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[alsa-devel] [PATCH 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 19 Apr '13

19 Apr '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..88fc02c 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,9 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

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Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Bjorn Helgaas 18 Apr '13

18 Apr '13

On Tue, Apr 16, 2013 at 12:27 PM, David Howells <dhowells(a)redhat.com> wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- For the drivers/pci part: Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> I assume you'll merge this via something other than my tree. > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > > diff --git a/drivers/char/ipmi/ipmi_msghandler.c b/drivers/char/ipmi/ipmi_msghandler.c > index 1420bbb..4d439d2 100644 > --- a/drivers/char/ipmi/ipmi_msghandler.c > +++ b/drivers/char/ipmi/ipmi_msghandler.c > @@ -4541,7 +4541,7 @@ static void __exit cleanup_ipmi(void) > del_timer_sync(&ipmi_timer); > > #ifdef CONFIG_PROC_FS > - remove_proc_entry(proc_ipmi_root->name, NULL); > + proc_remove(proc_ipmi_root); > #endif /* CONFIG_PROC_FS */ > > driver_unregister(&ipmidriver.driver); > diff --git a/drivers/misc/sgi-gru/gruprocfs.c b/drivers/misc/sgi-gru/gruprocfs.c > index 950dbe9..797d796 100644 > --- a/drivers/misc/sgi-gru/gruprocfs.c > +++ b/drivers/misc/sgi-gru/gruprocfs.c > @@ -355,7 +355,7 @@ static void delete_proc_files(void) > for (p = proc_files; p->name; p++) > if (p->entry) > remove_proc_entry(p->name, proc_gru); > - remove_proc_entry("gru", proc_gru->parent); > + proc_remove(proc_gru); > } > } > > diff --git a/drivers/of/base.c b/drivers/of/base.c > index 321d3ef..9c70436 100644 > --- a/drivers/of/base.c > +++ b/drivers/of/base.c > @@ -1452,16 +1452,7 @@ int of_attach_node(struct device_node *np) > #ifdef CONFIG_PROC_DEVICETREE > static void of_remove_proc_dt_entry(struct device_node *dn) > { > - struct device_node *parent = dn->parent; > - struct property *prop = dn->properties; > - > - while (prop) { > - remove_proc_entry(prop->name, dn->pde); > - prop = prop->next; > - } > - > - if (dn->pde) > - remove_proc_entry(dn->pde->name, parent->pde); > + proc_remove(dn->pde); > } > #else > static void of_remove_proc_dt_entry(struct device_node *dn) > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 7cde7c1..0812608 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -427,20 +427,14 @@ int pci_proc_attach_device(struct pci_dev *dev) > > int pci_proc_detach_device(struct pci_dev *dev) > { > - struct proc_dir_entry *e; > - > - if ((e = dev->procent)) { > - remove_proc_entry(e->name, dev->bus->procdir); > - dev->procent = NULL; > - } > + proc_remove(dev->procent); > + dev->procent = NULL; > return 0; > } > > int pci_proc_detach_bus(struct pci_bus* bus) > { > - struct proc_dir_entry *de = bus->procdir; > - if (de) > - remove_proc_entry(de->name, proc_bus_pci_dir); > + proc_remove(bus->procdir); > return 0; > } > > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 2c6d6be..4b48033 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -630,3 +630,10 @@ const char *get_proc_name(const struct proc_dir_entry *de) > return de->name; > } > EXPORT_SYMBOL_GPL(get_proc_name); > + > +void proc_remove(struct proc_dir_entry *de) > +{ > + if (de) > + remove_proc_subtree(de->name, de->parent); > +} > +EXPORT_SYMBOL(proc_remove); > diff --git a/fs/proc/vmcore.c b/fs/proc/vmcore.c > index 38edddc..17f7e08 100644 > --- a/fs/proc/vmcore.c > +++ b/fs/proc/vmcore.c > @@ -699,7 +699,7 @@ void vmcore_cleanup(void) > struct list_head *pos, *next; > > if (proc_vmcore) { > - remove_proc_entry(proc_vmcore->name, proc_vmcore->parent); > + proc_remove(proc_vmcore); > proc_vmcore = NULL; > } > > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 7abc72a..b8949b1 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -67,6 +67,7 @@ struct proc_dir_entry *proc_create_data(const char *name, umode_t mode, > struct proc_dir_entry *parent, > const struct file_operations *proc_fops, > void *data); > +extern void proc_remove(struct proc_dir_entry *); > extern void remove_proc_entry(const char *name, struct proc_dir_entry *parent); > extern int remove_proc_subtree(const char *name, struct proc_dir_entry *parent); > > @@ -103,6 +104,7 @@ static inline struct proc_dir_entry *proc_create_data(const char *name, > { > return NULL; > } > +static inline void proc_remove(struct proc_dir_entry *de) {} > #define remove_proc_entry(name, parent) do {} while (0) > #define remove_proc_subtree(name, parent) do {} while (0) > > diff --git a/kernel/irq/proc.c b/kernel/irq/proc.c > index d59ae37..19ed5c4 100644 > --- a/kernel/irq/proc.c > +++ b/kernel/irq/proc.c > @@ -366,11 +366,7 @@ void unregister_irq_proc(unsigned int irq, struct irq_desc *desc) > > void unregister_handler_proc(unsigned int irq, struct irqaction *action) > { > - if (action->dir) { > - struct irq_desc *desc = irq_to_desc(irq); > - > - remove_proc_entry(action->dir->name, desc->dir); > - } > + proc_remove(action->dir); > } > > static void register_default_affinity_proc(void) > diff --git a/net/8021q/vlanproc.c b/net/8021q/vlanproc.c > index 959ddbb..1d0e8921 100644 > --- a/net/8021q/vlanproc.c > +++ b/net/8021q/vlanproc.c > @@ -184,14 +184,9 @@ int vlan_proc_add_dev(struct net_device *vlandev) > */ > int vlan_proc_rem_dev(struct net_device *vlandev) > { > - struct vlan_net *vn = net_generic(dev_net(vlandev), vlan_net_id); > - > /** NOTE: This will consume the memory pointed to by dent, it seems. */ > - if (vlan_dev_priv(vlandev)->dent) { > - remove_proc_entry(vlan_dev_priv(vlandev)->dent->name, > - vn->proc_vlan_dir); > - vlan_dev_priv(vlandev)->dent = NULL; > - } > + proc_remove(vlan_dev_priv(vlandev)->dent); > + vlan_dev_priv(vlandev)->dent = NULL; > return 0; > } > > diff --git a/net/core/pktgen.c b/net/core/pktgen.c > index f6af4fe..6c41e97 100644 > --- a/net/core/pktgen.c > +++ b/net/core/pktgen.c > @@ -1904,7 +1904,7 @@ static void pktgen_change_name(const struct pktgen_net *pn, struct net_device *d > if (pkt_dev->odev != dev) > continue; > > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > pkt_dev->entry = proc_create_data(dev->name, 0600, > pn->proc_dir, > @@ -3576,8 +3576,6 @@ static void _rem_dev_from_if_list(struct pktgen_thread *t, > static int pktgen_remove_device(struct pktgen_thread *t, > struct pktgen_dev *pkt_dev) > { > - struct pktgen_net *pn = t->net; > - > pr_debug("remove_device pkt_dev=%p\n", pkt_dev); > > if (pkt_dev->running) { > @@ -3597,7 +3595,7 @@ static int pktgen_remove_device(struct pktgen_thread *t, > _rem_dev_from_if_list(t, pkt_dev); > > if (pkt_dev->entry) > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > #ifdef CONFIG_XFRM > free_SAs(pkt_dev); > diff --git a/net/ipv4/netfilter/ipt_CLUSTERIP.c b/net/ipv4/netfilter/ipt_CLUSTERIP.c > index e4738fe..0b732ef 100644 > --- a/net/ipv4/netfilter/ipt_CLUSTERIP.c > +++ b/net/ipv4/netfilter/ipt_CLUSTERIP.c > @@ -105,7 +105,7 @@ clusterip_config_entry_put(struct clusterip_config *c) > * functions are also incrementing the refcount on their own, > * so it's safe to remove the entry even if it's in use. */ > #ifdef CONFIG_PROC_FS > - remove_proc_entry(c->pde->name, c->pde->parent); > + proc_remove(c->pde); > #endif > return; > } > @@ -736,7 +736,7 @@ static void __exit clusterip_tg_exit(void) > { > pr_info("ClusterIP Version %s unloading\n", CLUSTERIP_VERSION); > #ifdef CONFIG_PROC_FS > - remove_proc_entry(clusterip_procdir->name, clusterip_procdir->parent); > + proc_remove(clusterip_procdir); > #endif > nf_unregister_hook(&cip_arp_ops); > xt_unregister_target(&clusterip_tg_reg); > diff --git a/net/ipv6/proc.c b/net/ipv6/proc.c > index 7ea6e18..537d9ee 100644 > --- a/net/ipv6/proc.c > +++ b/net/ipv6/proc.c > @@ -287,8 +287,7 @@ int snmp6_unregister_dev(struct inet6_dev *idev) > return -ENOENT; > if (!idev->stats.proc_dir_entry) > return -EINVAL; > - remove_proc_entry(idev->stats.proc_dir_entry->name, > - net->mib.proc_net_devsnmp6); > + proc_remove(idev->stats.proc_dir_entry); > idev->stats.proc_dir_entry = NULL; > return 0; > } > diff --git a/net/netfilter/xt_hashlimit.c b/net/netfilter/xt_hashlimit.c > index d2fd53b..40002ef 100644 > --- a/net/netfilter/xt_hashlimit.c > +++ b/net/netfilter/xt_hashlimit.c > @@ -330,7 +330,7 @@ static void htable_destroy(struct xt_hashlimit_htable *hinfo) > parent = hashlimit_net->ip6t_hashlimit; > > if(parent != NULL) > - remove_proc_entry(hinfo->pde->name, parent); > + proc_remove(hinfo->pde); > > htable_selective_cleanup(hinfo, select_all); > vfree(hinfo); > @@ -887,7 +887,7 @@ static void __net_exit hashlimit_proc_net_exit(struct net *net) > pde = hashlimit_net->ip6t_hashlimit; > > hlist_for_each_entry(hinfo, &hashlimit_net->htables, node) > - remove_proc_entry(hinfo->pde->name, pde); > + proc_remove(hinfo->pde); > > hashlimit_net->ipt_hashlimit = NULL; > hashlimit_net->ip6t_hashlimit = NULL; > diff --git a/sound/core/info.c b/sound/core/info.c > index c7f41c3..3c9bd6b 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -153,13 +153,6 @@ EXPORT_SYMBOL(snd_seq_root); > struct snd_info_entry *snd_oss_root; > #endif > > -static void snd_remove_proc_entry(struct proc_dir_entry *parent, > - struct proc_dir_entry *de) > -{ > - if (de) > - remove_proc_entry(de->name, parent); > -} > - > static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) > { > struct snd_info_private_data *data; > @@ -580,7 +573,7 @@ int __exit snd_info_done(void) > #ifdef CONFIG_SND_OSSEMUL > snd_info_free_entry(snd_oss_root); > #endif > - snd_remove_proc_entry(NULL, snd_proc_root); > + proc_remove(snd_proc_root); > } > return 0; > } > @@ -642,7 +635,7 @@ void snd_info_card_id_change(struct snd_card *card) > { > mutex_lock(&info_mutex); > if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > + proc_remove(card->proc_root_link); > card->proc_root_link = NULL; > } > if (strcmp(card->id, card->proc_root->name)) > @@ -661,10 +654,8 @@ void snd_info_card_disconnect(struct snd_card *card) > if (!card) > return; > mutex_lock(&info_mutex); > - if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > - card->proc_root_link = NULL; > - } > + proc_remove(card->proc_root_link); > + card->proc_root_link = NULL; > if (card->proc_root) > snd_info_disconnect(card->proc_root); > mutex_unlock(&info_mutex); > @@ -856,7 +847,7 @@ static void snd_info_disconnect(struct snd_info_entry *entry) > list_del_init(&entry->list); > root = entry->parent == NULL ? snd_proc_root : entry->parent->p; > snd_BUG_ON(!root); > - snd_remove_proc_entry(root, entry->p); > + proc_remove(entry->p); > entry->p = NULL; > } > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-pci" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH 00/17] ASoC: Towards a generic dmaengine PCM driver
by Lars-Peter Clausen 18 Apr '13

18 Apr '13

Hi, This series adds a (almost) generic dmaengine PCM driver for ASoC. The tegra, spear, omap, imx, mxs, ep93xx and ux500 platforms are converted to use the generic driver instead of their own custom implementation. The driver builds on top of the dmaengine PCM library and adds the missing bits. The driver was primarily built for handling devicetree based platforms, but the series also adds support for compat platforms which don't use devicetree yet or have to support non-devicetree instantiation for backwards compatibility. Since non of the converted platforms have devicetree support yet (at least not using the generic DMA bindings) this series doesn't enable devicetree support for them either. But enabling support for them is very easy and can be done by removing the SND_DMAENGINE_PCM_NO_DT flag for these platforms. For platforms using devicetree the DMA channels are requested early on before the ASoC platform driver is registered. This allows us, at least in theory, to use probe deferral if the DMA driver hasn't been probed yet. Unfortunately of_dma_request_slave_channel either returns a valid channel or NULL, so we can't distinguish between the case where there is no DMA channel assigned and the case where the DMA driver hasn't been probed yet. I plan to submit a patch which lets of_dma_request_slave_channel return a ERR_PTR, but for now the generic PCM dmaengine driver assumes that the DMA driver will always be probed before it. For the compat case the DMA channel gets requested from within the PCM driver's pcm_new callback. This requires some minor changes to some of the DAI drivers moving the initialization of the DAI DMA data pointer from the DAI startup callback to the DAI probe callback. Unfortunately the driver is not fully generic yet and we still need some platform specific data to make it work, but the plan is to eventually make the driver fully generic in the future. The platform specific bits currently are: * The snd_pcm_hardware struct which contains device specific parameters like the maximum period size and maximum number of periods as well as whether the driver supports pause and resume. The idea is to get rid of the per platform snd_pcm_hardware struct by extending the dmaengine API to a point where we can query all necessary information from the dmaengine driver to fill in the snd_pcm_hardware struct at runtime. * The size of the pre-allocated audio buffer. All platforms seem use different sizes and since some of them might be sensitive to changes to the buffer size this series keeps the original size for now. But we should probably try to decide on a sane default for this and maybe allow ASoC board drivers to overwrite it somehow. * A callback to fill in the dma_slave_config. Currently this is either NULL (for platforms which don't need to configure the DMA channel at runtime) or uses the generic snd_dmaengine_pcm_prepare_slave_config(), which uses snd_hwparams_to_dma_slave_config() and snd_dmaengine_pcm_set_config_from_dai_data(). The later requires the platform to use the generic DMA DAI data struct though, hopefully we'll be able to convert all platforms to use this generic struct, and once that's done we can always use snd_dmaengine_pcm_prepare_slave_config() unconditionally. * For platforms not using devicetree platforms a callback to request the dma channel. This can either be a completely custom function which returns the channel or a filter function which will be passed to dma_request_channel. The route forward here probably is to let the DAI driver provide the filter function or implement a table based DMA channel lookup like we have for other subsystems (like clk or regulator) where the board file provides a mapping based on the device name. Platforms which use the dmaengine pcm library but are not converted to the generic driver yet are atmel and mmp. mmp use a custom genalloc based memory allocator, we may add support for this to the generic driver at a later point. The atmel PCM driver is very interlocked with atmel SSC DAI driver, it'll probably require some serious refactoring before it can use the generic dmaengine PCM driver. The driver uses snd_pcm_lib_preallocate_pages to preallocate the memory buffer. snd_pcm_lib_preallocate_pages internally uses dma_alloc_coherent, while most of the platforms converted in this series use dma_alloc_writecombine. Laxman Dewangan submitted a patch[1] a couple of months ago to add support for dma_alloc_writecombine to snd_pcm_lib_preallocate_pages, I always thought the patch had been merged, but it looks like I was mistaken. So if there are concerns about performance regression due to using coherent instead of writecombine mappings we'll have to postpone converting these platforms until Laxman's patch has been merged. The compat path of this series hasn't been runtime tested yet and neither have the platforms which are converted in this series. I probably broke one or two of them by accident, so please test the patches. Thanks, - Lars [1] https://lkml.org/lkml/2012/7/2/94 Lars-Peter Clausen (17): ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional ASoC: Add snd_soc_{add,remove}_platform ASoC: Add a generic dmaengine_pcm driver ASoC: dmaengine_pcm: Add support for compat platforms ASoC: tegra: Use generic dmaengine PCM ASoC: spear: Setup dma data in DAI probe ASoC: spear: Use generic dmaengine PCM ASoC: dmaengine-pcm: Add support for platforms which can't report residue ASoC: mxs: Setup dma data in DAI probe ASoC: mxs: Use generic dmaengine PCM ASoC: imx: Setup dma data in DAI probe ASoC: imx: Use generic dmaengine PCM ASoC: omap: Setup dma data in DAI probe ASoC: omap: Use generic dmaengine PCM ASoC: ep93xx: Setup dma data in DAI probe ASoC: ep93xx: Use generic dmaengine PCM ASoC: ux500: Use generic dmaengine PCM include/sound/dmaengine_pcm.h | 65 +++++++- include/sound/soc.h | 4 + sound/soc/Kconfig | 4 + sound/soc/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 6 +- sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 14 +- sound/soc/cirrus/ep93xx-i2s.c | 14 +- sound/soc/cirrus/ep93xx-pcm.c | 137 ++-------------- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 19 ++- sound/soc/fsl/imx-pcm-dma.c | 76 ++------- sound/soc/fsl/imx-pcm.c | 6 +- sound/soc/fsl/imx-pcm.h | 5 + sound/soc/fsl/imx-ssi.c | 22 +-- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 135 ++-------------- sound/soc/mxs/mxs-saif.c | 3 +- sound/soc/omap/Kconfig | 2 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi.c | 12 +- sound/soc/omap/omap-mcbsp.c | 6 +- sound/soc/omap/omap-mcpdm.c | 7 +- sound/soc/omap/omap-pcm.c | 186 ++-------------------- sound/soc/pxa/mmp-pcm.c | 5 +- sound/soc/soc-core.c | 85 +++++++--- sound/soc/soc-dmaengine-pcm.c | 82 +++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 284 ++++++++++++++++++++++++++++++++++ sound/soc/spear/spdif_in.c | 12 +- sound/soc/spear/spdif_out.c | 7 +- sound/soc/spear/spear_pcm.c | 151 ++---------------- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra_pcm.c | 171 ++------------------ sound/soc/ux500/Kconfig | 2 +- sound/soc/ux500/ux500_pcm.c | 159 ++----------------- 35 files changed, 651 insertions(+), 1046 deletions(-) create mode 100644 sound/soc/soc-generic-dmaengine-pcm.c -- 1.8.0

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Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Bjorn Helgaas 18 Apr '13

18 Apr '13

On Tue, Apr 16, 2013 at 12:26 PM, David Howells <dhowells(a)redhat.com> wrote: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- > drivers/media/pci/ttpci/av7110_ir.c | 2 +- > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- For the drivers/pci part: Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL, &ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL, &av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long) &av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry, &proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry, &proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry, &proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry, &proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry, &proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i < 4 && io[i] && irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL, &proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list, &recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > -- > To unsubscribe from this list: send the line "unsubscribe linux-pci" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH v4 00/10] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 18 Apr '13

18 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. Regards, Markus Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (10): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 ++ Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 14 ++ arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++---------- sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 +- sound/soc/fsl/imx-pcm.h | 9 ++ sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 166 ++++++++++++++++--- 12 files changed, 633 insertions(+), 122 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

