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

[alsa-devel] [PATCH] ALSA: ctxfi: use enum type CT_SUM_CTL where appropriate
by Stefan Agner 18 Jun '18

18 Jun '18

Currently a variable of type enum CT_AMIXER_CTL is used for enum CT_SUM_CTL values. This leads to warnings when using clang: sound/pci/ctxfi/ctmixer.c:945:32: warning: implicit conversion from enumeration type 'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL' [-Wenum-conversion] for (i = AMIXER_MASTER_F, j = SUM_IN_F; ~ ^~~~~~~~ sound/pci/ctxfi/ctmixer.c:975:29: warning: implicit conversion from enumeration type 'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL' [-Wenum-conversion] for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { ~ ^~~~~~~~ Introduce enum CT_SUM_CTL k and it instead. Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/pci/ctxfi/ctmixer.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index db710d0a609f..4777d50fbbf8 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -938,17 +938,18 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) struct sum *sum; struct amixer *amix_d, *amix_s; enum CT_AMIXER_CTL i, j; + enum CT_SUM_CTL k; /* Build topology from destination to source */ /* Set up Master mixer */ - for (i = AMIXER_MASTER_F, j = SUM_IN_F; - i <= AMIXER_MASTER_S; i++, j++) { + for (i = AMIXER_MASTER_F, k = SUM_IN_F; + i <= AMIXER_MASTER_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL); } @@ -972,12 +973,12 @@ static int ct_mixer_topology_build(struct ct_mixer *mixer) amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL); /* Set up PCM-in mixer */ - for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) { + for (i = AMIXER_PCM_F, k = SUM_IN_F; i <= AMIXER_PCM_S; i++, k++) { amix_d = mixer->amixers[i*CHN_NUM]; - sum = mixer->sums[j*CHN_NUM]; + sum = mixer->sums[k*CHN_NUM]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); amix_d = mixer->amixers[i*CHN_NUM+1]; - sum = mixer->sums[j*CHN_NUM+1]; + sum = mixer->sums[k*CHN_NUM+1]; amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: ice1724: remove unused array
by Stefan Agner 18 Jun '18

18 Jun '18

This fixes a warning seen with clang: sound/pci/ice1712/prodigy_hifi.c:321:28: warning: variable 'wm_vol' is not needed and will not be emitted [-Wunneeded-internal-declaration] static const unsigned char wm_vol[256] = { ^ Signed-off-by: Stefan Agner <stefan(a)agner.ch> --- sound/pci/ice1712/prodigy_hifi.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index d7366ade5a25..c97b5528e4b8 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -314,26 +314,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] = { /* --------------- */ -/* - * Logarithmic volume values for WM87*6 - * Computed as 20 * Log10(255 / x) - */ -static const unsigned char wm_vol[256] = { - 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23, - 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17, - 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13, - 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11, - 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8, - 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6, - 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3, - 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, - 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, - 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0 -}; - -#define WM_VOL_MAX (sizeof(wm_vol) - 1) +#define WM_VOL_MAX 255 #define WM_VOL_MUTE 0x8000 -- 2.17.1

2 1

[alsa-devel] [PATCH v3 0/6] soundwire: Add multi link support
by Shreyas NC 18 Jun '18

18 Jun '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: - 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 changes in v3: - Added comments to explain the trigger for multi link bank switch - Added comments in the intel_post_bank_switch ops to explain how synchronous trigger of bank switch is handled. Sanyog Kale (2): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (3): soundwire: Initialize completion for defer messages 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 | 29 +- drivers/soundwire/bus.c | 1 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 456 ++++++++++++++++++-------- include/linux/soundwire/sdw.h | 8 +- 6 files changed, 432 insertions(+), 131 deletions(-) -- 2.7.4

3 9

[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for the Nuvison/TMax TM800W560 tablet
by Hans de Goede 18 Jun '18

18 Jun '18

Add a quirk for the Nuvison/TMax TM800W560 tablet, this tablet uses IN1 for the internal mic rather then the default IN3 and it uses JD2 rather then JD1 for its not-inverted jack-detect switch. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d675d550f6c1..27a05778584f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -549,6 +549,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_DIFF_MIC | BYT_RT5640_MCLK_EN), }, + { /* Nuvison/TMax TM800W560 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TMAX"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TM800W560L"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Pipo W4 */ .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), -- 2.17.1

3 6

[alsa-devel] [PATCH 0/3] sound/usb/line6: fix some styling issues
by Roman Sommer 18 Jun '18

