May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 19 May '16

19 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 4 of the patch: * minor correction in the binding example ("interrupt names") * re-added index note to the "mpu" property.

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[alsa-devel] [PATCH 1/1] ASoC: core: Do not print an actual error when deferring probe
by Alexander Stein 19 May '16

19 May '16

In case there is no DAI (yet), do not print an error, this might happen a lot of times. Print a notice instead. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..352f7c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1001,7 +1001,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, cpu_dai_component.dai_name = dai_link->cpu_dai_name; rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { - dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); goto _err_defer; } @@ -1013,7 +1013,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); goto _err_defer; } @@ -1042,7 +1042,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->platform = platform; } if (!rtd->platform) { - dev_err(card->dev, "ASoC: platform %s not registered\n", + dev_notice(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } -- 2.7.3

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[alsa-devel] [PATCH] ASoC: ak4642: Enable cache usage to fix crashes on resume
by Mark Brown 19 May '16

19 May '16

The ak4642 driver is using a regmap cache sync to restore the configuration of the chip on resume but (as Peter observed) does not actually define a register cache which means that the resume is never going to work and we trigger asserts in regmap. Fix this by enabling caching. Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/ak4642.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1ee8506c06c7..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { -- 2.8.1

3 4

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Add channel constraints for refcap
by Yong Zhi 19 May '16

19 May '16

Add constraint for ref DMIC to match with the topology firmware config. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 7cc7251..463f0e5 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -339,8 +339,22 @@ static struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int skylake_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); -- 1.9.1

2 2

[alsa-devel] [PATCH V2 1/2] dt: binding: ASOC: add dt-binding doc for pcm5102a
by kernel＠martin.sperl.org 18 May '16

18 May '16

From: Martin Sperl <kernel(a)martin.sperl.org> Add dt-binding documentation for the TI pcm5102a DAC. Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> --- Documentation/devicetree/bindings/sound/pcm5102a.txt | 13 +++++++++++++ 1 file changed, 13 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 0000000..87d72be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODEC + +This device is a i2s only device. + +Required properties: + +- compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; -- 2.1.4

2 2

[alsa-devel] [PATCH v2 1/1] ASoC: core: Do not print an actual error when deferring probe
by Alexander Stein 18 May '16

18 May '16

In case there is no DAI (yet), do not print an error, this might happen a lot of times. Print a notice instead. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- Changes in v2: * Add addional change in soc_bind_aux_dev() sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..ce5f835 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1001,7 +1001,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, cpu_dai_component.dai_name = dai_link->cpu_dai_name; rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { - dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); goto _err_defer; } @@ -1013,7 +1013,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); goto _err_defer; } @@ -1042,7 +1042,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->platform = platform; } if (!rtd->platform) { - dev_err(card->dev, "ASoC: platform %s not registered\n", + dev_notice(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1698,7 +1698,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) return 0; err_defer: - dev_err(card->dev, "ASoC: %s not registered\n", name); + dev_notice(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } -- 2.7.3

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[alsa-devel] [PATCH] ASoC: add api to retrieve dmic array info from coreboot nhlt
by Yong Zhi 18 May '16