3 13

[alsa-devel] More ASoC updates for v3.10
by Mark Brown 18 Apr '13

18 Apr '13

The following changes since commit b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next for you to fetch changes up to 24568ea4bef5ab8106206eddf5512434421c00ed: Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next (2013-04-18 15:05:35 +0100) There's a bunch of extra stuff in the diffstat/log since your for-next is further behind Linus' tree than one of my branches. ---------------------------------------------------------------- ASoC: More updates for v3.10 The main additional change here is Lars-Peter's DMA work plus the platform conversions which have been tested - getting this in mainline will make life easier for development after the merge window. These factor a large chunk of code out of the drivers for the platforms using dmaengine, greatly simplifying development. ---------------------------------------------------------------- Al Viro (3): ecryptfs: close rmmod race procfs: add proc_remove_subtree() palinfo fixes Alex Williamson (1): vfio-pci: Fix possible integer overflow Alexandre Belloni (1): gpio: pca953x: fix irq_domain_add_simple usage Alexey Khoroshilov (1): tty: mxser: fix cycle termination condition in mxser_probe() and mxser_module_init() Alexey Pelykh (1): OMAP/serial: Revert bad fix of Rx FIFO threshold granularity Andrea Arcangeli (2): x86/mm/cpa: Convert noop to functional fix x86/mm/cpa/selftest: Fix false positive in CPA self test Andrey Vagin (1): mnt: release locks on error path in do_loopback Arnd Bergmann (5): block: avoid using uninitialized value in from queue_var_store Merge tag 'mxs-fixes-3.9-4' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'imx-fixes-3.9-5' of git://git.linaro.org/people/shawnguo/linux-2.6 into fixes Merge tag 'samsung-fixes-for-v3.9' of git://git.kernel.org/.../kgene/linux-samsung into fixes Merge tag 'mvebu_fixes_for_v3.9_round3' of git://git.infradead.org/users/jcooper/linux into fixes Artem Savkov (1): cfg80211: sched_scan_mtx lock in cfg80211_conn_work() Arun Easi (1): [SCSI] qla2xxx: Fix crash during firmware dump procedure. Asai Thambi S P (3): mtip32xx: recovery from command timeout mtip32xx: return 0 from pci probe in case of rebuild mtip32xx: Add debugfs entry device_status Asias He (7): tcm_vhost: Use ACCESS_ONCE for vs->vs_tpg[target] access tcm_vhost: Use vq->private_data to indicate if the endpoint is setup tcm_vhost: Initialize vq->last_used_idx when set endpoint tcm_vhost: Remove double check of response tcm_vhost: Fix tv_cmd leak in vhost_scsi_handle_vq tcm_vhost: Add vhost_scsi_send_bad_target() helper tcm_vhost: Send bad target to guest when cmd fails Bing Zhao (1): mwifiex: complete last internal scan Boris Ostrovsky (2): x86, mm: Patch out arch_flush_lazy_mmu_mode() when running on bare metal x86/mm: Flush lazy MMU when DEBUG_PAGEALLOC is set Brian King (1): [SCSI] ibmvscsi: Fix slave_configure deadlock Calvin Owens (1): drm/nouveau: fix unconditional return waiting on memory Chen Gang (3): perf: Fix strncpy() use, always make sure it's NUL terminated perf: Fix strncpy() use, use strlcpy() instead of strncpy() ftrace: Fix strncpy() use, use strlcpy() instead of strncpy() Chris Metcalf (1): tile: comment assumption about __insn_mtspr for <asm/irqflags.h> Christian Ruppert (1): ARC: Add implicit compiler barrier to raw_local_irq* functions Christoph Paasch (1): ipv6/tcp: Stop processing ICMPv6 redirect messages Christopher Harvey (1): drm/mgag200: Index 24 in extended CRTC registers is 24 in hex, not decimal. Chuck Lever (1): SUNRPC: Remove extra xprt_put() Daniel Vetter (1): drm/fb-helper: Fix locking in drm_fb_helper_hotplug_event Dave Airlie (2): Merge branch 'drm-nouveau-fixes-3.9' of git://anongit.freedesktop.org/git/nouveau/linux-2.6 into drm-fixes udl: handle EDID failure properly. Dave Hansen (1): x86-32: Fix possible incomplete TLB invalidate with PAE pagetables David S. Miller (3): Merge branch 'infoleaks' Merge branch 'wireless' Merge branch 'fixes-for-3.9' of git://gitorious.org/linux-can/linux-can David Woodhouse (1): libata: fix DMA to stack in reading devslp_timing parameters Dirk Behme (1): ARM i.MX6: Fix ldb_di clock selection Dirk Brandewie (1): cpufreq / intel_pstate: Set timer timeout correctly Dmitry Tarnyagin (1): remoteproc/ste: fix memory leak on shutdown Dylan Reid (1): ASoC: max98088: Fix logging of hardware revision. Eldad Zack (1): ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_* Eric Dumazet (1): selinux: add a skb_owned_by() hook Franky Lin (1): brcmfmac: do not proceed if fail to download nvram to dongle Gabor Juhos (1): rt2x00: rt2x00pci: fix build error on Ralink RT3x5x SoCs Greg Ungerer (1): m68k: define a local gpio_request_one() function Gregory CLEMENT (1): arm: mvebu: Fix the irq map function in SMP mode Guennadi Liakhovetski (1): tty: Documentation: fix a path in a DocBook template Guenter Roeck (1): misc/vmw_vmci: Add dependency on CONFIG_NET Haiyang Zhang (2): hyperv: Fix a kernel warning from netvsc_linkstatus_callback() hyperv: Fix RNDIS send_completion code path Hante Meuleman (3): brcmfmac: fix tkip mic tx/rx ap swap bug. brcmfmac: fix stopping AP. brcmfmac: fix returning cipher_suite for get_key operation. Huacai Chen (1): PM / reboot: call syscore_shutdown() after disable_nonboot_cpus() Jan Kiszka (1): ftrace: Consistently restore trace function on sysctl enabling Jens Axboe (2): mtip32xx: fix two smatch warnings Revert "loop: cleanup partitions when detaching loop device" Jiri Pirko (2): net: ipv4: reset check_lifetime_work after changing lifetime net: ipv4: fix schedule while atomic bug in check_lifetime() Joe Carnuccio (1): [SCSI] Revert "qla2xxx: Add setting of driver version string for vendor application." Joe Lawrence (1): [SCSI] st: Take additional queue ref in st_probe John Gong (1): [SCSI] libsas: use right function to alloc smp response John W. Linville (2): Merge branch 'for-john' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Josef Bacik (1): Btrfs: make sure nbytes are right after log replay Jussi Kivilinna (1): crypto: gcm - fix assumption that assoc has one segment Keir Fraser (1): xen/events: Handle VIRQ_TIMER before any other hardirq in event loop. Kleber Sacilotto de Souza (1): [SCSI] ipr: fix addition of abort command to HRRQ free queue Konrad Rzeszutek Wilk (1): xen/mmu: On early bootup, flush the TLB when changing RO->RW bits Xen provided pagetables. Krzysztof Mazur (1): [libata] Fix HDIO_DRIVE_* ioctl() Linux 3.9 regression Lars-Peter Clausen (9): ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional ASoC: Add snd_soc_{add, remove}_platform ASoC: Add a generic dmaengine_pcm driver ASoC: dmaengine_pcm: Add support for compat platforms ASoC: tegra: Use generic dmaengine PCM ASoC: dmaengine-pcm: Add support for platforms which can't report residue ASoC: imx: Setup dma data in DAI probe ASoC: imx: Use generic dmaengine PCM ASoC: ux500: Use generic dmaengine PCM Li Fei (1): hwspinlock: fix __hwspin_lock_request error path Linus Torvalds (39): Merge tag 'trace-fixes-3.9-rc6' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'upstream-linus' of git://git.kernel.org/.../jgarzik/libata-dev Merge tag 'for-linus' of git://git.kernel.org/.../sfr/next-fixes Merge tag 'remoteproc-3.9-fixes' of git://git.kernel.org/.../ohad/remoteproc Merge tag 'hwspinlock-3.9-fix' of git://git.kernel.org/.../ohad/hwspinlock spinlocks and preemption points need to be at least compiler barriers Merge tag 'for-linus-20130409' of git://git.kernel.dk/linux-block Merge tag 'vfio-v3.9-rc7' of git://github.com/awilliam/linux-vfio Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge git://git.kernel.org/.../herbert/crypto-2.6 Merge tag 'nfs-for-3.9-4' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Merge tag 'nfs-for-3.9-5' of git://git.linux-nfs.org/projects/trondmy/linux-nfs Merge tag 'for-linus' of git://linux-c6x.org/git/projects/linux-c6x-upstreaming Merge git://git.kernel.org/.../davem/net Merge tag 'trace-fixes-3.9-rc-v2' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'stable/for-linus-3.9-rc6-tag' of git://git.kernel.org/.../konrad/xen Merge tag 'tty-3.9-rc6' of git://git.kernel.org/.../gregkh/tty Merge tag 'char-misc-3.9-rc6' of git://git.kernel.org/.../gregkh/char-misc Merge tag 'arm-soc-fixes' of git://git.kernel.org/.../arm/arm-soc Merge tag 'gpio-fixes-v3.9-rc6' of git://git.kernel.org/.../linusw/linux-gpio Merge tag 'regmap-v3.9-rc6' of git://git.kernel.org/.../broonie/regmap Merge tag 'pm-3.9-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma Merge tag 'sound-3.9' of git://git.kernel.org/.../tiwai/sound Merge branch 'for-next' of git://git.samba.org/sfrench/cifs-2.6 Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge git://git.kernel.org/.../nab/target-pending kobject: fix kset_find_obj() race with concurrent last kobject_put() Add file_ns_capable() helper function for open-time capability checking Merge tag 'trace-fixes-v3.9-rc-v3' of git://git.kernel.org/.../rostedt/linux-trace Merge branch 'for-linus' of git://git.kernel.org/.../mason/linux-btrfs Merge git://www.linux-watchdog.org/linux-watchdog Merge branch 'for-linus' of git://git.kernel.org/.../gerg/m68knommu Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 3.9-rc7 Linus Walleij (1): crypto: ux500 - add missing comma Lukasz Dorau (1): [SCSI] libsas: fix handling vacant phy in sas_set_ex_phy() Marek Vasut (1): ARM: mxs: Slow down the I2C clock speed Mark Brown (13): regmap: Back out work buffer fix Merge remote-tracking branch 'asoc/fix/compress' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/samsung' into tmp Merge remote-tracking branch 'asoc/fix/tegra' into tmp Merge remote-tracking branch 'asoc/fix/wm8903' into tmp Merge branch 'topic/core' of git://git.kernel.org/.../broonie/sound into asoc-dma Merge tag 'v3.9-rc7' into asoc-dma Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Mark Salter (1): add memory barrier to arch_local_irq_restore Markus Pargmann (3): ARM: clk-imx35: Bugfix iomux clock ARM: imx35 Bugfix admux clock ASoC: fsl-ssi: Add SACNT definitions Mathias Krause (16): atm: update msg_namelen in vcc_recvmsg() ax25: fix info leak via msg_name in ax25_recvmsg() Bluetooth: fix possible info leak in bt_sock_recvmsg() Bluetooth: RFCOMM - Fix missing msg_namelen update in rfcomm_sock_recvmsg() Bluetooth: SCO - Fix missing msg_namelen update in sco_sock_recvmsg() caif: Fix missing msg_namelen update in caif_seqpkt_recvmsg() irda: Fix missing msg_namelen update in irda_recvmsg_dgram() iucv: Fix missing msg_namelen update in iucv_sock_recvmsg() l2tp: fix info leak in l2tp_ip6_recvmsg() llc: Fix missing msg_namelen update in llc_ui_recvmsg() netrom: fix info leak via msg_name in nr_recvmsg() NFC: llcp: fix info leaks via msg_name in llcp_sock_recvmsg() rose: fix info leak via msg_name in rose_recvmsg() tipc: fix info leaks via msg_name in recv_msg/recv_stream VSOCK: vmci - fix possible info leak in vmci_transport_dgram_dequeue() VSOCK: Fix missing msg_namelen update in vsock_stream_recvmsg() Michael Riesch (1): rtnetlink: Call nlmsg_parse() with correct header length Michael Wolf (1): powerpc: pSeries_lpar_hpte_remove fails from Adjunct partition being performed before the ANDCOND test Namhyung Kim (2): tracing: Fix double free when function profile init failed tracing: Fix possible NULL pointer dereferences Neil Horman (1): e100: Add dma mapping error check Nicholas Bellinger (1): target: Fix incorrect fallthrough of ALUA Standby/Offline/Transition CDBs Nicolas Ferre (1): watchdog: Revert the AT91RM9200_WATCHDOG dependency Nigel Roberts (2): ARM: Kirkwood: Fix typo in the definition of ix2-200 rebuild LED Fix GE0/GE1 init on ix2-200 as GE0 has no PHY Padmavathi Venna (1): DMA: PL330: Add check if device tree compatible Paul Moore (1): lsm: add the missing documentation for the security_skb_owned_by() hook Peter Ujfalusi (1): dmaengine: omap-dma: Start DMA without delay for cyclic channels Robert Shade (1): ath9k: Re-enable interrupts after a channel change failure Robert Tivy (1): remoteproc: fix FW_CONFIG typo Roland Stigge (1): MAINTAINERS: Add maintainer for LPC32xx Sachin Prabhu (1): cifs: Allow passwords which begin with a delimitor Samu Kallio (1): x86, mm, paravirt: Fix vmalloc_fault oops during lazy MMU updates Saurav Kashyap (3): [SCSI] qla2xxx: Update copyright dates information in LICENSE.qla2xxx file. [SCSI] qla2xxx: Remove debug code that msleeps for random duration. [SCSI] qla2xxx: Update the driver version to 8.04.00.13-k. Sean Young (1): Revert "tty/8250_pnp: serial port detection regression since v3.7" Shan Hai (2): libata: Use integer return value for atapi_command_packet_set libata: Set max sector to 65535 for Slimtype DVD A DS8A8SH drive Shawn Guo (1): ARM: imx: provide twd clock lookup from device tree Sjur Brændeland (1): remoteproc: fix error path of handle_vdev Sridhar Samudrala (1): sctp: remove 'sridhar' from maintainers list Stanislaw Gruszka (1): sched/cputime: Fix accounting on multi-threaded processes Stas Sergeev (1): ASoC: define playback and capture streams in dummy codec Stefan Raspl (1): qeth: fix qeth_wait_for_threads() deadlock for OSN devices Stephane Eranian (2): perf/x86: Fix uninitialized pt_regs in intel_pmu_drain_bts_buffer() perf: Fix ring_buffer perf_output_space() boundary calculation Steven Rostedt (Red Hat) (3): tracing: Fix race with update_max_tr_single and changing tracers ftrace: Do not call stub functions in control loop ftrace: Move ftrace_filter_lseek out of CONFIG_DYNAMIC_FTRACE section Suleiman Souhlal (1): vfs: Revert spurious fix to spinning prevention in prune_icache_sb Suman Anna (1): remoteproc: fix the error check for idr_alloc Sylwester Nawrocki (2): ARM: S3C24XX: Correct NR_IRQS definition for s3c2440 ARM: S3C24XX: Fix interrupt pending register offset of the EINT controller Takashi Iwai (2): Merge tag 'asoc-v3.9-rc6' of git://git.kernel.org/.../broonie/sound into for-linus Merge tag 'asoc-maintainers-v3.9-rc6' of git://git.kernel.org/.../broonie/sound into for-linus Tejun Heo (1): sched: Convert BUG_ON()s in try_to_wake_up_local() to WARN_ON_ONCE()s Thomas Gleixner (1): sched_clock: Prevent 64bit inatomicity on 32bit systems Tim Gardner (1): rt2x00: rt2x00pci_regbusy_read() - only print register access failure once Trond Myklebust (5): NFSv4: Fix a memory leak in nfs4_discover_server_trunking NFSv4/4.1: Fix bugs in nfs4[01]_walk_client_list SUNRPC: Fix a potential memory leak in rpc_new_client NFSv4: Fix another potential state manager deadlock NFSv4: Doh! Typo in the fix to nfs41_walk_client_list Ursula Braun (1): af_iucv: fix recvmsg by replacing skb_pull() function Wei Yongjun (4): netrom: fix invalid use of sizeof in nr_recvmsg() can: gw: use kmem_cache_free() instead of kfree() [SCSI] lpfc: fix potential NULL pointer dereference in lpfc_sli4_rq_put() perf: Fix error return code Yaniv Rosner (1): bnx2x: Fix KR2 rapid link flap Youquan Song (1): ata_piix: Fix DVD not dectected at some Haswell platforms Yuval Mintz (1): bnx2x: Prevent null pointer dereference in AFEX mode libin (1): sched/debug: Fix sd->*_idx limit range avoiding overflow nikolay(a)redhat.com (2): Revert "bonding: remove sysfs before removing devices" bonding: fix bonding_masters race condition in bond unloading wenxiong(a)linux.vnet.ibm.com (2): [SCSI] ipr: ipr_test_msi() fails when running with msi-x enabled adapter [SCSI] ipr: dlpar failed when adding an adapter back willy tarreau (1): net: mvneta: enable features before registering the driver Documentation/DocBook/device-drivers.tmpl | 2 +- Documentation/scsi/LICENSE.qla2xxx | 2 +- MAINTAINERS | 7 +- Makefile | 2 +- arch/arc/include/asm/irqflags.h | 12 +- arch/arm/boot/dts/imx28-m28evk.dts | 1 - arch/arm/boot/dts/imx28-sps1.dts | 1 - arch/arm/boot/dts/imx6qdl.dtsi | 1 + arch/arm/boot/dts/kirkwood-iomega_ix2_200.dts | 14 +- arch/arm/mach-imx/clk-imx35.c | 2 + arch/arm/mach-imx/clk-imx6q.c | 3 +- arch/arm/mach-kirkwood/board-iomega_ix2_200.c | 7 +- arch/arm/mach-mvebu/irq-armada-370-xp.c | 16 +- arch/arm/mach-s3c24xx/include/mach/irqs.h | 4 +- arch/arm/mach-s3c24xx/irq.c | 2 +- arch/c6x/include/asm/irqflags.h | 2 +- arch/ia64/kernel/palinfo.c | 77 +---- arch/m68k/include/asm/gpio.h | 20 ++ arch/powerpc/platforms/pseries/lpar.c | 8 +- arch/tile/include/asm/irqflags.h | 10 +- arch/x86/include/asm/paravirt.h | 5 +- arch/x86/include/asm/paravirt_types.h | 2 + arch/x86/include/asm/tlb.h | 2 +- arch/x86/kernel/cpu/perf_event_intel_ds.c | 3 +- arch/x86/kernel/paravirt.c | 25 +- arch/x86/lguest/boot.c | 1 + arch/x86/mm/fault.c | 6 +- arch/x86/mm/pageattr-test.c | 2 +- arch/x86/mm/pageattr.c | 12 +- arch/x86/mm/pgtable.c | 7 + arch/x86/xen/mmu.c | 13 +- block/blk-sysfs.c | 2 + block/partition-generic.c | 1 - crypto/gcm.c | 17 +- drivers/ata/ata_piix.c | 14 +- drivers/ata/libata-core.c | 6 +- drivers/ata/libata-scsi.c | 8 +- drivers/base/regmap/regmap.c | 3 +- drivers/block/loop.c | 21 +- drivers/block/mtip32xx/mtip32xx.c | 327 ++++++++++++++------ drivers/block/mtip32xx/mtip32xx.h | 18 +- drivers/cpufreq/intel_pstate.c | 1 - drivers/crypto/ux500/cryp/cryp_core.c | 2 +- drivers/dma/omap-dma.c | 20 +- drivers/dma/pl330.c | 38 ++- drivers/gpio/gpio-pca953x.c | 2 +- drivers/gpu/drm/drm_fb_helper.c | 8 +- drivers/gpu/drm/mgag200/mgag200_mode.c | 13 +- drivers/gpu/drm/nouveau/nv50_display.c | 2 +- drivers/gpu/drm/udl/udl_connector.c | 4 + drivers/hwspinlock/hwspinlock_core.c | 2 + drivers/misc/vmw_vmci/Kconfig | 2 +- drivers/net/bonding/bond_main.c | 11 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_link.c | 18 +- drivers/net/ethernet/broadcom/bnx2x/bnx2x_main.c | 3 +- drivers/net/ethernet/intel/e100.c | 36 ++- drivers/net/ethernet/marvell/mvneta.c | 9 +- drivers/net/hyperv/netvsc.c | 17 +- drivers/net/hyperv/netvsc_drv.c | 2 - drivers/net/hyperv/rndis_filter.c | 14 +- drivers/net/wireless/ath/ath9k/main.c | 4 + drivers/net/wireless/brcm80211/brcmfmac/dhd_sdio.c | 6 +- .../net/wireless/brcm80211/brcmfmac/wl_cfg80211.c | 46 +-- drivers/net/wireless/mwifiex/scan.c | 11 +- drivers/net/wireless/rt2x00/Kconfig | 7 + drivers/net/wireless/rt2x00/Makefile | 1 + drivers/net/wireless/rt2x00/rt2400pci.c | 1 + drivers/net/wireless/rt2x00/rt2500pci.c | 1 + drivers/net/wireless/rt2x00/rt2800pci.c | 1 + drivers/net/wireless/rt2x00/rt2x00mmio.c | 216 +++++++++++++ drivers/net/wireless/rt2x00/rt2x00mmio.h | 119 +++++++ drivers/net/wireless/rt2x00/rt2x00pci.c | 176 ----------- drivers/net/wireless/rt2x00/rt2x00pci.h | 88 ------ drivers/net/wireless/rt2x00/rt61pci.c | 1 + drivers/remoteproc/Kconfig | 2 +- drivers/remoteproc/remoteproc_core.c | 6 +- drivers/remoteproc/ste_modem_rproc.c | 7 +- drivers/s390/net/qeth_core.h | 3 + drivers/s390/net/qeth_core_main.c | 19 ++ drivers/s390/net/qeth_l2_main.c | 2 + drivers/s390/net/qeth_l3_main.c | 2 + drivers/scsi/ibmvscsi/ibmvscsi.c | 2 +- drivers/scsi/ipr.c | 13 +- drivers/scsi/libsas/sas_expander.c | 14 +- drivers/scsi/lpfc/lpfc_sli.c | 3 +- drivers/scsi/qla2xxx/qla_attr.c | 5 - drivers/scsi/qla2xxx/qla_dbg.c | 3 +- drivers/scsi/qla2xxx/qla_def.h | 1 - drivers/scsi/qla2xxx/qla_gbl.h | 3 - drivers/scsi/qla2xxx/qla_init.c | 4 +- drivers/scsi/qla2xxx/qla_mbx.c | 58 ---- drivers/scsi/qla2xxx/qla_version.h | 2 +- drivers/scsi/st.c | 8 +- drivers/target/target_core_alua.c | 3 + drivers/tty/mxser.c | 8 +- drivers/tty/serial/8250/8250_pnp.c | 12 +- drivers/tty/serial/omap-serial.c | 11 + drivers/vfio/pci/vfio_pci.c | 3 +- drivers/vhost/tcm_vhost.c | 198 ++++++++---- drivers/watchdog/Kconfig | 2 +- drivers/xen/events.c | 19 +- fs/btrfs/tree-log.c | 48 ++- fs/cifs/connect.c | 16 +- fs/ecryptfs/miscdev.c | 14 +- fs/inode.c | 2 +- fs/namespace.c | 2 +- fs/nfs/nfs4client.c | 45 ++- fs/nfs/nfs4proc.c | 1 + fs/nfs/nfs4state.c | 8 +- fs/proc/generic.c | 119 +++++-- include/asm-generic/tlb.h | 7 +- include/linux/ata.h | 2 +- include/linux/capability.h | 2 + include/linux/ftrace.h | 5 +- include/linux/libata.h | 1 + include/linux/preempt.h | 22 +- include/linux/proc_fs.h | 2 + include/linux/security.h | 12 + include/linux/spinlock_up.h | 29 +- include/net/iucv/af_iucv.h | 8 + include/sound/dmaengine_pcm.h | 92 +++++- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 13 +- kernel/capability.c | 24 ++ kernel/events/core.c | 4 +- kernel/events/internal.h | 2 +- kernel/events/ring_buffer.c | 22 +- kernel/sched/clock.c | 26 ++ kernel/sched/core.c | 8 +- kernel/sched/cputime.c | 2 +- kernel/sys.c | 3 +- kernel/trace/ftrace.c | 54 ++-- kernel/trace/trace.c | 9 +- kernel/trace/trace_stack.c | 2 +- lib/kobject.c | 9 +- mm/memory.c | 1 + net/atm/common.c | 2 + net/ax25/af_ax25.c | 1 + net/bluetooth/af_bluetooth.c | 4 +- net/bluetooth/rfcomm/sock.c | 1 + net/bluetooth/sco.c | 1 + net/caif/caif_socket.c | 2 + net/can/gw.c | 6 +- net/core/rtnetlink.c | 4 +- net/ipv4/devinet.c | 60 +++- net/ipv4/tcp_output.c | 1 + net/ipv6/tcp_ipv6.c | 1 + net/irda/af_irda.c | 2 + net/iucv/af_iucv.c | 36 +-- net/l2tp/l2tp_ip6.c | 1 + net/llc/af_llc.c | 2 + net/netrom/af_netrom.c | 1 + net/nfc/llcp/sock.c | 3 + net/rose/af_rose.c | 1 + net/sunrpc/clnt.c | 11 +- net/tipc/socket.c | 7 + net/vmw_vsock/af_vsock.c | 2 + net/vmw_vsock/vmci_transport.c | 3 +- net/wireless/sme.c | 2 + security/capability.c | 6 + security/security.c | 5 + security/selinux/hooks.c | 7 + sound/soc/Kconfig | 4 + sound/soc/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 29 +- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 9 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 38 +-- sound/soc/cirrus/ep93xx-pcm.h | 20 -- sound/soc/cirrus/simone.c | 2 - sound/soc/cirrus/snappercl15.c | 1 - sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/cs4271.c | 166 +++++----- sound/soc/codecs/max98088.c | 32 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm8903.c | 2 + sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 55 ++-- sound/soc/fsl/fsl_ssi.h | 8 + sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 120 +------ sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-pcm.c | 6 +- sound/soc/fsl/imx-pcm.h | 24 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 51 ++- sound/soc/fsl/imx-ssi.h | 7 +- sound/soc/mxs/mxs-pcm.c | 43 +-- sound/soc/mxs/mxs-pcm.h | 4 +- sound/soc/mxs/mxs-saif.c | 6 +- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 +- sound/soc/omap/mcbsp.h | 7 +- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 38 +-- sound/soc/omap/omap-hdmi.c | 24 +- sound/soc/omap/omap-mcbsp.c | 18 +- sound/soc/omap/omap-mcpdm.c | 109 ++++--- sound/soc/omap/omap-pcm.c | 86 +---- sound/soc/omap/omap-pcm.h | 40 --- sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 9 +- sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/pxa/mmp-pcm.c | 34 +- sound/soc/samsung/i2s.c | 17 +- sound/soc/soc-compress.c | 14 +- sound/soc/soc-core.c | 116 ++++--- sound/soc/soc-dapm.c | 1 - sound/soc/soc-dmaengine-pcm.c | 150 +++++---- sound/soc/soc-generic-dmaengine-pcm.c | 284 +++++++++++++++++ sound/soc/soc-io.c | 5 +- sound/soc/soc-utils.c | 25 ++ sound/soc/spear/spear_pcm.c | 19 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 13 +- sound/soc/tegra/tegra20_ac97.h | 4 +- sound/soc/tegra/tegra20_i2s.c | 13 +- sound/soc/tegra/tegra20_i2s.h | 4 +- sound/soc/tegra/tegra20_spdif.c | 7 +- sound/soc/tegra/tegra20_spdif.h | 4 +- sound/soc/tegra/tegra30_ahub.c | 8 +- sound/soc/tegra/tegra30_ahub.h | 8 +- sound/soc/tegra/tegra30_i2s.c | 13 +- sound/soc/tegra/tegra30_i2s.h | 4 +- sound/soc/tegra/tegra_pcm.c | 209 +------------ sound/soc/tegra/tegra_pcm.h | 7 - sound/soc/ux500/Kconfig | 2 +- sound/soc/ux500/ux500_pcm.c | 229 ++------------ sound/soc/ux500/ux500_pcm.h | 14 - sound/usb/mixer_quirks.c | 4 +- sound/usb/quirks.c | 2 +- 235 files changed, 2790 insertions(+), 2211 deletions(-) create mode 100644 drivers/net/wireless/rt2x00/rt2x00mmio.c create mode 100644 drivers/net/wireless/rt2x00/rt2x00mmio.h delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.h create mode 100644 sound/soc/soc-generic-dmaengine-pcm.c

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[alsa-devel] [PATCH] sgtl5000.txt: Include supplies as required properties
by Fabio Estevam 18 Apr '13

18 Apr '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 driver cannot probe succesfully if VDDA and VDDIO regulators are not provided. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 9cc4444..520eb98 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -5,9 +5,15 @@ Required properties: - reg : the I2C address of the device +- VDDA-supply: regulator that provides VDDA voltage + +- VDDIO-supply: regulator that provides VDDIO voltage + Example: codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; + VDDA-supply = <&reg_2p5v>; + VDDIO-supply = <&reg_3p3v>; }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 18 Apr '13

18 Apr '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

2 2

[alsa-devel] [PATCH v3 0/5] ALSA: snd-usb: add support for DSD
by Daniel Mack 18 Apr '13