18 Jun '18

CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> CC: "Hans P. Möller Ebner" <hmoller(a)uc.cl> CC: Kees Cook <keescook(a)chromium.org> CC: Kate Stewart <kstewart(a)linuxfoundation.org> CC: Thomas Gleixner <tglx(a)linutronix.de> CC: "Fabián Inostroza" <soulsonceonfire(a)gmail.com> CC: Bhumika Goyal <bhumirks(a)gmail.com> CC: Philippe Ombredanne <pombredanne(a)nexb.com> CC: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> CC: alsa-devel(a)alsa-project.org CC: linux-kernel(a)vger.kernel.org This patchseries addresses some styling issues in sound/usb/line6 found by checkpatch. Roman Sommer (3): sound/line6: checkpatch: fix block comment style sound/line6: unsigned -> unsigned int sound/line6: add newline after declarations sound/usb/line6/capture.c | 20 ++++----- sound/usb/line6/driver.c | 110 ++++++++++++++++++++++----------------------- sound/usb/line6/driver.h | 33 +++++++------- sound/usb/line6/midi.c | 24 +++++----- sound/usb/line6/midibuf.c | 6 +-- sound/usb/line6/pcm.c | 26 +++++------ sound/usb/line6/pcm.h | 74 +++++++++++++++--------------- sound/usb/line6/playback.c | 36 ++++++++------- sound/usb/line6/pod.c | 60 ++++++++++++------------- sound/usb/line6/podhd.c | 8 ++-- sound/usb/line6/toneport.c | 30 ++++++------- sound/usb/line6/variax.c | 40 ++++++++--------- 12 files changed, 235 insertions(+), 232 deletions(-) -- 2.7.4

4 8

[alsa-devel] [PATCH] ASoC: DAPM: Fix potential DAI widget pointer deref when linking DAIs
by Liam Girdwood 18 Jun '18

18 Jun '18

Sometime a component or topology may configure a DAI widget with no private data leading to a dev_dbg() dereferencne of this data. Fix this to check for non NULL private data and let users know if widget is missing DAI. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 36a39ba30226..8ede773b1db8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4073,6 +4073,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) continue; } + /* let users know there is no DAI to link */ + if (!dai_w->priv) { + dev_dbg(card->dev, "dai widget %s has no DAI\n", + dai_w->name); + continue; + } + dai = dai_w->priv; /* ...find all widgets with the same stream and link them */ -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ASoC: topology: Give more data to clients via callbacks
by Liam Girdwood 18 Jun '18

18 Jun '18

Give topology clients more access to the topology data by passing index, pcm, link_config and dai_driver to clients. This allows clients to fully instantiate and track topology objects. The SOF driver is the first user of these new APIs and needs them to build component topology driver and FW objects. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1: - Fixed build error on skylake. --- include/sound/soc-topology.h | 23 +++++++++++-------- sound/soc/intel/skylake/skl-pcm.c | 7 +++--- sound/soc/intel/skylake/skl-topology.c | 5 +++-- sound/soc/intel/skylake/skl-topology.h | 5 +++-- sound/soc/soc-topology.c | 31 +++++++++++++++----------- 5 files changed, 42 insertions(+), 29 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f552c3f56368..e1f265e21ee1 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -30,6 +30,8 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; +struct snd_soc_dai_driver; +struct snd_soc_dai; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -109,35 +111,38 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, + int (*control_load)(struct snd_soc_component *, int index, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, + int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, + int (*widget_ready)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + int (*dai_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); + int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, - struct snd_soc_dai_link *link); + int (*link_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, + int (*vendor_load)(struct snd_soc_component *, int index, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -146,7 +151,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, + int (*manifest)(struct snd_soc_component *, int index, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index afa86b9e4dcf..1f4dd08d36c5 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1017,10 +1017,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 2c5129782959..dd9605ba1543 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3020,7 +3020,7 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { @@ -3127,6 +3127,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { @@ -3615,7 +3616,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 6d7e0569695f..af198ea0379e 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -512,8 +512,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, struct snd_soc_dapm_widget *w); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 3fd5d9c867b9..a30bb94fc800 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -259,7 +259,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -291,7 +291,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -302,27 +303,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -643,7 +647,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -1702,7 +1707,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, dai_drv->compress_new = snd_soc_new_compress; /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1772,7 +1777,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2080,7 +2085,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2202,7 +2207,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2311,7 +2316,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); -- 2.17.1

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: SOF driver dependencies
by Liam Girdwood 18 Jun '18