18 May '16

skylake can be configured with either both 2 and 4 channel DMIC array, or 2 channel DMIC array only, this patch provides an API to retrieve the DMIC info from nhlt. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/skylake/skl-nhlt.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.h | 22 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 10 +++++++++- sound/soc/intel/skylake/skl.h | 6 ++++++ 4 files changed, 74 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 7d73648..5448a48 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -149,6 +149,43 @@ struct nhlt_specific_cfg return NULL; } +int skl_get_ep_dmic(struct skl *skl) +{ + struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_endpoint *epnt; + struct nhlt_dmic_array_config *cfg; + struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct device *dev = bus->dev; + unsigned int dmic_geo = 0; + u8 j; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + + for (j = 0; j < nhlt->endpoint_count; j++) { + if (epnt->linktype == NHLT_LINK_DMIC) { + cfg = (struct nhlt_dmic_array_config *) + (epnt->config.caps); + switch (cfg->array_type) { + case NHLT_MIC_ARRAY_2CH_SMALL: + case NHLT_MIC_ARRAY_2CH_BIG: + dmic_geo |= MIC_ARRAY_2CH; + break; + case NHLT_MIC_ARRAY_4CH_1ST_GEOM: + case NHLT_MIC_ARRAY_4CH_L_SHAPED: + case NHLT_MIC_ARRAY_4CH_2ND_GEOM: + dmic_geo |= MIC_ARRAY_4CH; + break; + default: + dev_warn(dev, "undefined DMIC array_type 0x%0x\n", + cfg->array_type); + } + } + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return dmic_geo; +} + static void skl_nhlt_trim_space(struct skl *skl) { char *s = skl->tplg_name; diff --git a/sound/soc/intel/skylake/skl-nhlt.h b/sound/soc/intel/skylake/skl-nhlt.h index 3769f9f..8da502b 100644 --- a/sound/soc/intel/skylake/skl-nhlt.h +++ b/sound/soc/intel/skylake/skl-nhlt.h @@ -103,4 +103,26 @@ struct nhlt_resource_desc { u64 length; } __packed; +#define MIC_ARRAY_2CH 2 +#define MIC_ARRAY_4CH 4 + +struct nhlt_tdm_config { + uint8_t virtual_slot; + uint8_t config_type; +} __packed; + +struct nhlt_dmic_array_config { + struct nhlt_tdm_config tdm_config; + uint8_t array_type; +} __packed; + +enum { + NHLT_MIC_ARRAY_2CH_SMALL = 0xa, + NHLT_MIC_ARRAY_2CH_BIG = 0xb, + NHLT_MIC_ARRAY_4CH_1ST_GEOM = 0xc, + NHLT_MIC_ARRAY_4CH_L_SHAPED = 0xd, + NHLT_MIC_ARRAY_4CH_2ND_GEOM = 0xe, + NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf, +}; + #endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 06d8c26..a75b05f 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -35,6 +35,8 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +static struct skl_dmic_info skl_dmic_data; + /* * initialize the PCI registers */ @@ -397,6 +399,10 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) platform_device_put(pdev); return -EIO; } + + if (mach->pdata) + dev_set_drvdata(&pdev->dev, mach->pdata); + skl->i2s_dev = pdev; return 0; @@ -666,6 +672,8 @@ static int skl_probe(struct pci_dev *pci, pci_set_drvdata(skl->pci, ebus); + skl_dmic_data.dmic_num = skl_get_ep_dmic(skl); + /* check if dsp is there */ if (ebus->ppcap) { err = skl_machine_device_register(skl, @@ -789,7 +797,7 @@ static struct sst_acpi_mach sst_skl_devdata[] = { { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, { "MX98357A", "skl_nau88l25_max98357a_i2s", "intel/dsp_fw_release.bin", - NULL, NULL, NULL }, + NULL, NULL, &skl_dmic_data}, {} }; diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 4b4b387..889ec30 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -90,6 +90,11 @@ struct skl_dma_params { u8 stream_tag; }; +/* to pass dmic data */ +struct skl_dmic_info { + u32 dmic_num; +}; + struct skl_dsp_ops { int id; struct skl_dsp_loader_ops (*loader_ops)(void); @@ -108,6 +113,7 @@ void skl_nhlt_free(struct nhlt_acpi_table *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); +int skl_get_ep_dmic(struct skl *skl); int skl_nhlt_update_topology_bin(struct skl *skl); int skl_init_dsp(struct skl *skl); int skl_free_dsp(struct skl *skl); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
by Takashi Sakamoto 18 May '16

18 May '16

Commit a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") adds new members to tracepoint events of this module, to represent context information. One of the members is bool type and this causes sparse warnings. 16:1: warning: expression using sizeof bool 60:1: warning: expression using sizeof bool 16:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) 60:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) This commit suppresses the warnings, by changing type of the member to 'unsigned int'. Additionally, this commit applies '!!' idiom to get 0/1 from 'in_interrupt()'. Fixes: a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 1622579..9c04faf 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -26,7 +26,7 @@ TRACE_EVENT(in_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -39,7 +39,7 @@ TRACE_EVENT(in_packet, __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( @@ -70,7 +70,7 @@ TRACE_EVENT(out_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -83,7 +83,7 @@ TRACE_EVENT(out_packet, __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: twl6040: Disconnect AUX output pads on digital mute
by Peter Ujfalusi 18 May '16

18 May '16

Disconnect also the path to AUXL from the HF path during digital_mute to avoid pop noise leakage to Line-out pads. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index a7c50e54f3e0..8015db974157 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -134,6 +134,7 @@ #define TWL6040_HFDACENA (1 << 0) #define TWL6040_HFPGAENA (1 << 1) #define TWL6040_HFDRVENA (1 << 4) +#define TWL6040_HFSWENA (1 << 6) /* VIBCTLL/R (0x18/0x1A) fields */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7a744b581cc1..1f7081043566 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -987,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i if (mute) { /* Power down drivers and DACs */ hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); } twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); -- 2.8.2

1 0

[alsa-devel] [PATCH v7 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 18 May '16

18 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v7 Changes from v6: - DT bindings acknowledged by Rob Herring Changes from v5: - Fix code style, e.g. change "if" statements to "switch" statements - Fix SPDIF output enable register field - Set BUFFER_PAIR_ENABLE for both mono, stereo and TDM modes - Reflect PCM bit formats for SPDIF, SSP and TDM as supported by hardware Changes from v4: - Fix power suspend function and add power resume function - Move clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Remove surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fix print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2610 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.9.1

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