18 Apr '13

In contrast to the first two approaches, this one keeps the userspace unaware of the actual transport on the bus. More specifically, the DSD_U16_LE format may be implemented to actually send DOP samples to the device, depending on the quirks entries. Also, as there is quite some confusion in both data file formats as well as hardware sample formats about whether to stuff the oldest bit in the MSB or the LSB of a byte, ALSA defines that now to be in the MSB, as far as the ALSA userspace layer is concerned. The driver learned functions to re-encode the bytes according to what the hardware supports. I also wrote a very basic test tool which I published here: https://github.com/zonque/alsa-dsd-player Thanks, Daniel Daniel Mack (5): ALSA: add DSD formats ALSA: snd-usb: use ep->stride from urb callbacks ALSA: snd-usb: add support for DSD DOP stream transport ALSA: snd-usb: add support for bit-reversed byte formats ALSA: snd-usb: add quirks handler for DSD streams include/sound/pcm.h | 2 + include/uapi/sound/asound.h | 4 +- sound/core/pcm.c | 2 + sound/core/pcm_misc.c | 8 +++ sound/usb/card.h | 8 +++ sound/usb/endpoint.c | 9 ++++ sound/usb/format.c | 3 ++ sound/usb/pcm.c | 118 ++++++++++++++++++++++++++++++++++++------- sound/usb/quirks.c | 28 ++++++++++ sound/usb/quirks.h | 4 ++ 10 files changed, 167 insertions(+), 19 deletions(-) -- 1.7.10.4

3 12

Re: [alsa-devel] [PATCH -next] ALSA: hda - fix error return code in patch_alc662()
by Takashi Iwai 18 Apr '13

18 Apr '13

At Thu, 18 Apr 2013 11:12:59 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return a negative error code from the error handling > case instead of 0, as returned elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied. Takashi > --- > sound/pci/hda/patch_realtek.c | 3 ++- > 1 file changed, 2 insertions(+), 1 deletion(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index f15c36b..a3965b0 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3787,7 +3787,8 @@ static int patch_alc662(struct hda_codec *codec) > if ((alc_get_coef0(codec) & (1 << 14)) && > codec->bus->pci->subsystem_vendor == 0x1025 && > spec->cdefine.platform_type == 1) { > - if (alc_codec_rename(codec, "ALC272X") < 0) > + err = alc_codec_rename(codec, "ALC272X"); > + if (err < 0) > goto error; > } > >

1 0

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 17 Apr '13

17 Apr '13

On Tue, Apr 16, 2013 at 04:29:40PM -0600, Stephen Warren wrote: > a) There are many different VDD inputs to the chip. I imagine that some > of these only need to be enabled/active under certain conditions. For > example, perhaps the SPKVDD or MICVDD inputs are only required if > actively using the speakers or microphones, and could be disabled at > other times? If so, all of these "optional" or "part-time" VDD inputs > should be represented as regulators, and retrieved/manipulated using > Linux APIs such as regulator_get(), regulator_enable(), etc. There's framework support for regulator supplies in ASoC so you just need to specify the name and the core will do the rest. > b) Some VDD inputs are optional; for example, either I believe that a > board should either provide DCVDD, or provide LDO1_IN and assert the > LDO1_EN input signal. Similarly, I think that either MICVDD should be > provided, or SPKVDDL be provided, coupled with LDO2_EN register bit > being set. This may require the RT5640 driver to provide two regulator > objects (LDO1, LDO2), which the board file or device tree can connect > back to the DCVDD and MICVDD inputs if appropriate for the board's HW > configuration. In the case where LDO1 is used, a separate fixed > regulator with GPIO should be used to control the LDO1_EN GPIO input. That said if boards generally don't use external supplies and use the built in regulators then it's probably best to at least have the driver assume that by default. > > Bard: You are right. The mic type should be defined according to circuit. > > And it should be change in run time. > > So, should we just remove this control in codec driver? > Mark's point is that the driver should either/both accept some platform > data structure, or parse device tree, to configure the input pins. I > don't think completely removing all mechanisms to configure this would > be a good idea, since then how would the board/device-tree author > configure the CODEC? Indeed, there definitely should be configuration - however what I was commenting on there was a user-visible ALSA control which should definitely go.

3 6

Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Grant Likely 17 Apr '13

17 Apr '13

On Tue, 16 Apr 2013 19:27:34 +0100, David Howells <dhowells(a)redhat.com> wrote: > Supply a function (proc_remove()) to remove a proc entry (and any subtree > rooted there) by proc_dir_entry pointer rather than by name and (optionally) > root dir entry pointer. This allows us to eliminate all remaining pde->name > accesses outside of procfs. > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linux-acpi(a)vger.kernel.org > cc: openipmi-developer(a)lists.sourceforge.net > cc: devicetree-discuss(a)lists.ozlabs.org > cc: linux-pci(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org Looks okay to me. For the OF portion: Acked-by: Grant Likely <grant.likely(a)linaro.or> > --- > > drivers/acpi/sbs.c | 21 ++++----------------- > drivers/char/ipmi/ipmi_msghandler.c | 2 +- > drivers/misc/sgi-gru/gruprocfs.c | 2 +- > drivers/of/base.c | 11 +---------- > drivers/pci/proc.c | 12 +++--------- > fs/proc/generic.c | 7 +++++++ > fs/proc/vmcore.c | 2 +- > include/linux/proc_fs.h | 2 ++ > kernel/irq/proc.c | 6 +----- > net/8021q/vlanproc.c | 9 ++------- > net/core/pktgen.c | 6 ++---- > net/ipv4/netfilter/ipt_CLUSTERIP.c | 4 ++-- > net/ipv6/proc.c | 3 +-- > net/netfilter/xt_hashlimit.c | 4 ++-- > sound/core/info.c | 19 +++++-------------- > 15 files changed, 35 insertions(+), 75 deletions(-) > > diff --git a/drivers/acpi/sbs.c b/drivers/acpi/sbs.c > index a296e08..b6241ee 100644 > --- a/drivers/acpi/sbs.c > +++ b/drivers/acpi/sbs.c > @@ -521,19 +521,6 @@ acpi_sbs_add_fs(struct proc_dir_entry **dir, > return 0; > } > > -static void > -acpi_sbs_remove_fs(struct proc_dir_entry **dir, > - struct proc_dir_entry *parent_dir) > -{ > - if (*dir) { > - remove_proc_entry(ACPI_SBS_FILE_INFO, *dir); > - remove_proc_entry(ACPI_SBS_FILE_STATE, *dir); > - remove_proc_entry(ACPI_SBS_FILE_ALARM, *dir); > - remove_proc_entry((*dir)->name, parent_dir); > - *dir = NULL; > - } > -} > - > /* Smart Battery Interface */ > static struct proc_dir_entry *acpi_battery_dir = NULL; > > @@ -836,8 +823,8 @@ static void acpi_battery_remove(struct acpi_sbs *sbs, int id) > power_supply_unregister(&battery->bat); > } > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (battery->proc_entry) > - acpi_sbs_remove_fs(&battery->proc_entry, acpi_battery_dir); > + proc_remove(battery->proc_entry); > + battery->proc_entry = NULL; > #endif > } > > @@ -873,8 +860,8 @@ static void acpi_charger_remove(struct acpi_sbs *sbs) > if (sbs->charger.dev) > power_supply_unregister(&sbs->charger); > #ifdef CONFIG_ACPI_PROCFS_POWER > - if (sbs->charger_entry) > - acpi_sbs_remove_fs(&sbs->charger_entry, acpi_ac_dir); > + proc_remove(sbs->charger_entry); > + sbs->charger_entry = NULL; > #endif > } > > diff --git a/drivers/char/ipmi/ipmi_msghandler.c b/drivers/char/ipmi/ipmi_msghandler.c > index 1420bbb..4d439d2 100644 > --- a/drivers/char/ipmi/ipmi_msghandler.c > +++ b/drivers/char/ipmi/ipmi_msghandler.c > @@ -4541,7 +4541,7 @@ static void __exit cleanup_ipmi(void) > del_timer_sync(&ipmi_timer); > > #ifdef CONFIG_PROC_FS > - remove_proc_entry(proc_ipmi_root->name, NULL); > + proc_remove(proc_ipmi_root); > #endif /* CONFIG_PROC_FS */ > > driver_unregister(&ipmidriver.driver); > diff --git a/drivers/misc/sgi-gru/gruprocfs.c b/drivers/misc/sgi-gru/gruprocfs.c > index 950dbe9..797d796 100644 > --- a/drivers/misc/sgi-gru/gruprocfs.c > +++ b/drivers/misc/sgi-gru/gruprocfs.c > @@ -355,7 +355,7 @@ static void delete_proc_files(void) > for (p = proc_files; p->name; p++) > if (p->entry) > remove_proc_entry(p->name, proc_gru); > - remove_proc_entry("gru", proc_gru->parent); > + proc_remove(proc_gru); > } > } > > diff --git a/drivers/of/base.c b/drivers/of/base.c > index 321d3ef..9c70436 100644 > --- a/drivers/of/base.c > +++ b/drivers/of/base.c > @@ -1452,16 +1452,7 @@ int of_attach_node(struct device_node *np) > #ifdef CONFIG_PROC_DEVICETREE > static void of_remove_proc_dt_entry(struct device_node *dn) > { > - struct device_node *parent = dn->parent; > - struct property *prop = dn->properties; > - > - while (prop) { > - remove_proc_entry(prop->name, dn->pde); > - prop = prop->next; > - } > - > - if (dn->pde) > - remove_proc_entry(dn->pde->name, parent->pde); > + proc_remove(dn->pde); > } > #else > static void of_remove_proc_dt_entry(struct device_node *dn) > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 7cde7c1..0812608 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -427,20 +427,14 @@ int pci_proc_attach_device(struct pci_dev *dev) > > int pci_proc_detach_device(struct pci_dev *dev) > { > - struct proc_dir_entry *e; > - > - if ((e = dev->procent)) { > - remove_proc_entry(e->name, dev->bus->procdir); > - dev->procent = NULL; > - } > + proc_remove(dev->procent); > + dev->procent = NULL; > return 0; > } > > int pci_proc_detach_bus(struct pci_bus* bus) > { > - struct proc_dir_entry *de = bus->procdir; > - if (de) > - remove_proc_entry(de->name, proc_bus_pci_dir); > + proc_remove(bus->procdir); > return 0; > } > > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 2c6d6be..4b48033 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -630,3 +630,10 @@ const char *get_proc_name(const struct proc_dir_entry *de) > return de->name; > } > EXPORT_SYMBOL_GPL(get_proc_name); > + > +void proc_remove(struct proc_dir_entry *de) > +{ > + if (de) > + remove_proc_subtree(de->name, de->parent); > +} > +EXPORT_SYMBOL(proc_remove); > diff --git a/fs/proc/vmcore.c b/fs/proc/vmcore.c > index 38edddc..17f7e08 100644 > --- a/fs/proc/vmcore.c > +++ b/fs/proc/vmcore.c > @@ -699,7 +699,7 @@ void vmcore_cleanup(void) > struct list_head *pos, *next; > > if (proc_vmcore) { > - remove_proc_entry(proc_vmcore->name, proc_vmcore->parent); > + proc_remove(proc_vmcore); > proc_vmcore = NULL; > } > > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 7abc72a..b8949b1 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -67,6 +67,7 @@ struct proc_dir_entry *proc_create_data(const char *name, umode_t mode, > struct proc_dir_entry *parent, > const struct file_operations *proc_fops, > void *data); > +extern void proc_remove(struct proc_dir_entry *); > extern void remove_proc_entry(const char *name, struct proc_dir_entry *parent); > extern int remove_proc_subtree(const char *name, struct proc_dir_entry *parent); > > @@ -103,6 +104,7 @@ static inline struct proc_dir_entry *proc_create_data(const char *name, > { > return NULL; > } > +static inline void proc_remove(struct proc_dir_entry *de) {} > #define remove_proc_entry(name, parent) do {} while (0) > #define remove_proc_subtree(name, parent) do {} while (0) > > diff --git a/kernel/irq/proc.c b/kernel/irq/proc.c > index d59ae37..19ed5c4 100644 > --- a/kernel/irq/proc.c > +++ b/kernel/irq/proc.c > @@ -366,11 +366,7 @@ void unregister_irq_proc(unsigned int irq, struct irq_desc *desc) > > void unregister_handler_proc(unsigned int irq, struct irqaction *action) > { > - if (action->dir) { > - struct irq_desc *desc = irq_to_desc(irq); > - > - remove_proc_entry(action->dir->name, desc->dir); > - } > + proc_remove(action->dir); > } > > static void register_default_affinity_proc(void) > diff --git a/net/8021q/vlanproc.c b/net/8021q/vlanproc.c > index 959ddbb..1d0e8921 100644 > --- a/net/8021q/vlanproc.c > +++ b/net/8021q/vlanproc.c > @@ -184,14 +184,9 @@ int vlan_proc_add_dev(struct net_device *vlandev) > */ > int vlan_proc_rem_dev(struct net_device *vlandev) > { > - struct vlan_net *vn = net_generic(dev_net(vlandev), vlan_net_id); > - > /** NOTE: This will consume the memory pointed to by dent, it seems. */ > - if (vlan_dev_priv(vlandev)->dent) { > - remove_proc_entry(vlan_dev_priv(vlandev)->dent->name, > - vn->proc_vlan_dir); > - vlan_dev_priv(vlandev)->dent = NULL; > - } > + proc_remove(vlan_dev_priv(vlandev)->dent); > + vlan_dev_priv(vlandev)->dent = NULL; > return 0; > } > > diff --git a/net/core/pktgen.c b/net/core/pktgen.c > index f6af4fe..6c41e97 100644 > --- a/net/core/pktgen.c > +++ b/net/core/pktgen.c > @@ -1904,7 +1904,7 @@ static void pktgen_change_name(const struct pktgen_net *pn, struct net_device *d > if (pkt_dev->odev != dev) > continue; > > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > pkt_dev->entry = proc_create_data(dev->name, 0600, > pn->proc_dir, > @@ -3576,8 +3576,6 @@ static void _rem_dev_from_if_list(struct pktgen_thread *t, > static int pktgen_remove_device(struct pktgen_thread *t, > struct pktgen_dev *pkt_dev) > { > - struct pktgen_net *pn = t->net; > - > pr_debug("remove_device pkt_dev=%p\n", pkt_dev); > > if (pkt_dev->running) { > @@ -3597,7 +3595,7 @@ static int pktgen_remove_device(struct pktgen_thread *t, > _rem_dev_from_if_list(t, pkt_dev); > > if (pkt_dev->entry) > - remove_proc_entry(pkt_dev->entry->name, pn->proc_dir); > + proc_remove(pkt_dev->entry); > > #ifdef CONFIG_XFRM > free_SAs(pkt_dev); > diff --git a/net/ipv4/netfilter/ipt_CLUSTERIP.c b/net/ipv4/netfilter/ipt_CLUSTERIP.c > index e4738fe..0b732ef 100644 > --- a/net/ipv4/netfilter/ipt_CLUSTERIP.c > +++ b/net/ipv4/netfilter/ipt_CLUSTERIP.c > @@ -105,7 +105,7 @@ clusterip_config_entry_put(struct clusterip_config *c) > * functions are also incrementing the refcount on their own, > * so it's safe to remove the entry even if it's in use. */ > #ifdef CONFIG_PROC_FS > - remove_proc_entry(c->pde->name, c->pde->parent); > + proc_remove(c->pde); > #endif > return; > } > @@ -736,7 +736,7 @@ static void __exit clusterip_tg_exit(void) > { > pr_info("ClusterIP Version %s unloading\n", CLUSTERIP_VERSION); > #ifdef CONFIG_PROC_FS > - remove_proc_entry(clusterip_procdir->name, clusterip_procdir->parent); > + proc_remove(clusterip_procdir); > #endif > nf_unregister_hook(&cip_arp_ops); > xt_unregister_target(&clusterip_tg_reg); > diff --git a/net/ipv6/proc.c b/net/ipv6/proc.c > index 7ea6e18..537d9ee 100644 > --- a/net/ipv6/proc.c > +++ b/net/ipv6/proc.c > @@ -287,8 +287,7 @@ int snmp6_unregister_dev(struct inet6_dev *idev) > return -ENOENT; > if (!idev->stats.proc_dir_entry) > return -EINVAL; > - remove_proc_entry(idev->stats.proc_dir_entry->name, > - net->mib.proc_net_devsnmp6); > + proc_remove(idev->stats.proc_dir_entry); > idev->stats.proc_dir_entry = NULL; > return 0; > } > diff --git a/net/netfilter/xt_hashlimit.c b/net/netfilter/xt_hashlimit.c > index d2fd53b..40002ef 100644 > --- a/net/netfilter/xt_hashlimit.c > +++ b/net/netfilter/xt_hashlimit.c > @@ -330,7 +330,7 @@ static void htable_destroy(struct xt_hashlimit_htable *hinfo) > parent = hashlimit_net->ip6t_hashlimit; > > if(parent != NULL) > - remove_proc_entry(hinfo->pde->name, parent); > + proc_remove(hinfo->pde); > > htable_selective_cleanup(hinfo, select_all); > vfree(hinfo); > @@ -887,7 +887,7 @@ static void __net_exit hashlimit_proc_net_exit(struct net *net) > pde = hashlimit_net->ip6t_hashlimit; > > hlist_for_each_entry(hinfo, &hashlimit_net->htables, node) > - remove_proc_entry(hinfo->pde->name, pde); > + proc_remove(hinfo->pde); > > hashlimit_net->ipt_hashlimit = NULL; > hashlimit_net->ip6t_hashlimit = NULL; > diff --git a/sound/core/info.c b/sound/core/info.c > index c7f41c3..3c9bd6b 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -153,13 +153,6 @@ EXPORT_SYMBOL(snd_seq_root); > struct snd_info_entry *snd_oss_root; > #endif > > -static void snd_remove_proc_entry(struct proc_dir_entry *parent, > - struct proc_dir_entry *de) > -{ > - if (de) > - remove_proc_entry(de->name, parent); > -} > - > static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) > { > struct snd_info_private_data *data; > @@ -580,7 +573,7 @@ int __exit snd_info_done(void) > #ifdef CONFIG_SND_OSSEMUL > snd_info_free_entry(snd_oss_root); > #endif > - snd_remove_proc_entry(NULL, snd_proc_root); > + proc_remove(snd_proc_root); > } > return 0; > } > @@ -642,7 +635,7 @@ void snd_info_card_id_change(struct snd_card *card) > { > mutex_lock(&info_mutex); > if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > + proc_remove(card->proc_root_link); > card->proc_root_link = NULL; > } > if (strcmp(card->id, card->proc_root->name)) > @@ -661,10 +654,8 @@ void snd_info_card_disconnect(struct snd_card *card) > if (!card) > return; > mutex_lock(&info_mutex); > - if (card->proc_root_link) { > - snd_remove_proc_entry(snd_proc_root, card->proc_root_link); > - card->proc_root_link = NULL; > - } > + proc_remove(card->proc_root_link); > + card->proc_root_link = NULL; > if (card->proc_root) > snd_info_disconnect(card->proc_root); > mutex_unlock(&info_mutex); > @@ -856,7 +847,7 @@ static void snd_info_disconnect(struct snd_info_entry *entry) > list_del_init(&entry->list); > root = entry->parent == NULL ? snd_proc_root : entry->parent->p; > snd_BUG_ON(!root); > - snd_remove_proc_entry(root, entry->p); > + proc_remove(entry->p); > entry->p = NULL; > } > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -- Grant Likely, B.Sc, P.Eng. Secret Lab Technologies, Ltd.

1 0

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 17 Apr '13

17 Apr '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 435 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 431 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..080d05f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -215,6 +215,36 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -334,6 +364,175 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,7 +746,140 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ret = wmfw_add_ctl(dsp->wm_coeff, ctl); + if (ret < 0) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -730,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1108,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1125,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1142,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,6 +1363,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -1023,7 +1401,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1031,6 +1409,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1435,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -1099,6 +1492,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1164,7 +1558,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1172,6 +1566,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1613,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1247,36 +1656,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea5146..6e890b9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -55,6 +56,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.1

2 1

[alsa-devel] [PATCH 0/7] Add capture support to compress API
by Charles Keepax 17 Apr '13

17 Apr '13

This chain of patches does some clearing up and add capture support to the compressed API. Missing is support for capture streams that don't use the copy callback. Also I have included two patches at the end of the chain that remove the buffer pointers and just use the cumulative values however as our setup uses the copy callback these are untested past build so should be taken with caution. Thanks, Charles Charles Keepax (7): ALSA: compress_core: Update calc_avail to use cumulative values ALSA: compress_core: Calculate avail correctly for capture streams ALSA: compress_core: Deconstify copy callback buffer ALSA: compress_core: Add support for capture streams ASoC: soc-compress: Deduce stream direction ALSA: compress_core: Remove unused hw_pointer ALSA: compress_core: Rework writes to use cumulative values include/sound/compress_driver.h | 4 +- sound/core/compress_offload.c | 117 +++++++++++++++++++++++++------------- sound/soc/soc-compress.c | 11 +++- 3 files changed, 87 insertions(+), 45 deletions(-) -- 1.7.2.5

4 10

Re: [alsa-devel] [RFC] sound: ASoC: multiple DAIs/codecs on single audio stream
by Mark Brown 17 Apr '13

17 Apr '13

On Tue, Apr 16, 2013 at 07:45:08PM +0200, Sebastian Hesselbarth wrote: > The controller has a dma controller for one playback buffer. It can > play the stream to both I2S and SPDIF interfaces with independent mute. > Now, for example, I have a board with i2s codec and spdif tx diode > connected to the above controller: > DMA (platform) -+-> I2S (dai) ----> I2S_Codec (codec) > | > +-> SPDIF (dai) --> SPDIF_TX (codec) > Is there any way in the current ASoC API to have the above setup? As discussed on IRC this is probably best represented using soc-pcm at the minute.

