicit u64. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 229c12349803..a9d0ca4f92e1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3662,7 +3662,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || -- 2.18.0

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