18 Jun '18

This series is a set of ASoC core and topology updates needed by the SOF driver. The SOF driver uses topology extensively to define most of the characteristsics of an audio card with the exception of some immutable items like hardware backend DAI links. The SOF driver also allows for reuse of existing upstream machine drivers by re-writting some aspects of the DAI link and DAPM configuration with data from topology. i.e. existing machine drivers can be reused with different topologies. V2: Rebased on top of today for-next, V1 was probably missing 1/6 meaning the others would not apply. Liam Girdwood (6): ASoC: core: Allow topology to override machine driver FE DAI link config. ASoC: core: Add name prefix for machines with topology rewrites ASoC: topology: Give more data to clients via callbacks ASoC: topology: Add callback for DAPM route load/unload ASoC: topology: Check widget kcontrols before deref. ASoC: topology: Add support for compressed PCMs include/sound/soc-topology.h | 30 ++++++++---- include/sound/soc.h | 12 +++++ sound/soc/intel/skylake/skl-pcm.c | 7 +-- sound/soc/intel/skylake/skl-topology.c | 5 +- sound/soc/intel/skylake/skl-topology.h | 20 ++------ sound/soc/soc-core.c | 87 ++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 +++++ sound/soc/soc-topology.c | 51 ++++++++++++++------ 8 files changed, 177 insertions(+), 47 deletions(-) -- 2.14.1

2 16

[alsa-devel] [PATCH 0/8 v2] ASoC: sh: convert SPDX identifiers
by Kuninori Morimoto 18 Jun '18

18 Jun '18

Hi Mark These patches are v2 of convert license to SPDX identifiers v2 uses C++ style comment-out Kuninori Morimoto (8): ASoC: fsi: convert to SPDX identifiers ASoC: hac: convert to SPDX identifiers ASoC: ssi: convert to SPDX identifiers ASoC: siu: convert to SPDX identifiers ASoC: rsnd: convert to SPDX identifiers ASoC: migor: convert to SPDX identifiers ASoC: dma-sh7760: convert to SPDX identifiers ASoC: sh7760-ac97: convert to SPDX identifiers sound/soc/sh/dma-sh7760.c | 26 ++++++++++++-------------- sound/soc/sh/fsi.c | 22 +++++++++------------- sound/soc/sh/hac.c | 20 +++++++++----------- sound/soc/sh/migor.c | 14 +++++--------- sound/soc/sh/rcar/Makefile | 1 + sound/soc/sh/rcar/adg.c | 15 ++++++--------- sound/soc/sh/rcar/cmd.c | 17 +++++++---------- sound/soc/sh/rcar/core.c | 24 ++++++++++-------------- sound/soc/sh/rcar/ctu.c | 15 ++++++--------- sound/soc/sh/rcar/dma.c | 17 +++++++---------- sound/soc/sh/rcar/dvc.c | 16 ++++++---------- sound/soc/sh/rcar/gen.c | 16 ++++++---------- sound/soc/sh/rcar/mix.c | 14 +++++--------- sound/soc/sh/rcar/rsnd.h | 17 +++++++---------- sound/soc/sh/rcar/src.c | 16 ++++++---------- sound/soc/sh/rcar/ssi.c | 22 +++++++++------------- sound/soc/sh/rcar/ssiu.c | 15 ++++++--------- sound/soc/sh/sh7760-ac97.c | 14 ++++++-------- sound/soc/sh/siu.h | 26 ++++++-------------------- sound/soc/sh/siu_dai.c | 26 ++++++-------------------- sound/soc/sh/siu_pcm.c | 27 +++++++-------------------- sound/soc/sh/ssi.c | 21 +++++++++------------ 22 files changed, 151 insertions(+), 250 deletions(-) -- 1.9.1

2 17

[alsa-devel] [PATCH 0/3] ASoC: rsnd: fixup SWSP bit for SSI parent
by Kuninori Morimoto 18 Jun '18

18 Jun '18

Hi Mark Renesas BSP team Yokoyama-san noticed that SSI need to care about SWSP bit if SSI is working as parent, otherwise, it might have noise when recording. These patches solve this issue. Kuninori Morimoto (3): ASoC: rsnd: add rsnd_daidrv_get() ASoC: rsnd: has .symmetric_rates if SSIs are sharing WS pin ASoC: rsnd: SSI parent cares SWSP bit sound/soc/sh/rcar/core.c | 17 ++++++++++++++++- sound/soc/sh/rcar/ssi.c | 37 ++++++++++++++++++++++++------------- 2 files changed, 40 insertions(+), 14 deletions(-) -- 1.9.1

2 6