1 0

[alsa-devel] [PATCH v3 00/15] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 17 Apr '13

17 Apr '13

Hi, This series adds DT support for phycore-ac97. It is now based on the pcm dma cleanups from Lars. Beside ac97 support, the series adds fsl-ssi imx-pcm-fiq and generic DMA binding handling. @Lars: There are 4 new patches at the beginning of the series that fix some problems of your cleanups. I think they were in the changes you posted. If not, could you please add them to your branch? Regards, Markus Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (15): ASoC: imx-ssi: Fix includes ASoC: dmaengine-pcm: Replace of specific helper function ASoC: dmaengine-pcm: Fix function name ASoC: dmaengine-pcm: Fix pcm_request_chan function name ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add SACNT definitions ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 7 + Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 12 ++ arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- include/sound/dmaengine_pcm.h | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/fsl/fsl_ssi.c | 484 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi.h | 8 ++ sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 16 ++- sound/soc/fsl/imx-pcm.h | 9 ++ sound/soc/fsl/imx-ssi.c | 7 +- sound/soc/fsl/imx-ssi.h | 2 + sound/soc/fsl/phycore-ac97.c | 184 +++++++++++++++++++++++---- sound/soc/soc-generic-dmaengine-pcm.c | 9 +- sound/soc/spear/spear_pcm.c | 2 +- 16 files changed, 634 insertions(+), 128 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (92%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

4 22

[alsa-devel] Fwd: Solved Hercules RMX2
by Daniel Schürmann 17 Apr '13

17 Apr '13

Hi Daniel, thank your advice! I am not a Hercules developer, I am developer of the Mixxx DJ Software and I am just trying to make the thing work! I have just read within the USB 2.0 speck to find out if there are specific timeouts to deal with. I have found: § 9.2.6.4 Standard Device Requests If I understand it correct, there is a 5-second limit for storing data into the device. I think changing the sample rate is such a operation. So what about changing the Limit to 5000 ms for all "Set requests" bmRequest = 21. See Audio20 § 5.2.2 All other timeouts might be reduced to 500 ms but I would leave them untouched at 1000 ms , just in case. I would prefer this general solution over a device specific one. What do you think? Kind regards, Daniel 2013/4/16 Daniel Mack <zonque(a)gmail.com> > On 16.04.2013 21:23, Daniel Mack wrote: > > On 16.04.2013 21:14, Daniel Schürmann wrote: > > >> Against which Kernel should I patch? > > > > > https://git.kernel.org/cgit/linux/kernel/git/broonie/sound.git/log/?h=for-n… > > Eh, sorry. This one: > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/ for-next > >

2 1

[alsa-devel] [PATCH] ASoC: max98088: Fix logging of hardware revision.
by Dylan Reid 17 Apr '13

17 Apr '13

The hardware revision of the codec is based at 0x40. Subtract that before convering to ASCII. The same as it is done for 98095. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3a7b7fd..3eeada5 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2024,7 +2024,7 @@ static int max98088_probe(struct snd_soc_codec *codec) ret); goto err_access; } - dev_info(codec->dev, "revision %c\n", ret + 'A'); + dev_info(codec->dev, "revision %c\n", ret - 0x40 + 'A'); snd_soc_write(codec, M98088_REG_51_PWR_SYS, M98088_PWRSV); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] ASoC: define playback and capture streams in dummy codec
by Stas Sergeev 17 Apr '13

17 Apr '13

Hi. This patch adds a playback and capture streams to the dummy codec DAI configuration. Most permissive set of sampling rates and formats is used. Signed-off-by: Stas Sergeev <stsp(a)users.sourceforge.net>

5 12

Re: [alsa-devel] hda: sink info in case of AMD Radeon GPU card (no ELD?)
by Rafał Miłecki 17 Apr '13

17 Apr '13

2013/1/19 Rafał Miłecki <zajec5(a)gmail.com>: > Hello, > > I'm interested in polishing HDMI audio support for Radeon cards and I > hope to get some help here. One of the current problems is that hda > has no information about TV audio capabilities. It results in a well > known: > ALSA sound/pci/hda/hda_eld.c:337 HDMI: ELD buf size is 0, force 128 > ALSA sound/pci/hda/hda_eld.c:356 HDMI: invalid ELD data byte 0 > It's expected, as radeon (GPU driver) doesn't pass ELD struct anywhere > to the GPU and it's not done automatically neither. > > Unfortunately it doesn't seem to be trivial to solve. You may be > familiar with other GPUs like Nvidia or Intel, where it's done in a > pretty simple way, for example: >> for (i = 0; i < len; i++) >> I915_WRITE(hdmiw_hdmiedid, *((uint32_t *)eld + i)); > but it can't be handled that way on Radeon cards. > > If you take a look at evergreend.h [0] you can see we can't simply > write ELD to the GPU buffer, but we have to fill specific registers > which look more like SADs (Short Audio Descriptors). > > My question is how to handle this on the alsa side. We can't just read > ELD, we probably have to read some registers from the audio card. Is > something like that implemented anywhere else? Did anyone started > working on it? Anyone interested in it? > > [0] http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob;f=drive… I tried hacking this sink info problem again. I've installed "fglrx" on my AMD Radeon HD 6970M (BARTS == DCE5) and verified it sets SAD registers: 5f84 7f077f07 5f88 00500705 5f8c 00000001 5f90 00000001 5f94 00000001 5f98 00000001 5f9c 00c01e05 5fa0 00000001 5fa4 00000001 5fa8 00000607 5fac 00017e07 5fb0 00007e07 5fb4 00000001 5fb8 00000001 Reading ELD of course doesn't work (that's why we get warning in dmesg) and I did some manual confirmation with hda-verb (reading ELD length): $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2e 0x8 nid = 0x3, verb = 0xf2e, param = 0x8 value = 0x0 Plus (just for sure) reading first byte of ELD (in case length is always zero, but data is available): $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 0 | grep val value = 0x0 $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 1 | grep val value = 0x0 $ sudo ./hda-verb /dev/snd/hwC1D0 3 0xf2f 2 | grep val value = 0x0 So it seems ELD is really not available in this configuration. I thought about vendor specific verbs. Maybe values from the GPU registers are just copied to the audio? I've tried reading every possible vendor specific verb with the attached script (verbs.php). Unfortunately any verb didn't return anything interesting :( > grep value verbs.txt | sort | uniq -c 9336832 value = 0x0 18432 value = 0x1 6144 value = 0x40 2048 value = 0x415f 55296 value = 0x86 2048 value = 0xa6 16384 value = 0xb6 Does anyone have any idea how we can/should read that values from GPU in the audio driver? :| I guess the must be copied somewhere, but can't find the correct place. -- Rafał

1 0

Re: [alsa-devel] Enhance support for SigmaDSP chips
by Lars-Peter Clausen 17 Apr '13

17 Apr '13

On 04/11/2013 11:56 AM, Daniel Mack wrote: > Hi Lars, > Hi Cliff, > > I'm looking into the SigmaDSP AD1701 support, as a new prospective > project will use it. What's going to be used on this chip is not the > DAC/ADC features though, but the internal, programmable DSP things, and > designers will use the SigmaStudio IDE to generate the firmware. > > So I'm wondering how to support this kind of chip properly in ALSA, > which is not straight-forward, as the controls exported by it are > specific to the loaded firmware of course. Also, the IDE is free to > re-allocate every sub-addresses of its controls when something changes > in the layout of the building blocks. > > One idea I have in mind is to have a little parser script that reads the > generated sources of the IDE and dump a device-tree node snippet which > can then be put into the final DT. The driver would need to learn about > how to interpret that DT nodes, which will most likely just contain a > name, a sub-address, along with some mask and shift values. > > I wanted to ask for your opinion on that before I start. Have you ever > used any of the DSP features from ALSA/ASoC? Are there any pitfalls I > should be aware of? Is anyone actively working on improvements on the > driver? > > An alternative approach is to write a userspace library of course, but > that's going to be problematic once anybody wants to use the DSP > features in parallel to the DAC/ADC. Hi, I just yesterday wrote a small post how you can program the DSP registers. http://ez.analog.com/message/83094#83094 I'd like to be able to put the information about the different algorithms into the DSP firmware file itself, so we can auto instantiate the controls from the driver. The problem is that it is not so easy to generalize the export of the algorithms from SigmaStudio. Another problem is that you don't know how to calculate the algorithm parameters at runtime for some of the more complex algorithms. I don't think it's a good idea to put the control info into the devicetree and would rather prefer to see them go into the firmware. But as I said auto-generating this from exported SigmaStudio files is not so easy. So maybe manually creating the control info table might be an option. Btw. in case you haven't seen it the SigmaTcp tools is quite useful during the development of the firmware since it allows you to connect SigmaStudio directly to the DSP on the board. http://wiki.analog.com/resources/tools-software/linux-software/sigmatcp - Lars

3 4

[alsa-devel] [PATCH] ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls
by Dimitris Papastamos 17 Apr '13

17 Apr '13

Add initial support for runtime tuning for the ADSP cores. This is achieved by exposing the coefficient configuration blocks as ALSA binary controls. The current code assumes that no controls on the DSP are volatile. Change-Id: I9cd57efabe6c195c731f5a17f53617a99d2f09da Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 435 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 3 + 2 files changed, 431 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3470b64..080d05f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -215,6 +215,36 @@ static struct { [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; +struct wm_coeff_ctl_ops { + int (*xget)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xput)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*xinfo)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +struct wm_coeff { + struct device *dev; + struct list_head ctl_list; + struct regmap *regmap; +}; + +struct wm_coeff_ctl { + const char *name; + struct snd_card *card; + struct wm_adsp_alg_region region; + struct wm_coeff_ctl_ops ops; + struct wm_adsp *adsp; + void *private; + unsigned int enabled:1; + struct list_head list; + void *cache; + size_t len; + unsigned int dirty:1; + struct snd_kcontrol *kcontrol; +}; + static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -334,6 +364,175 @@ static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, } } +static int wm_coeff_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = ctl->len; + return 0; +} + +static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, + const void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmemdup(buf, ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ctl->len); + if (ret) { + adsp_err(adsp, "Failed to write %zu bytes to %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + kfree(scratch); + + return 0; +} + +static int wm_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(ctl->cache, p, ctl->len); + + if (!ctl->enabled) { + ctl->dirty = 1; + return 0; + } + + return wm_coeff_write_control(kcontrol, p, ctl->len); +} + +static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, + void *buf, size_t len) +{ + struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_adsp_alg_region *region = &ctl->region; + const struct wm_adsp_region *mem; + struct wm_adsp *adsp = ctl->adsp; + void *scratch; + int ret; + unsigned int reg; + + mem = wm_adsp_find_region(adsp, region->type); + if (!mem) { + adsp_err(adsp, "No base for region %x\n", + region->type); + return -EINVAL; + } + + reg = ctl->region.base; + reg = wm_adsp_region_to_reg(mem, reg); + + scratch = kmalloc(ctl->len, GFP_KERNEL | GFP_DMA); + if (!scratch) + return -ENOMEM; + + ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + if (ret) { + adsp_err(adsp, "Failed to read %zu bytes from %x\n", + ctl->len, reg); + kfree(scratch); + return ret; + } + + memcpy(buf, scratch, ctl->len); + kfree(scratch); + + return 0; +} + +static int wm_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + char *p = ucontrol->value.bytes.data; + + memcpy(p, ctl->cache, ctl->len); + return 0; +} + +static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl, + const struct snd_kcontrol_new *kctl) +{ + int ret; + struct snd_kcontrol *kcontrol; + + kcontrol = snd_ctl_new1(kctl, wm_coeff); + ret = snd_ctl_add(ctl->card, kcontrol); + if (ret < 0) { + dev_err(wm_coeff->dev, "Failed to add %s: %d\n", + kctl->name, ret); + return ret; + } + ctl->kcontrol = kcontrol; + return 0; +} + +static int wmfw_add_ctl(struct wm_coeff *wm_coeff, + struct wm_coeff_ctl *ctl) +{ + struct snd_kcontrol_new *kcontrol; + int ret; + + if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + return -EINVAL; + + kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); + if (!kcontrol) + return -ENOMEM; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + kcontrol->name = ctl->name; + kcontrol->info = wm_coeff_info; + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + kcontrol->private_value = (unsigned long)ctl; + + ret = wm_coeff_add_kcontrol(wm_coeff, + ctl, kcontrol); + if (ret < 0) + goto err_kcontrol; + + kfree(kcontrol); + + list_add(&ctl->list, &wm_coeff->ctl_list); + return 0; + +err_kcontrol: + kfree(kcontrol); + return ret; +} + static int wm_adsp_load(struct wm_adsp *dsp) { LIST_HEAD(buf_list); @@ -547,7 +746,140 @@ out: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp) +static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled || ctl->dirty) + continue; + ret = wm_coeff_read_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + } + + return 0; +} + +static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +{ + struct wm_coeff_ctl *ctl; + int ret; + + list_for_each_entry(ctl, &wm_coeff->ctl_list, + list) { + if (!ctl->enabled) + continue; + if (ctl->dirty) { + ret = wm_coeff_write_control(ctl->kcontrol, + ctl->cache, + ctl->len); + if (ret < 0) + return ret; + ctl->dirty = 0; + } + } + + return 0; +} + +static int wm_adsp_create_control(struct snd_soc_codec *codec, + const struct wm_adsp_alg_region *region) + +{ + struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); + struct wm_coeff_ctl *ctl; + char *name; + char *region_name; + int ret; + + name = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!name) + return -ENOMEM; + + switch (region->type) { + case WMFW_ADSP1_PM: + region_name = "PM"; + break; + case WMFW_ADSP1_DM: + region_name = "DM"; + break; + case WMFW_ADSP2_XM: + region_name = "XM"; + break; + case WMFW_ADSP2_YM: + region_name = "YM"; + break; + case WMFW_ADSP1_ZM: + region_name = "ZM"; + break; + default: + return -EINVAL; + } + + snprintf(name, PAGE_SIZE, "DSP%d %s %x", + dsp->num, region_name, region->alg); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + if (!strcmp(ctl->name, name)) { + if (!ctl->enabled) + ctl->enabled = 1; + return 0; + } + } + + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) { + ret = -ENOMEM; + goto err_name; + } + ctl->region = *region; + ctl->name = kmemdup(name, strlen(name) + 1, GFP_KERNEL); + if (!ctl->name) { + ret = -ENOMEM; + goto err_ctl; + } + ctl->enabled = 1; + ctl->dirty = 0; + ctl->ops.xget = wm_coeff_get; + ctl->ops.xput = wm_coeff_put; + ctl->card = codec->card->snd_card; + ctl->adsp = dsp; + + ctl->len = region->len; + ctl->cache = kzalloc(ctl->len, GFP_KERNEL); + if (!ctl->cache) { + ret = -ENOMEM; + goto err_ctl_name; + } + + ret = wmfw_add_ctl(dsp->wm_coeff, ctl); + if (ret < 0) { + ret = -ENOMEM; + goto err_ctl_cache; + } + + kfree(name); + + return 0; + +err_ctl_cache: + kfree(ctl->cache); +err_ctl_name: + kfree(ctl->name); +err_ctl: + kfree(ctl); +err_name: + kfree(name); + return ret; +} + +static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -730,7 +1062,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].dm); + region->len -= be32_to_cpu(adsp1_alg[i].dm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region DM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -738,7 +1079,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp1_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp1_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp1_alg[i].alg.id)); + } break; case WMFW_ADSP2: @@ -758,7 +1108,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].xm); + region->len -= be32_to_cpu(adsp2_alg[i].xm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region XM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -766,7 +1125,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].ym); + region->len -= be32_to_cpu(adsp2_alg[i].ym); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region YM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } region = kzalloc(sizeof(*region), GFP_KERNEL); if (!region) @@ -774,7 +1142,16 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); + region->len = 0; list_add_tail(&region->list, &dsp->alg_regions); + if (i + 1 < algs) { + region->len = be32_to_cpu(adsp2_alg[i + 1].zm); + region->len -= be32_to_cpu(adsp2_alg[i].zm); + wm_adsp_create_control(codec, region); + } else { + adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", + be32_to_cpu(adsp2_alg[i].alg.id)); + } break; } } @@ -986,6 +1363,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_coeff_ctl *ctl; int ret; int val; @@ -1023,7 +1401,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1031,6 +1409,16 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + /* Start the core running */ regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_CORE_ENA | ADSP1_START, @@ -1047,6 +1435,11 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); + + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } break; default: @@ -1099,6 +1492,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; struct wm_adsp_alg_region *alg_region; + struct wm_coeff_ctl *ctl; unsigned int val; int ret; @@ -1164,7 +1558,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp); + ret = wm_adsp_setup_algs(dsp, codec); if (ret != 0) goto err; @@ -1172,6 +1566,16 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + /* Initialize caches for enabled and non-dirty controls */ + ret = wm_coeff_init_control_caches(dsp->wm_coeff); + if (ret != 0) + goto err; + + /* Sync dirty controls */ + ret = wm_coeff_sync_controls(dsp->wm_coeff); + if (ret != 0) + goto err; + ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, ADSP2_CORE_ENA | ADSP2_START, @@ -1209,6 +1613,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } + list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list) { + ctl->enabled = 0; + } + while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, struct wm_adsp_alg_region, @@ -1247,36 +1656,48 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) INIT_LIST_HEAD(&adsp->alg_regions); + adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), + GFP_KERNEL); + if (!adsp->wm_coeff) + return -ENOMEM; + adsp->wm_coeff->regmap = adsp->regmap; + adsp->wm_coeff->dev = adsp->dev; + INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - return ret; + goto out_coeff; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - return ret; + goto out_coeff; } } return 0; + +out_coeff: + kfree(adsp->wm_coeff); + return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index fea5146..6e890b9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -30,6 +30,7 @@ struct wm_adsp_alg_region { unsigned int alg; int type; unsigned int base; + size_t len; }; struct wm_adsp { @@ -55,6 +56,8 @@ struct wm_adsp { bool running; struct regulator *dvfs; + + struct wm_coeff *wm_coeff; }; #define WM_ADSP1(wname, num) \ -- 1.8.2.1

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[alsa-devel] Can Realtek HD-A codec ALC282 support D3cold?
by Lin, Mengdong 17 Apr '13

17 Apr '13

Hi, (Sorry to post this message again) We want to put an idle HD-A codec into D3cold for more power saving and preparation for being power gated. The Realtek ALC282 is used in our test. Could anyone help to confirm whether ALC282 can support D3cold or not? I'm told that ALC282 can support D3cold. But my test shows it cannot. At first, ALC282 HW does not report that it can support D3cold. Its parameter "Supported Power States" returns 0xc000000f, means it only supports: EPSS, CLKSTOP, D3hot, D2, D1 and D0. Is this a just mistake in reporting the bit, or D3cold is not supported all? Then when I explicitly set the audio function group to D3cold by verb 0x705, and then get its power state after 200ms by verb 0xf05, The codec response is 0x00000000: No PS-Error, both PS-Act and PS-set is D0. So it seems codec HW does not process the set power state command at all. Is there someone from Realtek or familiar with ALC282 here can share some information about this issue? Many thanks Mengdong

2 1

Re: [alsa-devel] [PATCH 04/28] proc: Supply PDE attribute setting accessor functions [RFC]
by Mauro Carvalho Chehab 17 Apr '13

17 Apr '13

Em 16-04-2013 15:26, David Howells escreveu: > Supply accessor functions to set attributes in proc_dir_entry structs. > > The following are supplied: proc_set_size() and proc_set_user(). > > Signed-off-by: David Howells <dhowells(a)redhat.com> > cc: linuxppc-dev(a)lists.ozlabs.org > cc: linux-media(a)vger.kernel.org > cc: netdev(a)vger.kernel.org > cc: linux-wireless(a)vger.kernel.org > cc: linux-pci(a)vger.kernel.org > cc: netfilter-devel(a)vger.kernel.org > cc: alsa-devel(a)alsa-project.org > --- > > arch/powerpc/kernel/proc_powerpc.c | 2 +- > arch/powerpc/platforms/pseries/reconfig.c | 2 +- > drivers/media/pci/ttpci/av7110_ir.c | 2 +- Weird that av7110 IR uses /proc... Well, this is an old, obsolete driver for hardware that is not sold anymore for a long time... So, be it. Acked-by: Mauro Carvalho Chehab <mchehab(a)redhat.com> > drivers/net/irda/vlsi_ir.c | 2 +- > drivers/net/wireless/airo.c | 34 +++++++++-------------------- > drivers/pci/proc.c | 2 +- > fs/proc/generic.c | 13 +++++++++++ > include/linux/proc_fs.h | 5 ++++ > kernel/configs.c | 2 +- > kernel/profile.c | 2 +- > net/netfilter/xt_recent.c | 3 +-- > sound/core/info.c | 2 +- > 12 files changed, 38 insertions(+), 33 deletions(-) > > diff --git a/arch/powerpc/kernel/proc_powerpc.c b/arch/powerpc/kernel/proc_powerpc.c > index 41d8ee9..feb8580 100644 > --- a/arch/powerpc/kernel/proc_powerpc.c > +++ b/arch/powerpc/kernel/proc_powerpc.c > @@ -83,7 +83,7 @@ static int __init proc_ppc64_init(void) > &page_map_fops, vdso_data); > if (!pde) > return 1; > - pde->size = PAGE_SIZE; > + proc_set_size(pde, PAGE_SIZE); > > return 0; > } > diff --git a/arch/powerpc/platforms/pseries/reconfig.c b/arch/powerpc/platforms/pseries/reconfig.c > index d6491bd..f93cdf5 100644 > --- a/arch/powerpc/platforms/pseries/reconfig.c > +++ b/arch/powerpc/platforms/pseries/reconfig.c > @@ -452,7 +452,7 @@ static int proc_ppc64_create_ofdt(void) > > ent = proc_create("powerpc/ofdt", S_IWUSR, NULL, &ofdt_fops); > if (ent) > - ent->size = 0; > + proc_set_size(ent, 0); > > return 0; > } > diff --git a/drivers/media/pci/ttpci/av7110_ir.c b/drivers/media/pci/ttpci/av7110_ir.c > index eb82286..0e763a7 100644 > --- a/drivers/media/pci/ttpci/av7110_ir.c > +++ b/drivers/media/pci/ttpci/av7110_ir.c > @@ -375,7 +375,7 @@ int av7110_ir_init(struct av7110 *av7110) > if (av_cnt == 1) { > e = proc_create("av7110_ir", S_IWUSR, NULL, &av7110_ir_proc_fops); > if (e) > - e->size = 4 + 256 * sizeof(u16); > + proc_set_size(e, 4 + 256 * sizeof(u16)); > } > > tasklet_init(&av7110->ir.ir_tasklet, av7110_emit_key, (unsigned long) &av7110->ir); > diff --git a/drivers/net/irda/vlsi_ir.c b/drivers/net/irda/vlsi_ir.c > index e22cd4e..5f47584 100644 > --- a/drivers/net/irda/vlsi_ir.c > +++ b/drivers/net/irda/vlsi_ir.c > @@ -1678,7 +1678,7 @@ vlsi_irda_probe(struct pci_dev *pdev, const struct pci_device_id *id) > IRDA_WARNING("%s: failed to create proc entry\n", > __func__); > } else { > - ent->size = 0; > + proc_set_size(ent, 0); > } > idev->proc_entry = ent; > } > diff --git a/drivers/net/wireless/airo.c b/drivers/net/wireless/airo.c > index 66e398d..21d0233 100644 > --- a/drivers/net/wireless/airo.c > +++ b/drivers/net/wireless/airo.c > @@ -4507,73 +4507,63 @@ static int setup_proc_entry( struct net_device *dev, > airo_entry); > if (!apriv->proc_entry) > goto fail; > - apriv->proc_entry->uid = proc_kuid; > - apriv->proc_entry->gid = proc_kgid; > + proc_set_user(apriv->proc_entry, proc_kuid, proc_kgid); > > /* Setup the StatsDelta */ > entry = proc_create_data("StatsDelta", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_statsdelta_ops, dev); > if (!entry) > goto fail_stats_delta; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Stats */ > entry = proc_create_data("Stats", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_stats_ops, dev); > if (!entry) > goto fail_stats; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Status */ > entry = proc_create_data("Status", S_IRUGO & proc_perm, > apriv->proc_entry, &proc_status_ops, dev); > if (!entry) > goto fail_status; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the Config */ > entry = proc_create_data("Config", proc_perm, > apriv->proc_entry, &proc_config_ops, dev); > if (!entry) > goto fail_config; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the SSID */ > entry = proc_create_data("SSID", proc_perm, > apriv->proc_entry, &proc_SSID_ops, dev); > if (!entry) > goto fail_ssid; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the APList */ > entry = proc_create_data("APList", proc_perm, > apriv->proc_entry, &proc_APList_ops, dev); > if (!entry) > goto fail_aplist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the BSSList */ > entry = proc_create_data("BSSList", proc_perm, > apriv->proc_entry, &proc_BSSList_ops, dev); > if (!entry) > goto fail_bsslist; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > + proc_set_user(entry, proc_kuid, proc_kgid); > > /* Setup the WepKey */ > entry = proc_create_data("WepKey", proc_perm, > apriv->proc_entry, &proc_wepkey_ops, dev); > if (!entry) > goto fail_wepkey; > - entry->uid = proc_kuid; > - entry->gid = proc_kgid; > - > + proc_set_user(entry, proc_kuid, proc_kgid); > return 0; > > fail_wepkey: > @@ -5695,10 +5685,8 @@ static int __init airo_init_module( void ) > > airo_entry = proc_mkdir_mode("driver/aironet", airo_perm, NULL); > > - if (airo_entry) { > - airo_entry->uid = proc_kuid; > - airo_entry->gid = proc_kgid; > - } > + if (airo_entry) > + proc_set_user(airo_entry, proc_kuid, proc_kgid); > > for (i = 0; i < 4 && io[i] && irq[i]; i++) { > airo_print_info("", "Trying to configure ISA adapter at irq=%d " > diff --git a/drivers/pci/proc.c b/drivers/pci/proc.c > index 12e4fb5..7cde7c1 100644 > --- a/drivers/pci/proc.c > +++ b/drivers/pci/proc.c > @@ -419,7 +419,7 @@ int pci_proc_attach_device(struct pci_dev *dev) > &proc_bus_pci_operations, dev); > if (!e) > return -ENOMEM; > - e->size = dev->cfg_size; > + proc_set_size(e, dev->cfg_size); > dev->procent = e; > > return 0; > diff --git a/fs/proc/generic.c b/fs/proc/generic.c > index 1c07cad..5f6f6c3 100644 > --- a/fs/proc/generic.c > +++ b/fs/proc/generic.c > @@ -498,6 +498,19 @@ out: > return NULL; > } > EXPORT_SYMBOL(proc_create_data); > + > +void proc_set_size(struct proc_dir_entry *de, loff_t size) > +{ > + de->size = size; > +} > +EXPORT_SYMBOL(proc_set_size); > + > +void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) > +{ > + de->uid = uid; > + de->gid = gid; > +} > +EXPORT_SYMBOL(proc_set_user); > > static void free_proc_entry(struct proc_dir_entry *de) > { > diff --git a/include/linux/proc_fs.h b/include/linux/proc_fs.h > index 805edac..28a4d7e 100644 > --- a/include/linux/proc_fs.h > +++ b/include/linux/proc_fs.h > @@ -130,6 +130,9 @@ static inline struct proc_dir_entry *proc_create(const char *name, umode_t mode, > extern struct proc_dir_entry *proc_net_mkdir(struct net *net, const char *name, > struct proc_dir_entry *parent); > > +extern void proc_set_size(struct proc_dir_entry *, loff_t); > +extern void proc_set_user(struct proc_dir_entry *, kuid_t, kgid_t); > + > extern struct file *proc_ns_fget(int fd); > extern bool proc_ns_inode(struct inode *inode); > > @@ -158,6 +161,8 @@ static inline struct proc_dir_entry *proc_mkdir(const char *name, > struct proc_dir_entry *parent) {return NULL;} > static inline struct proc_dir_entry *proc_mkdir_mode(const char *name, > umode_t mode, struct proc_dir_entry *parent) { return NULL; } > +static inline void proc_set_size(struct proc_dir_entry *de, loff_t size) {} > +static inline void proc_set_user(struct proc_dir_entry *de, kuid_t uid, kgid_t gid) {} > > struct tty_driver; > static inline void proc_tty_register_driver(struct tty_driver *driver) {}; > diff --git a/kernel/configs.c b/kernel/configs.c > index 42e8fa0..c18b1f1 100644 > --- a/kernel/configs.c > +++ b/kernel/configs.c > @@ -79,7 +79,7 @@ static int __init ikconfig_init(void) > if (!entry) > return -ENOMEM; > > - entry->size = kernel_config_data_size; > + proc_set_size(entry, kernel_config_data_size); > > return 0; > } > diff --git a/kernel/profile.c b/kernel/profile.c > index 524ce5e..0bf4007 100644 > --- a/kernel/profile.c > +++ b/kernel/profile.c > @@ -600,7 +600,7 @@ int __ref create_proc_profile(void) /* false positive from hotcpu_notifier */ > NULL, &proc_profile_operations); > if (!entry) > return 0; > - entry->size = (1+prof_len) * sizeof(atomic_t); > + proc_set_size(entry, (1 + prof_len) * sizeof(atomic_t)); > hotcpu_notifier(profile_cpu_callback, 0); > return 0; > } > diff --git a/net/netfilter/xt_recent.c b/net/netfilter/xt_recent.c > index 3db2d38..1e657cf 100644 > --- a/net/netfilter/xt_recent.c > +++ b/net/netfilter/xt_recent.c > @@ -401,8 +401,7 @@ static int recent_mt_check(const struct xt_mtchk_param *par, > ret = -ENOMEM; > goto out; > } > - pde->uid = uid; > - pde->gid = gid; > + proc_set_user(pde, uid, gid); > #endif > spin_lock_bh(&recent_lock); > list_add_tail(&t->list, &recent_net->tables); > diff --git a/sound/core/info.c b/sound/core/info.c > index 3aa8864..c7f41c3 100644 > --- a/sound/core/info.c > +++ b/sound/core/info.c > @@ -970,7 +970,7 @@ int snd_info_register(struct snd_info_entry * entry) > mutex_unlock(&info_mutex); > return -ENOMEM; > } > - p->size = entry->size; > + proc_set_size(p, entry->size); > } > entry->p = p; > if (entry->parent) > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html >

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[alsa-devel] [PATCH] tinycompress: cplay: Correct a couple of small message typos
by Richard Fitzgerald 16 Apr '13

16 Apr '13

Change-Id: I2b25d006328f1a9458f232f77c61e509c9b54d3c Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- cplay.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/cplay.c b/cplay.c index 60995a4..d9364f4 100644 --- a/cplay.c +++ b/cplay.c @@ -337,7 +337,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, goto BUF_EXIT; } if (wrote != num_read) { - /* TODO: Buufer pointer needs to be set here */ + /* TODO: Buffer pointer needs to be set here */ fprintf(stderr, "We wrote %d, DSP accepted %d\n", num_read, wrote); } if (verbose) { @@ -348,7 +348,7 @@ void play_samples(char *name, unsigned int card, unsigned int device, } while (num_read > 0); if (verbose) - printf("%s: exit sucess\n", __func__); + printf("%s: exit success\n", __func__); /* issue drain if it supports */ compress_drain(compress); free(buffer); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: fsi: fixup sparse errors
by Kuninori Morimoto 16 Apr '13

16 Apr '13

This patch fixup below sparse errors ${LINUX}/sound/soc/sh/fsi.c:1459:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1634:25: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:1639:17: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2093:9: \ error: incompatible types in conditional expression (different base types) ${LINUX}/sound/soc/sh/fsi.c:2105:9: \ error: incompatible types in conditional expression (different base types) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c724026a..7bc767b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -276,7 +276,7 @@ struct fsi_stream_handler { int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); - void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, + int (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, int enable); }; #define fsi_stream_handler_call(io, func, args...) \ @@ -1189,7 +1189,7 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples); } -static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int enable) { struct fsi_master *master = fsi_get_master(fsi); @@ -1202,6 +1202,8 @@ static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_pio_push_init(struct fsi_priv *fsi, struct fsi_stream *io) @@ -1410,7 +1412,7 @@ static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, +static int fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, int start) { struct fsi_master *master = fsi_get_master(fsi); @@ -1423,6 +1425,8 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); + + return 0; } static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) -- 1.7.9.5

1 0

[alsa-devel] about snd_pcm_writei() parameter
by Zhang wei 16 Apr '13

16 Apr '13

hi all. Is there have some limitation on the parameter size of the interface: snd_pcm_sframes_t snd_pcm_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size)? should be the SIZE as any value ? For example, set the size < period_size, ok? does the parameter depend on the period_size? Thanks a lot. ---- BR zw

1 0

Re: [alsa-devel] [PATCH 01/17] ASoC: dmaengine-pcm: Make requesting the DMA channel at PCM open optional
by Lars-Peter Clausen 16 Apr '13

16 Apr '13

On 04/16/2013 01:28 AM, Ryan Mallon wrote: > On 16/04/13 03:19, Lars-Peter Clausen wrote: >> Refactor the dmaengine PCM library to allow the DMA channel to be requested >> before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA >> channel instead of a filter function and filter parameter as its parameters. >> snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows >> a dmaengine based PCM driver to request its channels before the substream is >> opened. >> >> The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan() >> and snd_dmaengine_pcm_close_release_chan(), which have the same signature and >> behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new >> variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are >> updated to use snd_dmaengine_pcm_open_request_chan() and >> snd_dmaengine_pcm_close_release_chan(). >> >> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > <snip> > >> --- a/sound/soc/cirrus/ep93xx-pcm.c >> +++ b/sound/soc/cirrus/ep93xx-pcm.c >> @@ -69,7 +69,8 @@ static int ep93xx_pcm_open(struct snd_pcm_substream *substream) >> >> snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); >> >> - return snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, >> + return snd_dmaengine_pcm_open_request_chan(substream, >> + ep93xx_pcm_dma_filter, >> snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); > > ep93xx_pcm_dma_filter is now unused and can be removed? It's still used in the same way as before. - Lars

1 0

Re: [alsa-devel] [PATCH 16/17] ASoC: ep93xx: Use generic dmaengine PCM
by Lars-Peter Clausen 16 Apr '13

16 Apr '13

On 04/16/2013 01:48 AM, Ryan Mallon wrote: > On 16/04/13 03:20, Lars-Peter Clausen wrote: >> Use the generic dmaengine PCM driver instead of a custom implementation. >> >> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> >> --- >> sound/soc/cirrus/Kconfig | 2 +- >> sound/soc/cirrus/ep93xx-i2s.c | 4 +- >> sound/soc/cirrus/ep93xx-pcm.c | 138 ++++-------------------------------------- >> 3 files changed, 14 insertions(+), 130 deletions(-) >> >> diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig >> index 88143db..2c20f01 100644 >> --- a/sound/soc/cirrus/Kconfig >> +++ b/sound/soc/cirrus/Kconfig >> @@ -1,7 +1,7 @@ >> config SND_EP93XX_SOC >> tristate "SoC Audio support for the Cirrus Logic EP93xx series" >> depends on ARCH_EP93XX && SND_SOC >> - select SND_SOC_DMAENGINE_PCM >> + select SND_SOC_GENERIC_DMAENGINE_PCM >> help >> Say Y or M if you want to add support for codecs attached to >> the EP93xx I2S or AC97 interfaces. >> diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c >> index 453a89c..ef731e6 100644 >> --- a/sound/soc/cirrus/ep93xx-i2s.c >> +++ b/sound/soc/cirrus/ep93xx-i2s.c >> @@ -140,8 +140,8 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream) >> >> static int ep93xx_i2s_dai_probe(struct snd_soc_dai *dai) >> { >> - dai->playback_dma_data = ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; >> - dai->capute_dma_data = ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE] >> + dai->playback_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_PLAYBACK]; >> + dai->capture_dma_data = &ep93xx_i2s_dma_data[SNDRV_PCM_STREAM_CAPTURE]; > > Why did this change? I can't see a change to ep93xx_i2s_dma_data. Was > this just incorrect in the previous patch? All patches in a series > should be buildable/runnable to assist things like git bisect. > Yes, that part belongs to the previous patch and only ended up here by accident. >> >> return 0; >> } >> diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c >> index 4880326..0e9f56e 100644 >> --- a/sound/soc/cirrus/ep93xx-pcm.c >> +++ b/sound/soc/cirrus/ep93xx-pcm.c >> @@ -14,20 +14,14 @@ >> >> #include <linux/module.h> >> #include <linux/init.h> >> -#include <linux/device.h> >> -#include <linux/slab.h> >> +#include <linux/platform_device.h> >> #include <linux/dmaengine.h> >> -#include <linux/dma-mapping.h> >> >> -#include <sound/core.h> >> #include <sound/pcm.h> >> -#include <sound/pcm_params.h> >> #include <sound/soc.h> >> #include <sound/dmaengine_pcm.h> >> >> #include <linux/platform_data/dma-ep93xx.h> >> -#include <mach/hardware.h> >> -#include <mach/ep93xx-regs.h> > > Awesome. It looks like this header can also be removed from ep93xx-i2s.c > also, which gets us a step closer to making this header private to the > ep93xx soc code in arch/arm. > > ~Ryan >

1 0

[alsa-devel] [PATCH] ALSA: hda - Add codec delay to the capture time stamp.
by Dylan Reid 16 Apr '13

16 Apr '13

For capture, the delay through the codec contributes to the time stamp of the sample recorded at the A to D. Rename the codec time stamp function appropriately. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 63734b5..7b213d5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,7 +1889,7 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } -static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, +static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); @@ -1903,6 +1903,9 @@ static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, codec_nsecs = div_u64(codec_frames * 1000000000LL, substream->runtime->rate); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return nsec + codec_nsecs; + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; } @@ -1914,7 +1917,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ - nsec = azx_subtract_codec_delay(substream, nsec); + nsec = azx_adjust_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] ASoC updates for v3.10
by Mark Brown 15 Apr '13

15 Apr '13

The following changes since commit b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10 for you to fetch changes up to 5cbad7d39ad229c68a724e5e139fd845b93766b2: Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next (2013-04-12 13:57:31 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A bunch of changes here, the most interesting one subsystem wise being Morimoto-san's work to create snd_soc_component which doesn't do much for now but will be pretty important going forwards: - Add a new component object type which will form the basis of moving to a more generic handling of SoC and off-SoC components, contributed by Kuninori Morimoto. - A fairly large set of cleanups for the dmaengine integration from Lars-Peter Clausen, starting to move towards being able to have a generic driver based on the library. - Performance optimisations to DAPM from Ryo Tsutsui. - Support for mixer control sharing in DAPM from Stephen Warren. - Multiplatform ARM cleanups from Arnd Bergmann. - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack. ---------------------------------------------------------------- Alban Bedel (1): ASoC: wm8903: Add the DAC boost control Alexandru Gheorghiu (1): ASoC: core: Use PTR_RET function Andrey Smirnov (2): ASoC: si476x: Convert SI476X codec to use regmap ASoC: si476x: Cosmetic changes to SI476X codec driver Arnd Bergmann (4): ASoC: samsung: use irq resource for idma ASoC: samsung: fix module_device_table ASoC: samsung: export idma_reg_addr_init ASoC: samsung: move plat/ headers to local directory Charles Keepax (12): ASoC: arizona: Move selection of FLL REFCLK into init ASoC: arizona: Tidy up SYNCCLK selection and cache values ASoC: arizona: Factor out check for enabled FLL ASoC: arizona: Factor out FLL disable ASoC: arizona: Factor out FLL enable ASoC: arizona: Improve suppression of noop FLL updates ASoC: arizona: Add support for directly setting the FLL REFCLK ASoC: arizona: Add convience define for clearing SYNCCLK ASoC: arizona: Don't enable FLL on REFCLK configuration ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls ASoC: wm0010: Constify usage of firmware filenames ASoC: wm0010: Report filename when we fail to load firmware Dan Carpenter (1): ASoC: tas5086: signedness bug in tas5086_hw_params() Daniel Mack (7): ASoC: ak4104: convert to direct regmap API usage ASoC: ak4104: correct tranceiver enable handling ASoC: Add codec driver for AK5386 ALSA: ASoC: add codec driver for TI TAS5086 ASoC: davinci-mcasp: clean up davinci_hw_common_param() ASoC: cs4271: convert to direct regmap API usage ASoC: cs4271: switch to mute_stream Dylan Reid (1): ASoC: max98088: Add TLV data for volume controls. Fabio Estevam (2): ASoC: fsl: imx-audmux: Use devm_clk_get() ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period Heiko Stübner (1): ASoC: samsung: fix neo1973-wm8753 compilation Jeeja KP (1): ASoC: compress - add support for metadata apis Kuninori Morimoto (61): ASoC: core: tidyup snd_soc_register_codec() fail case ASoC: core: use snd_soc_register_dais() on codec ASoC: core: remove codec from list if registration failed ASoC: fsi: remove unused irq ASoC: add snd_soc_register_component() ASoC: add .name for snd_soc_component_driver ASoC: fsi: use snd_soc_register_component() instead of snd_soc_register_dais() ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: snd_soc_register_component() uses properly snd_soc_register_dai[s]() ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function ASoC: switch over to use snd_soc_register_component() on sh4 siu Lars-Peter Clausen (28): ASoC: core: Drop unused "dapm" field form soc_enum struct ASoC: core: Remove unused "n_widgets" field from snd_soc_dapm struct ASoC: spear_pcm: Staticize non-exported structs ASoC: omap: Call omap_mcbsp_set_threshold() from mcbsp hw_params ASoC: ux500_pcm: Remove duplicated SNDRV_PCM_HW_PARAM_PERIODS constraint ASoC: spear_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: omap-pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: tegra_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: atmel-pcm-dma: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: speaer_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: atmel-pcm-dma: Do not use snd_dmaengine_pcm_{set,get}_data() ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params ASoC: mmp-pcm: Allocate dma filter parameters on the stack ASoC: imx-pcm: Embed the imx_dma_data struct in the dma_params struct ASoC: mxs: Embed the mxs_dma_data struct in the mxs_pcm_dma_params struct ASoC: dmaengine-pcm: Remove snd_dmaengine_pcm_{set,get}_data ASoC: Constify the 'driver' field of snd_soc_platform ASoC: Constify the 'ops' field of snd_soc_platform_driver ASoC: Constify the 'compr_ops' field of snd_soc_platform_driver ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: omap: Use common DAI DMA data ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: ep93xx_pcm: Fix compile error ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture Mark Brown (72): ASoC: arizona: Only allow input volume updates when inputs are enabled ASoC: wm8960: Add input boost volume control ASoC: omap: Check regulator enable for DAC on Pandora ASoC: arizona: If we only have a clock to synchronise with make it REFCLK mfd: arizona: Define additional FLL control registers mfd: wm5102: Map in additional FLL control registers ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies ASoC: arizona: Optimise FLL loop gains ASoC: arizona: Ensure synchroniser is disabled when not needed ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 Merge tag 'v3.9-rc2' into asoc-core ASoC: arizona: Provide defines for the clock rates ASoC: wm_adsp: Handle old .bin files ASoC: arizona: Ensure we clock two channels for I2S mode ASoC: wm8994: Support constraining the maximum number of channels clocked mfd: wm5102: Add additional speaker control registers ASoC: arizona: Factor out speaker widgets from CODEC drivers ASoC: arizona: Log thermal events ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged ASoC: arizona: Basic support for ISRC rate selection ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates extcon: arizona: Factor out magic application ASoC: arizona: Fix interaction between headphone outputs and identification extcon: arizona: Fix interaction between headphone outputs and identification Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: arizona: Add delay after powering up line level outputs Merge branch 'topic/pxa' of git://git.kernel.org/.../broonie/sound into asoc-component ASoC: wm5102: Implement OSR support ASoC: wm2000: Expose some more registers for diagnostics ASoC: wm_adsp: Provide defines for firmwares ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Merge branch 'topic/omap' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'fix/samsung' of git://git.kernel.org/.../broonie/sound into asoc-samsung ASoC: core: Merge Samsung fixes to avoid trivial context conflict ASoC: wm8994: Remove duplicate revision cache ASoC: wm_adsp: Add support for firmware wide coefficient blocks Merge remote-tracking branch 'asoc/fix/compress' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/samsung' into asoc-next Merge remote-tracking branch 'asoc/fix/tegra' into asoc-next Merge remote-tracking branch 'asoc/fix/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1373' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/component' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/maintainers' into asoc-next Merge remote-tracking branch 'asoc/topic/max98088' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5086' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm-hubs' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm2000' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8903' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Michal Bachraty (1): ASoC: davinci-mcasp: Add support for multichannel playback Paul Bolle (3): ASoC: samsung: remove last traces of neo1973-gta01 ASoC: codecs: remove hidden prompt ASoC: Ux500: remove test for undefined Kconfig macro Paul Handrigan (1): ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Peter Ujfalusi (4): ASoC: omap3pandora: Fix compilation error ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address Ryo Tsutsui (2): ASoC: dapm: Remove redundant clear_walk() for supply widgets ASoC: dapm: Only clear paths we've walked Sachin Kamat (5): ASoC: max98090: Remove unneeded version.h header include ASoC: max98090: Convert to devm_regmap_init_i2c() ASoC: max98090: Make struct dev_pm_ops const ASoC: max98090: Fix checkpatch errors related to spacing ASoC: max98090: Remove unnecessary braces Sascha Hauer (1): ASoC: imx-ssi: Fix AC97 rates Sebastien Guiriec (2): ASoC: omap-mcpdm: Clean up with devm_* function ASoC: omap-dmic: Clean up with devm_request_and_ioremap Shawn Guo (1): ASoC: imx-sgtl5000: use of_node to match cpu dai Silviu-Mihai Popescu (1): ASoC: omap: convert to devm_ioremap_resource() Stephen Warren (10): ASoC: tegra: assume CONFIG_OF in tegra_asoc_utils_init ASoC: tegra_wm8753: minor cleanup ASoC: tegra_wm9712: assume CONFIG_OF ASoC: tegra_alc5632: assume CONFIG_OF, and other cleanup ASoC: tegra trimslice: assume CONFIG_OF, and other cleanup ASoC: tegra_wm8903: assume CONFIG_OF, remove platform data ASoC: tegra: add Tegra114 support to the AHUB driver ASoC: tegra: add Tegra114 support to tegra_asoc_utils.c ASoC: export snd_soc_register_component ASoC: dapm: Implement mixer control sharing Tim Gardner (2): ASoC: adau1373: adau1373_hw_params: Silence overflow warning ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Vinod Koul (1): ASoC: mid-x86 - add support for meaadata apis Wei Yongjun (2): ASoC: tas5086: use module_i2c_driver to simplify the code ASoC: arizona: remove duplicated include from arizona.c Yegor Yefremov (2): ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode ASoC: davinci-mcasp: don't overwrite DIT settings Zoltan Puskas (1): ASoC: atmel: Add slave mode support to SSC in DSP Mode A Documentation/devicetree/bindings/sound/ak5386.txt | 19 + .../bindings/sound/nvidia,tegra30-ahub.txt | 26 +- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 - arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 - arch/arm/plat-samsung/devs.c | 6 + drivers/extcon/extcon-arizona.c | 107 ++-- drivers/mfd/wm5102-tables.c | 10 + include/linux/mfd/arizona/core.h | 3 + include/linux/mfd/arizona/registers.h | 42 ++ include/linux/mfd/wm8994/pdata.h | 8 + include/sound/dmaengine_pcm.h | 27 +- include/sound/soc-dai.h | 8 - include/sound/soc-dapm.h | 1 - include/sound/soc.h | 29 +- include/sound/tas5086.h | 7 + include/sound/tegra_wm8903.h | 26 - sound/soc/atmel/atmel-pcm-dma.c | 25 +- sound/soc/atmel/atmel_ssc_dai.c | 54 +- sound/soc/au1x/ac97c.c | 9 +- sound/soc/au1x/i2sc.c | 9 +- sound/soc/au1x/psc-ac97.c | 9 +- sound/soc/au1x/psc-i2s.c | 9 +- sound/soc/blackfin/bf5xx-ac97.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 9 +- sound/soc/blackfin/bf6xx-i2s.c | 9 +- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 18 +- sound/soc/cirrus/ep93xx-i2s.c | 25 +- sound/soc/cirrus/ep93xx-pcm.c | 37 +- sound/soc/cirrus/ep93xx-pcm.h | 20 - sound/soc/cirrus/simone.c | 2 - sound/soc/cirrus/snappercl15.c | 1 - sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/adau1373.c | 5 +- sound/soc/codecs/ak4104.c | 55 +- sound/soc/codecs/ak5386.c | 152 +++++ sound/soc/codecs/arizona.c | 530 +++++++++++++++--- sound/soc/codecs/arizona.h | 34 +- sound/soc/codecs/cs4271.c | 166 +++--- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/max98088.c | 30 +- sound/soc/codecs/max98090.c | 45 +- sound/soc/codecs/si476x.c | 47 +- sound/soc/codecs/tas5086.c | 591 ++++++++++++++++++++ sound/soc/codecs/wm0010.c | 6 +- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2000.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 131 ++--- sound/soc/codecs/wm5102.h | 6 +- sound/soc/codecs/wm5110.c | 54 +- sound/soc/codecs/wm5110.h | 6 +- sound/soc/codecs/wm8903.c | 6 + sound/soc/codecs/wm8960.c | 10 + sound/soc/codecs/wm8994.c | 68 ++- sound/soc/codecs/wm8994.h | 3 +- sound/soc/codecs/wm_adsp.c | 124 +++- sound/soc/codecs/wm_adsp.h | 5 +- sound/soc/codecs/wm_hubs.c | 9 +- sound/soc/davinci/davinci-i2s.c | 15 +- sound/soc/davinci/davinci-mcasp.c | 97 +++- sound/soc/davinci/davinci-pcm.c | 16 +- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/davinci-vcif.c | 11 +- sound/soc/dwc/designware_i2s.c | 9 +- sound/soc/fsl/fsl_ssi.c | 45 +- sound/soc/fsl/imx-audmux.c | 3 +- sound/soc/fsl/imx-pcm-dma.c | 58 +- sound/soc/fsl/imx-pcm-fiq.c | 6 +- sound/soc/fsl/imx-pcm.h | 19 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 42 +- sound/soc/fsl/imx-ssi.h | 7 +- sound/soc/fsl/mpc5200_psc_ac97.c | 8 +- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +- sound/soc/jz4740/jz4740-i2s.c | 9 +- sound/soc/kirkwood/kirkwood-i2s.c | 11 +- sound/soc/mid-x86/sst_platform.c | 20 +- sound/soc/mid-x86/sst_platform.h | 2 + sound/soc/mxs/mxs-pcm.c | 43 +- sound/soc/mxs/mxs-pcm.h | 4 +- sound/soc/mxs/mxs-saif.c | 17 +- sound/soc/nuc900/nuc900-ac97.c | 9 +- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 +- sound/soc/omap/mcbsp.h | 7 +- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 47 +- sound/soc/omap/omap-hdmi.c | 33 +- sound/soc/omap/omap-mcbsp.c | 27 +- sound/soc/omap/omap-mcpdm.c | 118 ++-- sound/soc/omap/omap-pcm.c | 83 +-- sound/soc/omap/omap-pcm.h | 40 -- sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 9 +- sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/pxa/mmp-pcm.c | 33 +- sound/soc/pxa/mmp-sspa.c | 9 +- sound/soc/pxa/pxa-ssp.c | 9 +- sound/soc/pxa/pxa2xx-ac97.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/s6000/s6000-i2s.c | 9 +- sound/soc/samsung/Kconfig | 5 +- sound/soc/samsung/ac97.c | 14 +- sound/soc/samsung/goni_wm8994.c | 11 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 33 +- sound/soc/samsung/idma.c | 11 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/pcm.c | 11 +- .../include/plat => sound/soc/samsung}/regs-ac97.h | 0 .../include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 9 +- sound/soc/samsung/s3c-i2s-v2.h | 7 +- sound/soc/samsung/s3c2412-i2s.c | 12 +- sound/soc/samsung/s3c24xx-i2s.c | 13 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- sound/soc/samsung/spdif.c | 11 +- sound/soc/sh/fsi.c | 14 +- sound/soc/sh/hac.c | 10 +- sound/soc/sh/migor.c | 2 +- sound/soc/sh/siu_dai.c | 11 +- sound/soc/sh/ssi.c | 10 +- sound/soc/soc-compress.c | 41 +- sound/soc/soc-core.c | 130 ++++- sound/soc/soc-dapm.c | 320 ++++++----- sound/soc/soc-dmaengine-pcm.c | 68 ++- sound/soc/soc-io.c | 5 +- sound/soc/spear/spdif_in.c | 9 +- sound/soc/spear/spdif_out.c | 9 +- sound/soc/spear/spear_pcm.c | 22 +- sound/soc/tegra/tegra20_ac97.c | 28 +- sound/soc/tegra/tegra20_ac97.h | 4 +- sound/soc/tegra/tegra20_i2s.c | 28 +- sound/soc/tegra/tegra20_i2s.h | 4 +- sound/soc/tegra/tegra20_spdif.c | 22 +- sound/soc/tegra/tegra20_spdif.h | 4 +- sound/soc/tegra/tegra30_ahub.c | 72 ++- sound/soc/tegra/tegra30_ahub.h | 24 +- sound/soc/tegra/tegra30_i2s.c | 28 +- sound/soc/tegra/tegra30_i2s.h | 4 +- sound/soc/tegra/tegra_alc5632.c | 15 +- sound/soc/tegra/tegra_asoc_utils.c | 34 +- sound/soc/tegra/tegra_asoc_utils.h | 1 + sound/soc/tegra/tegra_pcm.c | 48 +- sound/soc/tegra/tegra_pcm.h | 7 - sound/soc/tegra/tegra_wm8753.c | 15 +- sound/soc/tegra/tegra_wm8903.c | 179 +++--- sound/soc/tegra/tegra_wm9712.c | 7 +- sound/soc/tegra/trimslice.c | 56 +- sound/soc/txx9/txx9aclc-ac97.c | 9 +- sound/soc/ux500/ux500_msp_dai.c | 13 +- sound/soc/ux500/ux500_msp_dai.h | 5 - sound/soc/ux500/ux500_pcm.c | 76 +-- sound/soc/ux500/ux500_pcm.h | 14 - 164 files changed, 3437 insertions(+), 1710 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 include/sound/tas5086.h delete mode 100644 include/sound/tegra_wm8903.h delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h create mode 100644 sound/soc/codecs/ak5386.c create mode 100644 sound/soc/codecs/tas5086.c delete mode 100644 sound/soc/omap/omap-pcm.h rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%)

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic support for Asus X101CH
by David Henningsson 15 Apr '13

15 Apr '13

With this patch, a TRRS headset mic cannot be successfully detected on the Asus X101CH, and we can also distinguish between headphone and headset automatically. Buglink: https://bugs.launchpad.net/bugs/1169138 Co-authored-by: Kailang <kailang(a)realtek.com> Tested-by: Luis Henriques <luis.henriques(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Now that we have better headset mic support infrastructure, I remembered I forgot to upstream this patch. It was originally provided by Realtek (therefore, I cannot answer for the msleeps and exactly why this is needed). I just rewrote it for the current kernel version. sound/pci/hda/patch_realtek.c | 56 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6cf27f5..1b7aed5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2680,6 +2680,34 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->gen.automute_hook = alc269_quanta_automute; } +static void alc269_x101_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + msleep(100); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); + msleep(500); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_automute_hook = alc269_x101_hp_automute_hook; + } +} + + /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -2837,6 +2865,9 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_ASUS_X101_FUNC, + ALC269_FIXUP_ASUS_X101_VERB, + ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3027,30 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_ASUS_X101_FUNC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_x101_headset_mic, + }, + [ALC269_FIXUP_ASUS_X101_VERB] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0310}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_FUNC + }, + [ALC269_FIXUP_ASUS_X101] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x04a1182c }, /* Headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_ASUS_X101_VERB + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3044,6 +3099,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- 1.7.9.5

2 2

[alsa-devel] [PATCH v2 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 15 Apr '13

15 Apr '13

Hi, This series adds DT support for phycore-ac97 using the fsl-ssi driver. In version 2 I discarded most of the imx-ssi work and integrated the ac97-slave support in fsl-ssi, including support for imx-pcm-fiq as alternative to dma. There are some other changes to get phycore-ac97 working. The first two patches have notes about the detailed changes since version 1. The rest of the patches is new and necessary for the ac97-slave support. Regards, Markus Markus Pargmann (11): ASoC: phycore-ac97: Add DT support ASoC: imx-pcm-dma: Add support for DMA init by ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl-ssi: Add SACNT definitions ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Setup generic imx dma params ARM: imx: Export ac97 reset functions ASoC: fsl-ssi: imx ac97 support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc Documentation/devicetree/bindings/powerpc/fsl/ssi.txt | 73 - Documentation/devicetree/bindings/sound/fsl,ssi.txt | 13 b/Documentation/devicetree/bindings/sound/fsl,ssi.txt | 73 + b/Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt | 12 b/arch/arm/mach-imx/mach-pca100.c | 7 b/arch/arm/mach-imx/mach-pcm043.c | 7 b/sound/soc/fsl/Kconfig | 2 b/sound/soc/fsl/fsl_ssi.c | 70 + b/sound/soc/fsl/fsl_ssi.h | 8 b/sound/soc/fsl/imx-pcm-dma.c | 31 b/sound/soc/fsl/imx-pcm-fiq.c | 14 b/sound/soc/fsl/imx-pcm.h | 6 b/sound/soc/fsl/imx-ssi.c | 7 b/sound/soc/fsl/imx-ssi.h | 1 b/sound/soc/fsl/phycore-ac97.c | 185 ++++ sound/soc/fsl/fsl_ssi.c | 379 +++++++--- sound/soc/fsl/imx-pcm.h | 9 17 files changed, 664 insertions(+), 233 deletions(-)

4 34

[alsa-devel] Stupid question #1 of many -- unit test harness?
by TJ Olaes 15 Apr '13

15 Apr '13

Hi. So, I'm attempting to write the driver for the eClaro card for a couple of reasons, one of which is that I currently don't have sound on mine. So... is there some kind of test harness framework for new drivers? I have no prior idea how to move forward to build this thing, so I'm snooping around in the oxygen source files and trying to copy things that look like they make sense. For instance, I noted that there's mention of an "AK4396" related to output. I assume this is some kind of chip, and I saw in the Claro code that something called "ak4396_init(...)" is called, so I just copied that line and pasted it into the eclaro_init(...) function I'm writing. I have no idea how I'm going to test this short of just compiling the kernel and booting into it. I feel like that's the brute-force, non-smart way to do it, and that you guys who've been doing this for a while might have ideas and recommendations on how to better test my code so I'm not rebooting my machine each time I change a line and recompile. I'd appreciate your guidance, community. Thanks. -TJ

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: disable autopm for MIDI devices
by Clemens Ladisch 15 Apr '13

15 Apr '13

Commit 88a8516a2128 (ALSA: usbaudio: implement USB autosuspend) introduced autopm for all USB audio/MIDI devices. However, many MIDI devices, such as synthesizers, do not merely transmit MIDI messages but use their MIDI inputs to control other functions. With autopm, these devices would get powered down as soon as the last MIDI port device is closed on the host. Even some plain MIDI interfaces could get broken: they automatically send Active Sensing messages while powered up, but as soon as these messages cease, the receiving device would interpret this as an accidental disconnection. Commit f5f165418cab (ALSA: usb-audio: Fix missing autopm for MIDI input) introduced another regression: some devices (e.g. the Roland GAIA SH-01) are self-powered but do a reset whenever the USB interface's power state changes. To work around all this, just disable autopm for all USB MIDI devices. Reported-by: Laurens Holst Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/midi.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 34b9bb7..e5fee18 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -126,7 +126,6 @@ struct snd_usb_midi { struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; - bool autopm_reference; unsigned int opened[2]; unsigned char disconnected; unsigned char input_running; @@ -1040,7 +1039,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, { struct snd_usb_midi* umidi = substream->rmidi->private_data; struct snd_kcontrol *ctl; - int err; down_read(&umidi->disc_rwsem); if (umidi->disconnected) { @@ -1051,13 +1049,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, mutex_lock(&umidi->mutex); if (open) { if (!umidi->opened[0] && !umidi->opened[1]) { - err = usb_autopm_get_interface(umidi->iface); - umidi->autopm_reference = err >= 0; - if (err < 0 && err != -EACCES) { - mutex_unlock(&umidi->mutex); - up_read(&umidi->disc_rwsem); - return -EIO; - } if (umidi->roland_load_ctl) { ctl = umidi->roland_load_ctl; ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; @@ -1080,8 +1071,6 @@ static int substream_open(struct snd_rawmidi_substream *substream, int dir, snd_ctl_notify(umidi->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); } - if (umidi->autopm_reference) - usb_autopm_put_interface(umidi->iface); } } mutex_unlock(&umidi->mutex); @@ -2256,6 +2245,8 @@ int snd_usbmidi_create(struct snd_card *card, return err; } + usb_autopm_get_interface_no_resume(umidi->iface); + list_add_tail(&umidi->list, midi_list); return 0; }

2 1

[alsa-devel] [PATCH] ALSA: hda - Implement headset jack functionality for some Dell hw
by David Henningsson 15 Apr '13

15 Apr '13

On some machines, there is a headset jack that can support both headphone, headsets (of both CTIA and OMTP type) and mic-in. On other machines, the headset jack supports headphone, headsets (both CTIA and OMTP), but not mic-in. This patch implements that functionality as different capture sources. Buglink: https://bugs.launchpad.net/bugs/1169143 Tested-by: David Chen <david.chen(a)canonical.com> Co-authored-by: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 386 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 386 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6cf27f5..f3307eb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -52,6 +52,20 @@ enum { ALC_INIT_GPIO3, }; +enum { + ALC_HEADSET_MODE_UNKNOWN, + ALC_HEADSET_MODE_UNPLUGGED, + ALC_HEADSET_MODE_HEADSET, + ALC_HEADSET_MODE_MIC, + ALC_HEADSET_MODE_HEADPHONE, +}; + +enum { + ALC_HEADSET_TYPE_UNKNOWN, + ALC_HEADSET_TYPE_CTIA, + ALC_HEADSET_TYPE_OMTP, +}; + struct alc_customize_define { unsigned int sku_cfg; unsigned char port_connectivity; @@ -87,6 +101,11 @@ struct alc_spec { unsigned int gpio_led; /* used for alc269_fixup_hp_gpio_led() */ + hda_nid_t headset_mic_pin; + hda_nid_t headphone_mic_pin; + int current_headset_mode; + int current_headset_type; + /* hooks */ void (*init_hook)(struct hda_codec *codec); #ifdef CONFIG_PM @@ -2798,6 +2817,302 @@ static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc_headset_mode_unplugged(struct hda_codec *codec) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x1b, 0x0c0b); + alc_write_coef_idx(codec, 0x45, 0xc429); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val & 0xbfff); + alc_write_coef_idx(codec, 0x06, 0x2104); + alc_write_coef_idx(codec, 0x1a, 0x0001); + alc_write_coef_idx(codec, 0x26, 0x0004); + alc_write_coef_idx(codec, 0x32, 0x42a3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x18, 0x7308); + alc_write_coef_idx(codec, 0x6b, 0xc429); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to unplugged mode.\n"); +} + + +static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, + hda_nid_t mic_pin) +{ + int val; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xc429); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + val = alc_read_coef_idx(codec, 0x35); + alc_write_coef_idx(codec, 0x35, val | 1<<14); + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x1a, 0x0021); + alc_write_coef_idx(codec, 0x26, 0x008c); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + case 0x10ec0292: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0x19, 0xa208); + alc_write_coef_idx(codec, 0x2e, 0xacf0); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0xb5, 0x1040); + val = alc_read_coef_idx(codec, 0xc3); + alc_write_coef_idx(codec, 0xc3, val | 1<<12); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; + } + snd_printdd("Headset jack set to mic-in mode.\n"); +} + +static void alc_headset_mode_default(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x06, 0x2100); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x76, 0x000e); + alc_write_coef_idx(codec, 0x6c, 0x2400); + alc_write_coef_idx(codec, 0x6b, 0xc429); + alc_write_coef_idx(codec, 0x18, 0x7308); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0041); + alc_write_coef_idx(codec, 0x15, 0x0d40); + alc_write_coef_idx(codec, 0xb7, 0x802b); + break; + } + snd_printdd("Headset jack set to headphone (default) mode.\n"); +} + +/* Iphone type */ +static void alc_headset_mode_ctia(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to iPhone-style headset mode.\n"); +} + +/* Nokia type */ +static void alc_headset_mode_omtp(struct hda_codec *codec) +{ + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xe429); + alc_write_coef_idx(codec, 0x1b, 0x0c2b); + alc_write_coef_idx(codec, 0x32, 0x4ea3); + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xe429); + alc_write_coef_idx(codec, 0x76, 0x0008); + alc_write_coef_idx(codec, 0x18, 0x7388); + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x15, 0x0d50); + alc_write_coef_idx(codec, 0xc3, 0x0000); + break; + } + snd_printdd("Headset jack set to Nokia-style headset mode.\n"); +} + +static void alc_determine_headset_type(struct hda_codec *codec) +{ + int val; + bool is_ctia = false; + struct alc_spec *spec = codec->spec; + + switch (codec->vendor_id) { + case 0x10ec0283: + alc_write_coef_idx(codec, 0x45, 0xd029); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; + case 0x10ec0292: + alc_write_coef_idx(codec, 0x6b, 0xd429); + msleep(300); + val = alc_read_coef_idx(codec, 0x6c); + is_ctia = (val & 0x001c) == 0x001c; + break; + case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); + alc_write_coef_idx(codec, 0xb7, 0x802b); + alc_write_coef_idx(codec, 0x15, 0x0d60); + alc_write_coef_idx(codec, 0xc3, 0x0c00); + msleep(300); + val = alc_read_coef_idx(codec, 0xbe); + is_ctia = (val & 0x1c02) == 0x1c02; + break; + } + + snd_printdd("Headset jack detected iPhone-style headset: %s\n", + is_ctia ? "yes" : "no"); + spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; +} + +static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; + + if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} + +static void alc_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + alc_update_headset_mode(codec); +} + +static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + spec->current_headset_type = ALC_HEADSET_MODE_UNKNOWN; + snd_hda_gen_hp_automute(codec, jack); +} + +static void alc_probe_headset_mode(struct hda_codec *codec) +{ + int i; + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + /* Find mic pins */ + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].is_headset_mic && !spec->headset_mic_pin) + spec->headset_mic_pin = cfg->inputs[i].pin; + if (cfg->inputs[i].is_headphone_mic && !spec->headphone_mic_pin) + spec->headphone_mic_pin = cfg->inputs[i].pin; + } + + spec->gen.cap_sync_hook = alc_update_headset_mode_hook; + spec->gen.automute_hook = alc_update_headset_mode; + spec->gen.hp_automute_hook = alc_update_headset_jack_cb; +} + +static void alc_fixup_headset_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + alc_probe_headset_mode(codec); + break; + case HDA_FIXUP_ACT_INIT: + spec->current_headset_mode = 0; + alc_update_headset_mode(codec); + break; + } +} + +static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + } + else + alc_fixup_headset_mode(codec, fix, action); +} + +static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + int val; + alc_write_coef_idx(codec, 0xc4, 0x8000); + val = alc_read_coef_idx(codec, 0xc2); + alc_write_coef_idx(codec, 0xc2, val & 0xfe); + snd_hda_set_pin_ctl_cache(codec, 0x18, 0); + } + alc_fixup_headset_mode(codec, fix, action); +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2837,6 +3152,10 @@ enum { ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, + ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_HEADSET_MODE, + ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, @@ -2996,6 +3315,35 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, }, + [ALC269_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, + [ALC269_FIXUP_DELL2_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x16, 0x21014020 }, /* dock line out */ + { 0x19, 0x21a19030 }, /* dock mic */ + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, + [ALC269_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode, + }, + [ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_no_hp_mic, + }, [ALC271_FIXUP_AMIC_MIC2] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3030,6 +3378,26 @@ static const struct hda_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c7, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ec, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ed, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05ee, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f3, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -3536,6 +3904,8 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC668_FIXUP_HEADSET_MODE, }; static const struct hda_fixup alc662_fixups[] = { @@ -3696,6 +4066,20 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE + }, + [ALC668_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc668, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -3704,6 +4088,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), -- 1.7.9.5

2 1

[alsa-devel] PROBLEM: iMX6 SSI - set up as a master of external audio codec
by Andrey Gelman 15 Apr '13

15 Apr '13

Hi, everyone ! I'm bringing up a wm8731 audio codec on an iMX6 based board. I use the slave mode, i.e. SSI outputs all the clocks towards the codec IC. I work with 3.0.35 Android kernel provided by freescale. While, generally, the playback is Ok, I face 2 issues: - cannot play back mono streams (too high rate, one speaker produces only noise) - cannot play back streams with sampling rate as low as 8kHz (the sound is accompanied by high distortion) In the mainline kernel (commits to v3.6), there is a workaround to the 1-st issue by simply removing mono support from imx-ssi.c. Similar approach would fix the 2-nd issue as well - SSI would report it does not support 8kHz sampling rate. In a later commit, this workaround has been reverted, enabling mono support back. My question is this: Is this a real limitation of SSI ? Does it support mono playback or not ? Does SSI support 8kHz sampling rate ? If yes, what could be possible reason to the distortions I get, while higher rates are played back well ? Is there a real fix to these issues, or only workarounds are available ? Thanks, Andrey Gelman

1 0

[alsa-devel] [PATCH 20/30] ASoC: samsung: convert to dmaengine API
by Arnd Bergmann 15 Apr '13

15 Apr '13

In order to build the exynos kernel with CONFIG_ARCH_MULTIPLATFORM, we must convert all users of the Samsung private DMA interface to the generic dmaengine API. This version of the patch adds the generic dmaengine API as an alternative to the existing samsung specific one. Once all the older platforms provide support for the common dmaengine interfaces, we can remove the old code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/dma.c | 219 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/dma.h | 15 +++- sound/soc/samsung/i2s.c | 2 - sound/soc/samsung/pcm.c | 1 - sound/soc/samsung/spdif.c | 1 - 5 files changed, 232 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21b7926..9fd53df 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -22,8 +22,14 @@ #include <sound/pcm_params.h> #include <asm/dma.h> + +#ifdef CONFIG_SAMSUNG_DMADEV #include <mach/hardware.h> #include <mach/dma.h> +#else +#include <linux/dmaengine.h> +#include <linux/amba/pl330.h> +#endif #include "dma.h" @@ -62,11 +68,13 @@ struct runtime_data { static void audio_buffdone(void *data); +#ifdef CONFIG_SAMSUNG_DMADEV /* dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ + static void dma_enqueue(struct snd_pcm_substream *substream) { struct runtime_data *prtd = substream->runtime->private_data; @@ -265,6 +273,217 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } +#else +/* dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. + */ + +static void dma_enqueue(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned long period = prtd->dma_period; + unsigned int limit; + enum dma_transfer_direction direction; + struct dma_chan *chan = prtd->params->ch; + struct dma_async_tx_descriptor *desc; + + pr_debug("Entered %s\n", __func__); + + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + direction = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); + + while (prtd->dma_loaded < limit) { + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + period) > prtd->dma_end) { + period = prtd->dma_end - pos; + pr_debug("%s: corrected dma len %ld\n", + __func__, period); + } + + desc = dmaengine_prep_dma_cyclic(chan, pos, + prtd->dma_period*limit, period, direction, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + + if (desc) { + desc->callback = audio_buffdone; + desc->callback_param = substream; + dmaengine_submit(desc); + } + + prtd->dma_loaded++; + pos += period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } + + prtd->dma_pos = pos; +} + +static void audio_buffdone(void *data) +{ + struct snd_pcm_substream *substream = data; + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (prtd->state & ST_RUNNING) { + prtd->dma_pos += prtd->dma_period; + if (prtd->dma_pos >= prtd->dma_end) + prtd->dma_pos = prtd->dma_start; + + if (substream) + snd_pcm_period_elapsed(substream); + } +} + +static int dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + struct dma_slave_config config; + dma_cap_mask_t mask; + + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, channel %d\n", prtd->params, + prtd->params->channel); + + dma_cap_zero(mask); + dma_cap_set(DMA_CYCLIC, mask); + + prtd->params->ch = dma_request_slave_channel_compat(mask, + pl330_filter, (void *)prtd->params->channel, + rtd->cpu_dai->dev, prtd->params->ch_name); + + memset(&config, 0, sizeof(struct dma_slave_config)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + config.direction = DMA_MEM_TO_DEV; + config.dst_addr = prtd->params->dma_addr; + config.dst_addr_width = prtd->params->dma_size; + config.dst_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } else { + config.direction = DMA_DEV_TO_MEM; + config.src_addr = prtd->params->dma_addr; + config.src_addr_width = prtd->params->dma_size; + config.src_maxburst = 1; + dmaengine_slave_config(prtd->params->ch, &config); + } + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int dma_hw_free(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + dmaengine_terminate_all(prtd->params->ch); + dma_release_channel(prtd->params->ch); + prtd->params = NULL; + } + + return 0; +} + +static int dma_prepare(struct snd_pcm_substream *substream) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* flush the DMA channel */ + dmaengine_terminate_all(prtd->params->ch); + + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + dma_enqueue(substream); + + return ret; +} + +static int dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->state |= ST_RUNNING; + dma_async_issue_pending(prtd->params->ch); + break; + + case SNDRV_PCM_TRIGGER_STOP: + prtd->state &= ~ST_RUNNING; + dmaengine_terminate_all(prtd->params->ch); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} +#endif static snd_pcm_uframes_t dma_pointer(struct snd_pcm_substream *substream) diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..6bd1857 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,13 +12,24 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H +#ifdef CONFIG_SAMSUNG_DMADEV +#include <mach/dma.h> +#endif + struct s3c_dma_params { +#ifdef CONFIG_SAMSUNG_DMADEV struct s3c2410_dma_client *client; /* stream identifier */ + unsigned ch; + struct samsung_dma_ops *ops; +#else + struct s3c2410_dma_client { + char *name; /* unused */ + } *client; + struct dma_chan *ch; +#endif int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ - unsigned ch; - struct samsung_dma_ops *ops; char *ch_name; }; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e3..61f2622 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include <sound/pcm_params.h> -#include <mach/dma.h> - #include <linux/platform_data/asoc-s3c.h> #include "dma.h" diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 13bab79..b5f267c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5008e5b..ee792aa 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" -- 1.8.1.2

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[alsa-devel] [PATCH v2] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 15 Apr '13

15 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Calculate nsec from frames, subtract delay for playback as well as capture. sound/pci/hda/hda_intel.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..3e6f2c6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,23 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_subtract_codec_delay(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + u64 codec_frames, codec_nsecs; + + if (!hinfo->ops.get_delay) + return nsec; + + codec_frames = hinfo->ops.get_delay(hinfo, apcm->codec, substream); + codec_nsecs = div_u64(codec_frames * 1000000000LL, + substream->runtime->rate); + + return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1914,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_subtract_codec_delay(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 9

[alsa-devel] Can Realtek HD-A codec ALC282 support D3cold?
by Lin, Mengdong 15 Apr '13

15 Apr '13

Hi, We want to put an idle HD-A codec into D3cold for more power saving and preparation for being power gated. The Realtek ALC282 is used in our test. Could anyone help to confirm whether ALC282 can support D3cold or not? I'm told that ALC282 can support D3cold. But my test shows it cannot. At first, ALC282 HW does not report that it can support D3cold. Its parameter "Supported Power States" returns 0xc000000f, means it only supports: EPSS, CLKSTOP, D3hot, D2, D1 and D0. Is this a just mistake in reporting the bit, or D3cold is not supported all? Then when I explicitly set the audio function group to D3cold by verb 0x705, and then get its power state after 200ms by verb 0xf05, The codec response is 0x00000000: No PS-Error, both PS-Act and PS-set is D0. So it seems the command to D3cold is not processed by codec HW at all. Is there someone from Realtek or with experience on ALC282 here can share some information about this issue? Many thanks Mengdong

1 0

[alsa-devel] M-Audio Delta 1010LT: S/PDIF clock source not working
by Jonas Petersen 15 Apr '13

15 Apr '13

Hi, I am trying to capture from S/PDIF. I get a signal but the clocking is broken somehow. I have 'Multi Track Internal Clock' on 'ICE958 Input'. 'Word Clock Status' and 'Word Clock Sync' are 'Off'. What happens is the following: It captures what comes in on S/PDIF but it seems it will always use the internal clock with the setting from 'Multi Track Internal Clock Default'. So even though 'Multi Track Internal Clock' is on 'ICE958 Input', the setting of 'Multi Track Internal Clock Default' will clock the signal. A 1 kHz sine wave at 48 kHz samplerate captured with 'Multi Track Internal Clock Default' on 96 kHz will result in a 2 kHz sine. When the sample frequency rates are matched, it will capture at correct rate but with hiccups due to the rates not perfectly aligned. What can I do narrow down the cause? Reards Jonas

4 8

[alsa-devel] USB Quirks for Boss JS-8
by Keith A. Milner 15 Apr '13

15 Apr '13

Hi, I'm looking at developing a quirk for the Boss JS-8. The lsusb is as follows: Bus 001 Device 003: ID 0582:0109 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0109 bcdDevice 0.01 iManufacturer 1 BOSS iProduct 2 JS-8 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered I wasn't entirely sure about the quirk syntax as the two interfaces are at EP 4 and EP5. Does this mean I need to ignore the (non-existant) lower interfaces? Here's what I came up with: { /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "BOSS", */ /* .product_name = "JS-8", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 3, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = -1 } } } }, regards, -- Keith A. Milner

5 9

[alsa-devel] alsa-project.org access, I want to tackle building an eClaro driver
by TJ Olaes 13 Apr '13

13 Apr '13

Hello. I hate to post to this mailing list regarding access to the alsa-project.org wiki, but I've gone through the site and can't, for the life of me, seem to find an admin email or something like it on the wiki. I'm trying to sign up to the wiki and am having trouble with the Captcha. I put in different version numbers of linux kernels but all my attempts to register seem to be thwarted because I'm not putting in the correct info. I have no idea what I'm doing wrong. What I'm really trying to do is get involved in alsa driver development... I think this would be an appropriate place for me to attempt to address my issue, which is the lack of an HT Omega eClaro card driver. I really, really want to contribute but it seems I'm being thwarted by the developer website. If I could get contact info for an admin or something, I would go bug them, but it seems that info isn't posted on the wiki. Sorry to bug y'all. Thanks. -TJ

3 3

Re: [alsa-devel] [PATCH] alsa/usb: Add quirk for 192KHz recording on E-Mu devices
by Takashi Iwai 13 Apr '13

13 Apr '13

At Fri, 12 Apr 2013 22:33:59 -0500, Calvin Owens wrote: > > When recording at 176.2KHz or 192Khz, the device adds a 32-bit length > header to the capture packets, which obviously needs to be ignored for > recording to work properly. > > Userspace expected: L0 L1 L2 R0 R1 R2 > ...but actually got: R2 L0 L1 L2 R0 R1 > > Also, the last byte of the length header being interpreted as L0 of > the first sample caused spikes every 0.5ms, resulting in a loud 16KHz > tone (about the highest 'B' on a piano) being present throughout > captures. > > Tested at all sample rates on an E-Mu 0404USB, and tested for > regressions on a generic USB headset. > > Signed-off-by: Calvin Owens <jcalvinowens(a)gmail.com> Thanks, applied for 3.10 kernel (but with Cc to stable) now. Takashi > --- > sound/usb/card.h | 1 + > sound/usb/pcm.c | 2 +- > sound/usb/quirks.c | 1 + > sound/usb/stream.c | 1 + > 4 files changed, 4 insertions(+), 1 deletion(-) > > diff --git a/sound/usb/card.h b/sound/usb/card.h > index 8a751b4..d32ea41 100644 > --- a/sound/usb/card.h > +++ b/sound/usb/card.h > @@ -116,6 +116,7 @@ struct snd_usb_substream { > unsigned int altset_idx; /* USB data format: index of alternate setting */ > unsigned int txfr_quirk:1; /* allow sub-frame alignment */ > unsigned int fmt_type; /* USB audio format type (1-3) */ > + unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ > > unsigned int running: 1; /* running status */ > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index f94397b..a481fea 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -1170,7 +1170,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, > stride = runtime->frame_bits >> 3; > > for (i = 0; i < urb->number_of_packets; i++) { > - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; > + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset + subs->pkt_offset_adj; > if (urb->iso_frame_desc[i].status && printk_ratelimit()) { > snd_printdd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); > // continue; > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 5325a38..7e292b9 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -837,6 +837,7 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, > break; > } > snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); > + subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; > } > > void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > diff --git a/sound/usb/stream.c b/sound/usb/stream.c > index ad181d5..0927cc6 100644 > --- a/sound/usb/stream.c > +++ b/sound/usb/stream.c > @@ -94,6 +94,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, > subs->dev = as->chip->dev; > subs->txfr_quirk = as->chip->txfr_quirk; > subs->speed = snd_usb_get_speed(subs->dev); > + subs->pkt_offset_adj = 0; > > snd_usb_set_pcm_ops(as->pcm, stream); > > -- > 1.8.1.5 >

1 0

Re: [alsa-devel] [PATCH] ASoC: samsung: fix neo1973-wm8753 compilation
by Mark Brown 12 Apr '13

12 Apr '13

On Thu, Apr 11, 2013 at 09:02:43PM +0200, Heiko Stübner wrote: > Commit b2ca78717cea (ARM: S3C24XX: make gta02.h local) already replaced > the GTA02_GPIO_* constants in neo1973-wm8753.c but forgot to remove the > inclusion of mach/gta02.h before moving the file out of mach/. Applied, thanks.

1 0

[alsa-devel] ASoC update for v3.9-rc6
by Mark Brown 12 Apr '13

12 Apr '13

The following changes since commit 31880c37c11e28cb81c70757e38392b42e695dc6: Linux 3.9-rc6 (2013-04-07 20:49:54 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-maintainers-v3.9-rc6 for you to fetch changes up to b02e48f2f8755e51fe8ce246988a29bd96f6b85e: MAINTAINERS: Update e-mail address (2013-04-12 11:53:41 +0100) May as well go along with the other sound updates. ---------------------------------------------------------------- MAINTAINERS: Update e-mail address Update the e-mail address I use for subsystems. ---------------------------------------------------------------- Mark Brown (1): MAINTAINERS: Update e-mail address MAINTAINERS | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-)

2 1

[alsa-devel] [PATCH 18/30] ASoC: samsung: move plat/ headers to local directory
by Arnd Bergmann 12 Apr '13

12 Apr '13

The plat/iis.h and plat/ac97.h files in the samsung platform are only needed by the ASoC drivers, so they can be moved into the same directory, as one more step towards a multiplatform build. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- arch/arm/mach-s3c24xx/dma-s3c2410.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2412.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2440.c | 2 -- arch/arm/mach-s3c24xx/dma-s3c2443.c | 2 -- sound/soc/samsung/ac97.c | 2 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h | 0 {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h | 0 sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 12 files changed, 6 insertions(+), 14 deletions(-) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-ac97.h (100%) rename {arch/arm/plat-samsung/include/plat => sound/soc/samsung}/regs-iis.h (100%) diff --git a/arch/arm/mach-s3c24xx/dma-s3c2410.c b/arch/arm/mach-s3c24xx/dma-s3c2410.c index a6c94b8..30aa53f 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2410.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2410.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2410_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2412.c b/arch/arm/mach-s3c24xx/dma-s3c2412.c index c0e8c3f..ab1700e 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2412.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2412.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID } diff --git a/arch/arm/mach-s3c24xx/dma-s3c2440.c b/arch/arm/mach-s3c24xx/dma-s3c2440.c index 1c08eccd..cd25de2 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2440.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2440.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> static struct s3c24xx_dma_map __initdata s3c2440_dma_mappings[] = { diff --git a/arch/arm/mach-s3c24xx/dma-s3c2443.c b/arch/arm/mach-s3c24xx/dma-s3c2443.c index 000e4c6..5fe3539 100644 --- a/arch/arm/mach-s3c24xx/dma-s3c2443.c +++ b/arch/arm/mach-s3c24xx/dma-s3c2443.c @@ -25,10 +25,8 @@ #include <plat/regs-serial.h> #include <mach/regs-gpio.h> -#include <plat/regs-ac97.h> #include <plat/regs-dma.h> #include <mach/regs-lcd.h> -#include <plat/regs-iis.h> #include <plat/regs-spi.h> #define MAP(x) { \ diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 0df3c56..c76abdf 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <mach/dma.h> -#include <plat/regs-ac97.h> +#include "regs-ac97.h" #include <linux/platform_data/asoc-s3c.h> #include "dma.h" diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 15a3817..fa91376 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -20,7 +20,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index a301d8c..ccc601d 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -21,7 +21,7 @@ #include <sound/soc.h> #include <asm/mach-types.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <mach/gta02.h> #include "../codecs/wm8753.h" diff --git a/arch/arm/plat-samsung/include/plat/regs-ac97.h b/sound/soc/samsung/regs-ac97.h similarity index 100% rename from arch/arm/plat-samsung/include/plat/regs-ac97.h rename to sound/soc/samsung/regs-ac97.h diff --git a/arch/arm/plat-samsung/include/plat/regs-iis.h b/sound/soc/samsung/regs-iis.h similarity index 100% rename from arch/arm/plat-samsung/include/plat/regs-iis.h rename to sound/soc/samsung/regs-iis.h diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index a5826ea..704460a 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -24,7 +24,7 @@ #include <sound/soc.h> #include <sound/jack.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include <asm/mach-types.h> #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 13f6dd1..a7b17c1 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -24,7 +24,7 @@ #include <sound/pcm_params.h> #include <mach/dma.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "dma.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 333e1b7..1b7b52b 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -18,7 +18,7 @@ #include <sound/soc.h> #include <sound/s3c24xx_uda134x.h> -#include <plat/regs-iis.h> +#include "regs-iis.h" #include "s3c24xx-i2s.h" -- 1.8.1.2

3 5

[alsa-devel] [PATCH] MAINTAINERS: Update e-mail address
by Mark Brown 12 Apr '13

12 Apr '13

Update the e-mail address I use for subsystems. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 836a618..c6108c3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6625,7 +6625,7 @@ S: Supported F: fs/reiserfs/ REGISTER MAP ABSTRACTION -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git S: Supported F: drivers/base/regmap/ @@ -7374,7 +7374,7 @@ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC @@ -7463,7 +7463,7 @@ F: drivers/clk/spear/ SPI SUBSYSTEM M: Grant Likely <grant.likely(a)secretlab.ca> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> L: spi-devel-general(a)lists.sourceforge.net Q: http://patchwork.kernel.org/project/spi-devel-general/list/ T: git git://git.secretlab.ca/git/linux-2.6.git @@ -8708,7 +8708,7 @@ F: drivers/scsi/vmw_pvscsi.h VOLTAGE AND CURRENT REGULATOR FRAMEWORK M: Liam Girdwood <lrg(a)ti.com> -M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> +M: Mark Brown <broonie(a)kernel.org> W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 T: git git://git.kernel.org/pub/scm/linux/kernel/git/lrg/regulator.git -- 1.7.10.4

1 0

[alsa-devel] alsa 1.0.27 release
by Jaroslav Kysela 12 Apr '13

12 Apr '13

Hello all, the alsa 1.0.27 packages were released. You may download these packages from the main ALSA page: http://www.alsa-project.org . There are no updates for pyalsa and alsa-oss. The full changelog is provided here: http://www.alsa-project.org/main/index.php/Changes_v1.0.26_v1.0.27 Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH] asoc: add dummy codec
by Stas Sergeev 12 Apr '13

12 Apr '13

Hi. The attached patch adds a dummy codec for the asoc subsystem. I use it with the driver that programs the soc to drive I2S from an internal PLLs, and the audio then goes to HDMI, so no external codec is used. I've found spdif_transciever.c, which exactly is such a dummy driver, but it is bound to the particular plat device, so I made the device name a config option, and if not set, the plat device is created with the default name. It may be possible to avoid the usage of the codec driver at all, but I haven't succeeded with this, and the existance of spdif_transciever.c suggests there is no such way. Thoughts? Signed-off-by: Stas Sergeev <stsp(a)users.sourceforge.net>

3 14

Re: [alsa-devel] [PATCH 2/4] mfd: wm5102: Map in additional FLL control registers
by Mark Brown 11 Apr '13

11 Apr '13

On Mon, Apr 08, 2013 at 06:22:23PM +0200, Samuel Ortiz wrote: > Hi Mark, > > On Wed, Mar 06, 2013 at 02:28:14PM +0800, Mark Brown wrote: > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/wm5102-tables.c | 8 ++++++++ > > 1 file changed, 8 insertions(+) > Did you merge this patchset through the ASoC tree ? Yes, let me know if that's a problem and I can back stuff out but it seemed sensible.

1 0

[alsa-devel] [PATCH 22/30] ASoC: samsung/idma: export idma_reg_addr_init
by Arnd Bergmann 11 Apr '13

11 Apr '13

The idma_reg_addr_init function is used by the samsung i2s driver, which can be a loadable module, so we have to export this function. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/idma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index f36a541..6e5fed3 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -411,6 +411,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.regs = regs; idma.lp_tx_addr = addr; } +EXPORT_SYMBOL_GPL(idma_reg_addr_init); static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 21/30] ASoC: samsung/i2s: fix module_device_table
by Arnd Bergmann 11 Apr '13

11 Apr '13

The second argument to the module_device_table macro must be the name of the device id array. In the samsung i2s driver, there was a small typo, resulting in a build error when building it as a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 61f2622..f7c6816 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1289,7 +1289,7 @@ static struct platform_device_id samsung_i2s_driver_ids[] = { }, {}, }; -MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); +MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { -- 1.8.1.2

2 1

[alsa-devel] [PATCH 19/30] ASoC: samsung: use irq resource for idma
by Arnd Bergmann 11 Apr '13

11 Apr '13

With multiplatform kernels, we cannot use hardwired IRQ numbers in device drivers. This changes the idma driver to use a proper resource, like all other drivers do. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- arch/arm/plat-samsung/devs.c | 6 ++++++ sound/soc/samsung/idma.c | 10 ++++++++-- 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-samsung/devs.c b/arch/arm/plat-samsung/devs.c index 78be9c0..b441908 100644 --- a/arch/arm/plat-samsung/devs.c +++ b/arch/arm/plat-samsung/devs.c @@ -145,14 +145,20 @@ struct platform_device s3c_device_camif = { /* ASOC DMA */ +#ifdef CONFIG_PLAT_S5P +static struct resource samsung_asoc_idma_resource = DEFINE_RES_IRQ(IRQ_I2S0); + struct platform_device samsung_asoc_idma = { .name = "samsung-idma", .id = -1, + .num_resources = 1, + .resource = &samsung_asoc_idma_resource, .dev = { .dma_mask = &samsung_device_dma_mask, .coherent_dma_mask = DMA_BIT_MASK(32), } }; +#endif /* FB */ diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index a07950b..f36a541 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -68,6 +68,8 @@ static struct idma_info { dma_addr_t lp_tx_addr; } idma; +static int idma_irq; + static void idma_getpos(dma_addr_t *src) { *src = idma.lp_tx_addr + @@ -305,7 +307,7 @@ static int idma_open(struct snd_pcm_substream *substream) if (prtd == NULL) return -ENOMEM; - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + ret = request_irq(idma_irq, iis_irq, 0, "i2s", prtd); if (ret < 0) { pr_err("fail to claim i2s irq , ret = %d\n", ret); kfree(prtd); @@ -324,7 +326,7 @@ static int idma_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct idma_ctrl *prtd = runtime->private_data; - free_irq(IRQ_I2S0, prtd); + free_irq(idma_irq, prtd); if (!prtd) pr_err("idma_close called with prtd == NULL\n"); @@ -418,6 +420,10 @@ static struct snd_soc_platform_driver asoc_idma_platform = { static int asoc_idma_platform_probe(struct platform_device *pdev) { + idma_irq = platform_get_irq(pdev, 0); + if (idma_irq < 0) + return idma_irq; + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); } -- 1.8.1.2

2 1

[alsa-devel] [PATCH resend] sound: at73c213: Use dev_pm_ops
by Lars-Peter Clausen 11 Apr '13

11 Apr '13

Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/spi/at73c213.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd60d8..a1a24b9 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1075,10 +1075,11 @@ out: return 0; } -#ifdef CONFIG_PM -static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +#ifdef CONFIG_PM_SLEEP + +static int snd_at73c213_suspend(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); @@ -1087,9 +1088,9 @@ static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) return 0; } -static int snd_at73c213_resume(struct spi_device *spi) +static int snd_at73c213_resume(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; clk_enable(chip->board->dac_clk); @@ -1097,18 +1098,21 @@ static int snd_at73c213_resume(struct spi_device *spi) return 0; } + +static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, + snd_at73c213_resume); +#define AT73C213_PM_OPS (&at73c213_pm_ops) + #else -#define snd_at73c213_suspend NULL -#define snd_at73c213_resume NULL +#define AT73C213_PM_OPS NULL #endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", + .pm = AT73C213_PM_OPS, }, .probe = snd_at73c213_probe, - .suspend = snd_at73c213_suspend, - .resume = snd_at73c213_resume, .remove = snd_at73c213_remove, }; -- 1.8.0

2 1

[alsa-devel] [PATCH 0/3] Headphone Mic patches
by David Henningsson 11 Apr '13

11 Apr '13

I'm preparing upstreaming of some new hardware, which has headphone mic support, i e, you can use the headphone jack to plug in a TRS/TS mic instead of plugging in a headphone in that jack. For this hardware, the existing hp_mic functionality won't work as the headphone and mic are on different pins. David Henningsson (3): ALSA: hda - allow "Headphone Mic" parser flag ALSA: hda - add some small convenience functions to auto parser ALSA: hda - Handle Headphone Mic jack more generic sound/pci/hda/hda_auto_parser.c | 68 ++++++++++++++++++++++++++++++++------- sound/pci/hda/hda_auto_parser.h | 29 +++++++++++++++-- sound/pci/hda/hda_generic.c | 12 +------ sound/pci/hda/hda_jack.c | 43 +++++++++++++++++-------- 4 files changed, 113 insertions(+), 39 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Zhang wei 11 Apr '13

11 Apr '13

There are two definitions In the file include/sound/asound.h SNDRV_PCM_FORMAT_S24_3LE and SNDRV_PCM_FORMAT_S24_LE. What is the difference between them? Thanks a lot. zw

2 1

[alsa-devel] VIA VT1731
by ALEKSANDR BOIKO 11 Apr '13

11 Apr '13

Hello! I'm using DAC Audio-GD Master 7 with Voyage MPD. There is a problem with playback audio files different frequency. I've read there is a path - solution. How can I get it? Help please.

2 1

[alsa-devel] ASOC snd_soc_dai_ops for clock mux control
by Gellner, Christoph (ADITG/SW1) 10 Apr '13

10 Apr '13

Hi all, I'm working on the driver of an audio CPU DAI IP providing - Internal clocks which can be used for master clock generation by TX as well as RX section - One oversampling clock available to RX section - One oversampling clock available to TX section. Kernel version is 3.5.7 I want to add support to the CPU DAI driver to - Specify the rate of the input clocks - Select which of the clock inputs shall be used by RX and which shall be used by TX section I'm currently wondering how to map this to existing snd_soc_dai_ops functions. Currently I see only set_sysclk, set_pll, set_clkdiv for clock control. I plan to use set_sysclk to specify the rate as well as the direction of each clock. What is currently missing for me is a way to specify the clock to be used by RX or TX section. I can't really add this in set_sysclk. In case I switch to clock X when set_sysclk is called with clock ID of X, I need a way to distinguish if RX or TX section or both need to switch to this clock. But I can't really double the internal clock X in virtual clocks X_TX, X_RX (one time for TX and one time for RX section) as the internal clock X has only one rate for both sections and this would not represent the hardware correctly. Do you have any recommendation how to implement ? What do you think about adding a new function pointer to snd_soc_dai_ops ? int (*set_clkmux)(struct snd_soc_dai *dai, int mux_id, int source_id); I think this would be the cleanest solution as it models the features of the hardware (the clock multiplexers for RX or TX sections) It might be possible to use set_pll or set_clkdiv to realize. I could use set_pll and ignore the freq_in and freq_out I could use set_clkdiv and interpret div (divider integer value) as a source_id But this is not a 100% clean solution in my opinion. Best regards Christoph

3 3

[alsa-devel] [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
by Charles Keepax 10 Apr '13

10 Apr '13

The app_pointer is managed locally by the compress core for memory mapped DSPs but for DSPs that are not memory mapped this would have to be manually updated from within the DSP driver itself, which is hardly very idiomatic. This patch switches to using the cumulative values to calculate the available buffer space because these are already gracefully passed out of the DSP driver to the compress core and otherwise should be functionally equivalent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 23 +++++------------------ 1 files changed, 5 insertions(+), 18 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..27bd81a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -160,8 +160,6 @@ static int snd_compr_update_tstamp(struct snd_compr_stream *stream, static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, struct snd_compr_avail *avail) { - long avail_calc; /*this needs to be signed variable */ - memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); /* Still need to return avail even if tstamp can't be filled in */ @@ -184,22 +182,11 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, return stream->runtime->buffer_size; } - /* FIXME: this routine isn't consistent, in one test we use - * cumulative values and in the other byte offsets. Do we - * really need the byte offsets if the cumulative values have - * been updated? In the PCM interface app_ptr and hw_ptr are - * already cumulative */ - - avail_calc = stream->runtime->buffer_size - - (stream->runtime->app_pointer - stream->runtime->hw_pointer); - pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc, - stream->runtime->app_pointer, - stream->runtime->hw_pointer); - if (avail_calc >= stream->runtime->buffer_size) - avail_calc -= stream->runtime->buffer_size; - pr_debug("ret avail as %ld\n", avail_calc); - avail->avail = avail_calc; - return avail_calc; + avail->avail = stream->runtime->buffer_size - + (stream->runtime->total_bytes_available - + stream->runtime->total_bytes_transferred); + pr_debug("ret avail as %lld\n", avail->avail); + return avail->avail; } static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) -- 1.7.2.5

5 19

[alsa-devel] [PATCH v2] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 10 Apr '13

10 Apr '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index afef3fb..b84bb73 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

5 23

Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.26-26-ge05b903
by Takashi Iwai 10 Apr '13

10 Apr '13

At Mon, 8 Apr 2013 14:51:54 +0200 (CEST), GIT server wrote: > > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "ALSA utilities repository". > > The branch, master has been updated > via e05b903b1fb16e967d838edac408304cd4470fee (commit) > from 46cb355d4ff41c8065b02644b0534ab42d16cb72 (commit) > > Those revisions listed above that are new to this repository have > not appeared on any other notification email; so we list those > revisions in full, below. > > - Log ----------------------------------------------------------------- > commit e05b903b1fb16e967d838edac408304cd4470fee > Author: Jaroslav Kysela <perex(a)perex.cz> > Date: Mon Apr 8 14:49:31 2013 +0200 > > alsactl: move systemd config to the daemon mode > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I'm not thrilled by the silent default behavior change like this. This will affect all systems using systemd from now on. I understand the reason behind it, but I wonder whether it's an overkill. Yet another daemon, unconditionally no matter whether hotplug or not? Hmm... Takashi

4 13

[alsa-devel] [PATCH] ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
by Daniel Mack 10 Apr '13

10 Apr '13

It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/clock.c | 3 +++ sound/usb/pcm.c | 7 +------ sound/usb/quirks.c | 10 ++++++++++ sound/usb/quirks.h | 1 + 4 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ae35e7d..b0ec364 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -32,6 +32,7 @@ #include "card.h" #include "helper.h" #include "clock.h" +#include "quirks.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -392,7 +393,9 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + snd_usb_set_interface_quirk(dev); } return 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 754cb5b..815a37d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -349,12 +349,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; - /* - * "Playback Design" products need a 50ms delay after setting the - * USB interface. - */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + snd_usb_set_interface_quirk(dev); } subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7882a82..551ae1a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -892,6 +892,16 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 16; } +void snd_usb_set_interface_quirk(struct usb_device *dev) +{ + /* + * "Playback Design" products need a 50ms delay after setting the + * USB interface. + */ + if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) + mdelay(50); +} + void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index f10d00e..665e972 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -26,6 +26,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); +void snd_usb_set_interface_quirk(struct usb_device *dev); void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Tue, Apr 09, 2013 at 05:13:59PM +0200, Alban Bedel wrote: > The Charge Pump needs the DSP clock to work properly, without it the > bypass to HP/LINEOUT is not working properly. This requirement is not > mentioned in the datasheet but has been confirmed by Mark Brown from > Wolfson. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Wed, Mar 20, 2013 at 05:37:33PM +0100, Alban Bedel wrote: > output enable sequence requiere the DSP clock. Some input from Wolfson > will be needed to find out if this is just an undocumented requirement > or a chip bug. It seems it's an undocumented requirement. > + { "Left Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Left Output Mixer", "Right Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Right Bypass Switch", "CLK_DSP" }, The actual change here should be to add a route: { "Charge Pump", NULL, "CLK_DSP" }, Can you submit an update for this or should I do one?

1 0

[alsa-devel] [PATCH v3 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..22f276c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,60 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printk(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume do { } while (0) +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1952,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba147d7..81e91b8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to reset all codecs on the bus. This is implemented as temporary suspend-and-resume calls to codecs as an ad hoc solution. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error, replacing calls to snd_hda_suspend and snd_hda_resume. It's because snd_hda_resume need change for system resume: It will delay resuming a codec that is not ready to resume for some external dependency, to avoid blocking system resume. But this change of "delay resume" is not suitable for recovering from codec communication error, which need resume the codec immediately after suspend to simulate a bus reset. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..ba147d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,29 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs on the bus. + * @bus: the HDA bus + * + * This is implemented as temporary suspend-and-resume calls + * to codecs as an ad hoc solution. + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..f39e374 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,61 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1953,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 09 Apr '13

09 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..ec8ac71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,27 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_add_codec_delay_nsec(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + int stream = substream->stream; + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_codec *codec = apcm->codec; + u64 codec_nsec; + + if (hinfo->ops.get_delay) { + codec_nsec = + hinfo->ops.get_delay(hinfo, codec, substream) * 1000000; + if (stream == SNDRV_PCM_STREAM_CAPTURE) + nsec = (nsec > codec_nsec) ? nsec - codec_nsec : 0; + else if (stream == SNDRV_PCM_STREAM_PLAYBACK) + nsec += codec_nsec; + } + + return nsec; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1918,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_add_codec_delay_nsec(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 7

[alsa-devel] [PATCH 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a05af8b..53b1a1c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to suspend and resume all codecs on the hda bus. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error. snd_hda_suspend & snd_hda_resume will no longer be called by azx_bus_reset. So snd_hda_resume will only be used in system resume and may delay resuming a codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..a05af8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,26 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs, i.e. suspend and resume the codecs + * @bus: the HDA bus + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

2 1

Re: [alsa-devel] [RFC] revoke(2) and generic handling of things like remove_proc_entry()
by Clemens Ladisch 08 Apr '13

08 Apr '13

Al Viro wrote: > BTW, snd_card_disconnect() doesn't do anything to existing mappings; smells > like a bug, and there we do have ones with non-trivial ->mmap(). Could > ALSA folks comment? I don't know of any hotplug sound driver that maps memory from a device. All hotplug buses (PCIe, USB, FireWire) involve DMAing from/to main memory. When a sound device is unplugged, sample transfers just stop, but the buffer itself stays as long as the device node is kept opened. (In the case of a capture device, that actually makes sense.) Regards, Clemens

1 0

Re: [alsa-devel] [PATCH] ASoC: Ux500: remove test for undefined Kconfig macro
by Mark Brown 08 Apr '13

08 Apr '13

On Fri, Mar 29, 2013 at 01:28:21PM +0100, Paul Bolle wrote: > A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there > never was a corresponding Kconfig symbol so this test has always > evaluated to true. And since AB5500 support was removed in v3.5 it > appears safe to remove this test and a few lines of code. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA: hda: fix mute led on HP Envy 15
by Dan Savilonis 08 Apr '13

08 Apr '13

This laptop reports the string HP_Mute_LED_P_G in DMI, but the guess made in hp_set_mute_led() is wrong. So, hard-code the GPIO id. Signed-off-by: Dan Savilonis <djs(a)n-cube.org> --- sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..3a6b111 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, + STAC_HP_ENVY_15, STAC_92HD83XXX_MODELS }; @@ -2092,6 +2093,15 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ } +static void stac92hd83xxx_fixup_hp_envy_15(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_led = 0x01; /* GPIO0 */ +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2101,6 +2111,7 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } + static const struct hda_fixup stac92hd83xxx_fixups[] = { [STAC_92HD83XXX_REF] = { .type = HDA_FIXUP_PINS, @@ -2169,6 +2180,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_15] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_envy_15, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2184,6 +2201,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HP_MIC_LED, .name = "hp-mic-led" }, { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, + { .id = STAC_HP_ENVY_15, .name = "hp-envy-15" }, {} }; @@ -2269,6 +2287,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Pavilion dv5", STAC_92HD83XXX_HP_INV_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1688, + "HP Envy 15", STAC_HP_ENVY_15), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -3968,6 +3988,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_add_verbs(codec, stac92hd83xxx_core_init); + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); -- 1.8.1.2

1 0

[alsa-devel] [PATCH 0/8] More dmaengine PCM cleanups
by Lars-Peter Clausen 07 Apr '13

07 Apr '13

Hi, The first few patches in this series are a few unrelated cleanups to different dmaengine PCM drivers and one fix for a small compile error I introduced in the last dmaengine PCM cleanup series. The last four patches introduce a common DAI DMA data struct, which will be used by tegra, omap and fsl/imx. Having a common DAI DMA data struct allows us to factor out some redundant code from these drivers into a common helper function in the dmaengine PCM library. This should hopefully be the last set of cleanups before the generic dmaengine PCM driver. - Lars Lars-Peter Clausen (8): ASoC: ep93xx_pcm: Fix compile error ASoC: tegra: Don't claim to support PCM pause and resume ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: omap: Use common DAI DMA data include/sound/dmaengine_pcm.h | 24 ++++++++++++++++++ sound/soc/atmel/atmel-pcm-dma.c | 2 -- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++------ sound/soc/fsl/imx-pcm-dma.c | 25 ++++++------------- sound/soc/fsl/imx-pcm-fiq.c | 4 +-- sound/soc/fsl/imx-pcm.h | 16 ++++-------- sound/soc/fsl/imx-ssi.c | 21 +++++++++------- sound/soc/fsl/imx-ssi.h | 7 ++++-- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 ++++++----- sound/soc/omap/mcbsp.h | 7 +++--- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 22 ++++++++--------- sound/soc/omap/omap-hdmi.c | 24 ++++++++---------- sound/soc/omap/omap-mcbsp.c | 6 ++--- sound/soc/omap/omap-mcpdm.c | 31 +++++++++++------------- sound/soc/omap/omap-pcm.c | 42 ++++---------------------------- sound/soc/omap/omap-pcm.h | 39 ------------------------------ sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 1 - sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/soc-dmaengine-pcm.c | 39 ++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 13 +++++----- sound/soc/tegra/tegra20_ac97.h | 4 +-- sound/soc/tegra/tegra20_i2s.c | 13 +++++----- sound/soc/tegra/tegra20_i2s.h | 4 +-- sound/soc/tegra/tegra20_spdif.c | 7 +++--- sound/soc/tegra/tegra20_spdif.h | 4 +-- sound/soc/tegra/tegra30_ahub.c | 8 +++--- sound/soc/tegra/tegra30_ahub.h | 8 +++--- sound/soc/tegra/tegra30_i2s.c | 13 +++++----- sound/soc/tegra/tegra30_i2s.h | 4 +-- sound/soc/tegra/tegra_pcm.c | 40 +++--------------------------- sound/soc/tegra/tegra_pcm.h | 7 ------ sound/soc/ux500/ux500_pcm.c | 51 ++++++++++++--------------------------- sound/soc/ux500/ux500_pcm.h | 14 ----------- 40 files changed, 226 insertions(+), 322 deletions(-) delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.8.0

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[alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 07 Apr '13

07 Apr '13

Hi All, I'm attempting to play audio into a WM8960 codec attached to I2S 0 on a Samsung S5PV210. The DMA engine emits a lot of messages complaining of (I think) a read fault: dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 ... I've had audio playing using the I2S idma driver, so I know clocks + mixer settings are OK. I'd like to get the external DMA working because I also need to do capture. Any suggestions on how to debug? Cheers, Mike

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
by Dylan Reid 07 Apr '13

07 Apr '13

Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bd2d49..145d078 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2617,7 +2617,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X")) + if (strcmp(codec->chip_name, "ALC271X") && + strcmp(codec->chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -2838,6 +2839,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC269_FIXUP_ACER_AC700, }; static const struct hda_fixup alc269_fixups[] = { @@ -3010,6 +3012,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC269_FIXUP_ACER_AC700] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x03a11c20 }, /* mic */ + { 0x1e, 0x0346101e }, /* SPDIF1 */ + { 0x21, 0x0321101f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC271_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3034,6 +3049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
by Eldad Zack 07 Apr '13

07 Apr '13

The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/mixer_quirks.c | 4 ++-- sound/usb/quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d274..ebe9144 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a2ac004..68e9220 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -499,7 +499,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; -- 1.8.1.5

3 2

[alsa-devel] [PATCH 1/2] ALSA: add DSD formats
by Daniel Mack 07 Apr '13

07 Apr '13

This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for a copy of the documentation. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 352.8kHz 705.6KHz 1411.2KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz `-----------------------------------' actual DSD sample rates Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index aa7b0a8..d957046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U8 _SNDRV_PCM_FMTBIT(DSD_DOP_U8) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U16 _SNDRV_PCM_FMTBIT(DSD_DOP_U16) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..eb9eda8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_DOP_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD DOP format (x8) */ +#define SNDRV_PCM_FORMAT_DSD_DOP_U16 ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD DOP format (x16) */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_DOP_U16 #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 578327e..578a761 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_DOP_U8), + FORMAT(DSD_DOP_U16), }; const char *snd_pcm_format_name(snd_pcm_format_t format) -- 1.8.1.4

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Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Jussi Laako 07 Apr '13

07 Apr '13

On 04/05/2013 10:59 PM, Andreas Koch wrote: > However, they are interpreting the ASIO spec for DSD differently. They > express buffer size in samples, which means that for DSD it is measured > in bits. That's how pretty much everybody, including Sony who co-authored the spec, has been interpreting it so far... Otherwise all the latency calculations become inconsistent and need special exceptions for DSD. I agree the ASIO spec is not very clear, however it always talks about number of samples and then specifically mentions that DSD samples are _packed_ into bytes and thus all buffer sizes have to multiples of eight. I have a hidden configuration flag to flip that either way as necessary, but default is that sizes are in _bits_. - Jussi

2 5

[alsa-devel] [PATCH v2 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 06 Apr '13

06 Apr '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

1 1

[alsa-devel] got a segmentation fault when use multi-channel device
by Zhang wei 06 Apr '13

06 Apr '13

hello everone. i am now debuging the multi-channel device on my embeded board with alsa-lib.The version alsa-driver and alsa-lib is 1.0.24. -----------------------------> zz@board:$ ls /dev/snd/ by-path controlC0 midiC0D0 pcmC0D0c pcmC0D0p pcmC0D1p seq timer <--------------------------- i edit the asoundrc configure file.like this: ---------------------------------------------------> pcm.!default{ type hw card 0 } ctl.!default{ type hw card 0 } ################################# ## create a virtul device which has 4 channels for playback. ################################# pcm.multi_playback{ type multi; slaves.a.pcm "hw:0,0"; slaves.a.channels 2; slaves.b.pcm "hw:0,1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } ctl.multi_playback{ type hw; card 0; } pcm.ttable{ type route; slave.pcm "multi_playback"; slave.channels 4; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.ttable { type hw; card 0; } <-------------------------------------------------- and then get the Segmentation fault: --------------------------------------------------------------> [root@board /tmp]#./pb_pcm_min open...ver:[0.71] playback...capture...device=[multi_playback] #####################snd_pcm_open... zz[snd_dlsym_verify:122],vname=[_snd_config_hook_load_dlsym_config_hook_ 001] zz[snd_dlsym_verify:127] zz[snd_dlsym:167] zz[snd_dlsym:171],name=[snd_config_hook_load] zz[snd_pcm_open_conf:2163] zz[snd_dlsym_verify:122],vname=[__snd_pcm_multi_open_dlsym_pcm_001] Segmentation fault [root@board /tmp]# <--------------------------------------------------------------- To my surpris, it will be fine when run on the PC under the Ubuntu system.I used the same alsa-lib source code and the configure file asoundrc. in my codec-driver .i have not call the functions :snd_soc_add_controls() / snd_soc_dapm_new_controls() / snd_soc_dapm_add_routes(). is it the reason? But the programm have not arrived here. i have traced the source code when the Segmentation fault happened the alsa-lib was calling the function dlysm() with the param vname=__snd_pcm_multi_open_dlsym_pcm_001. But this will not happen on my PC under Ubuntu. could you help me find out what's wrong with me? thanks a lot. zw

1 0

[alsa-devel] PROBLEM: very loud sound of electric discharge on shutdown and restart with ALC882M with all kernels I tried up to 3.9 rc5
by Jenya Y 06 Apr '13

06 Apr '13

I always had this problem in linux and only today tried to search for answers. Solutions I came across involved disabling power management on Intel HDA, which is already 0 by default and also the option of muting on shutdown/ restart, which had absolutely no effect. I wish I could figure this out myself but after 4 hours of searching and trying things out and countless restarts I've given up and decided to come to you for help. Jenya On 03/20/2013 08:41 AM, Jenya Y wrote: > I still didn't receive anything from you as I already mentioned I tried > most of the 'obvious' solutions I came accross and still looking for > answers. > Maybe I'm using the wrong ML for this question. If so, maybe you could at > least suggest another HDA/ ALSA maintainer to contact in this situation. > > TIA, Jenya.

1 0

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
by Dylan Reid 06 Apr '13

06 Apr '13

The DSP in the CA0132 codec adds a variable latency to audio depending on what processing is being done. Add a new patch op to return that latency for capture and playback streams. The latency is determined by which blocks are enabled and knowing how much latency is added by each block. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Rebase on tiwai's patch to add a pcm_op, this is dependent on that. Get ms to frames conversion right. Test loopback and compare measured vs reported latency. sound/pci/hda/patch_ca0132.c | 55 ++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 53 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 12eb21a..841842c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -131,6 +131,13 @@ enum { /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* Latency introduced by DSP blocks in milliseconds. */ +#define DSP_CAPTURE_INIT_LATENCY 0 +#define DSP_CRYSTAL_VOICE_LATENCY 124 +#define DSP_PLAYBACK_INIT_LATENCY 13 +#define DSP_PLAY_ENHANCEMENT_LATENCY 30 +#define DSP_SPEAKER_OUT_LATENCY 7 + struct ct_effect { char name[44]; hda_nid_t nid; @@ -2743,6 +2750,31 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_PLAYBACK_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + /* Add latency if playback enhancement and either effect is enabled. */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) { + if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) || + (spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID])) + latency += DSP_PLAY_ENHANCEMENT_LATENCY; + } + + /* Applying Speaker EQ adds latency as well. */ + if (spec->cur_out_type == SPEAKER_OUT) + latency += DSP_SPEAKER_OUT_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Digital out */ @@ -2811,6 +2843,23 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_CAPTURE_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) + latency += DSP_CRYSTAL_VOICE_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Controls stuffs. */ @@ -4002,7 +4051,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_max = 6, .ops = { .prepare = ca0132_playback_pcm_prepare, - .cleanup = ca0132_playback_pcm_cleanup + .cleanup = ca0132_playback_pcm_cleanup, + .get_delay = ca0132_playback_pcm_delay, }, }; @@ -4012,7 +4062,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .channels_max = 2, .ops = { .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup + .cleanup = ca0132_capture_pcm_cleanup, + .get_delay = ca0132_capture_pcm_delay, }, }; -- 1.8.1.5

3 9

[alsa-devel] [PATCH 00/10] UAC2: Automatic clock switching
by Eldad Zack 05 Apr '13

05 Apr '13

Hi, This patch series is intended for-next. It provides automatic clock switching for UAC2 devices (patches #6-#8). If the current selected clock is not valid, other clocks will be checked and the first (indexwise) valid clock will be used. Patch #8 provides a module flag to turn this logic off, since in some use cases (e.g., studio work) it might not be desired. But I think most users want their audio devices to "just work", so the default is on. To make the change more obvious, I first moved the validity check around without changing any logic (#6). Patch #10 adds support for clocks with read-only sample frequency control. All other patches included are trivial clean up patches. I'd appreciate a review and/or testing this patch series. This applies to all UAC2 devices, so please let me know if anyone sees any issue. Tested with mainline. Applies against mainline (3.9-rc4), HEAD at 46a1f21a679abaaeae6db9969963dc998c9f1c1c Applies against Takashi's for-next, HEAD at 4abdbd1c2c1832e7270e546307ffb3e56b286db2 Cheers, Eldad Zack (10): ALSA: usb-audio: convert list_for_each to entry variant ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement ALSA: usb-audio: neaten EXPORT_SYMBOLS placement ALSA: usb-audio: spelling correction ALSA: usb-audio: use endianness macros ALSA: usb-audio: UAC2: do clock validity check earlier ALSA: usb-audio: UAC2: try to find and switch to valid clock ALSA: usb-audio: UAC2: auto clock selection module param ALSA: usb-audio: show err in set_sample_rate_v2 debug ALSA: usb-audio: UAC2: support read-only freq control sound/usb/card.c | 11 ++-- sound/usb/clock.c | 150 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/clock.h | 3 +- sound/usb/endpoint.c | 18 +++---- sound/usb/endpoint.h | 2 +- sound/usb/format.c | 2 +- sound/usb/midi.c | 12 ++--- sound/usb/pcm.c | 32 ++++------- sound/usb/proc.c | 7 ++- sound/usb/stream.c | 12 ++--- sound/usb/usbaudio.h | 1 + 11 files changed, 155 insertions(+), 95 deletions(-) -- 1.8.1.5

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Re: [alsa-devel] Deadlock over semaphore issue with aplay while using dmix
by mateen 05 Apr '13

05 Apr '13

Hi, I checked with Takshi's patch and would like to suggest couple of changes. 1. We keep dmix_down_sem(dmix) and dmix_up_sem(dmix) as macros instead of defining them as functions. 2. We free up the semaphore in snd_pcm_dmix_close() if it is recognized that the semaphore is held up by the same process, which will be indicated by dmix->semlocked flag. Mateen. diff -Nuar dir2/pcm_direct.h dir1/pcm_direct.h --- dir2/pcm_direct.h 2009-12-16 20:48:51.000000000 +0530 +++ dir1/pcm_direct.h 2013-04-05 17:06:48.331497000 +0530 @@ -123,6 +123,7 @@ int ipc_gid; /* IPC socket gid */ int semid; /* IPC global semaphore identification */ int shmid; /* IPC global shared memory identification */ + int semlocked; snd_pcm_direct_share_t *shmptr; /* pointer to shared memory area */ snd_pcm_t *spcm; /* slave PCM handle */ snd_pcm_uframes_t appl_ptr; diff -Nuar dir2/pcm_dmix.c dir1/pcm_dmix.c --- dir2/pcm_dmix.c 2009-12-16 20:48:51.000000000 +0530 +++ dir1/pcm_dmix.c 2013-04-05 17:04:02.781109000 +0530 @@ -285,8 +285,17 @@ */ #ifndef DOC_HIDDEN #ifdef NO_CONCURRENT_ACCESS -#define dmix_down_sem(dmix) snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT) -#define dmix_up_sem(dmix) snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT) +#define dmix_down_sem(dmix) \ +{\ +if (!dmix->semlocked++)\ + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT);\ +} +#define dmix_up_sem(dmix) \ +{\ + if (!--dmix->semlocked)\ + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT);\ +} + #else #define dmix_down_sem(dmix) #define dmix_up_sem(dmix) @@ -764,7 +773,13 @@ if (dmix->timer) snd_timer_close(dmix->timer); - snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + if(!dmix->semlocked) + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + else{ + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + } + snd_pcm_close(dmix->spcm); if (dmix->server) snd_pcm_direct_server_discard(dmix); ------------------------------ ---------------------------------------- Message: 1 Date: Thu, 04 Apr 2013 18:30:01 +0200 From: Takashi Iwai <tiwai(a)suse.de> To: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] Deadlock over semaphore issue with aplay while using dmix Message-ID: <s5h38v645ti.wl%tiwai(a)suse.de> Content-Type: text/plain; charset=US-ASCII At Thu, 04 Apr 2013 13:33:08 +0200, Jaroslav Kysela wrote: > > Date 4.4.2013 11:27, mateen wrote: > > Hi, > > > > I seeing sometimes deadlock issue with dmix when I press CTRL+C. > > > > Aplay's signal handler calls snd_pcm_close() if an interrupt occurs. > > snd_pcm_close() will internally call pcm->ops->close() which will fall to > > snd_pcm_dmix_close() in case you are using dmix. > > > > snd_pcm_dmix_close() will try to acquire the semaphore with > > snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT). > > The same semaphore is acquired in snd_pcm_dmix_sync_area() with > > dmix_down_sem() in case of non-concurrent access. > > > > If semaphore is acquired in snd_pcm_dmix_sync_area() which is in thread > > context and interrupt comes, which invokes the signal handler which is in > > ISR context, which calls snd_pcm_close() which in turn calls > > snd_pcm_dmix_close() then we see a deadlock since semaphore is not released > > from thread context and ISR is waiting indefinitely on the same semaphore. > > > > Please suggest a suitable solution for this. > > It seems that also other configurations (alsa-lib plugins) have trouble > with the closing from the signal handler - I hit mutex issues with the > PulseAudio plugin, too. > > The question is, how we can do a clean path in this case. Looking to the > current alsa-lib code, I would suggest to add the snd_pcm_abort() > function (may be called from the interrupt handler) to notify the > library to not ignore -EINTR return codes from poll() and other i/o ops > and pass it to the caller (application) to finish the normal close sequence. > > Opinions? Isn't it only about the direct plugins with the case where no coherent memory is available? If so, we may just avoid the deadlock by checking some internal flag like the patch below (untested)? It's no perfect but just some proof, of course. Takashi -- Regards, Shaikh Mateen S. 09423350444

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[alsa-devel] Deadlock over semaphore issue with aplay while using dmix
by mateen 05 Apr '13

05 Apr '13

Hi, I seeing sometimes deadlock issue with dmix when I press CTRL+C. Aplay's signal handler calls snd_pcm_close() if an interrupt occurs. snd_pcm_close() will internally call pcm->ops->close() which will fall to snd_pcm_dmix_close() in case you are using dmix. snd_pcm_dmix_close() will try to acquire the semaphore with snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT). The same semaphore is acquired in snd_pcm_dmix_sync_area() with dmix_down_sem() in case of non-concurrent access. If semaphore is acquired in snd_pcm_dmix_sync_area() which is in thread context and interrupt comes, which invokes the signal handler which is in ISR context, which calls snd_pcm_close() which in turn calls snd_pcm_dmix_close() then we see a deadlock since semaphore is not released from thread context and ISR is waiting indefinitely on the same semaphore. Please suggest a suitable solution for this. -- Regards, Shaikh Mateen S. 09423350444

3 6

[alsa-devel] [PATCH v6 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 05 Apr '13

05 Apr '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> Tested-by: Daniel Mack <zonque(a)gmail.com> --- v6: CC Mark Brown v5: add Daniel's Tested-by v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

2 2

Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Daniel Mack 05 Apr '13

05 Apr '13

Hi Andreas, On 28.03.2013 05:58, Andreas Koch wrote: >> Am I right to assume that in your case it's the same as for DoP, >> IOW, oldest in MSB? > > Good point - thanks for bringing it up - another can of worms. > With Windows having a little endian philosophy we started early on > with the first DSD workstation by putting the oldest sample in the > LSB position. ASIO came along and adopted the same thing, but then > extended support for the opposite as well. Rigisystems (offering OEM > modules for USB audio) came along about a year ago and insisted that > there is only their way, i.e. putting the oldest bit in the MSB > position. I tried to explain to them that there are already hundreds > of products out in the market that do it differently, but that didn't > matter to them. The result is that now we have implementations with > both. Up until now that was not a huge issue, because there is no > generic Windows driver and every hardware manufacturer had to have a > custom driver. > Now of course, with your heroic effort in creating a generic driver > that would help everyone, we have a mess. Some DACs expect little > endian (mine and others as well) and others (all that are based on > Rigisystems modules) big endian positions. > > Since the time I had the discussions with Rigisystems I didn't have > any further communication with them. Maybe it is a good idea to touch > base with them again and find out what the latest is. I will try my > contact there and let you know. Did you hear back from them yet? Do you think we should allocate types for Big Endian DOP formats? Thanks, Daniel

1 0

[alsa-devel] [PATCH] pcm: fix typo, should be SND_PCM_STATE_PREPARED.
by dsvensson＠gmail.com 05 Apr '13

05 Apr '13

From: Daniel Svensson <dsvensson(a)gmail.com> Signed-off-by: Daniel Svensson <dsvensson(a)gmail.com> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..0868dd5 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -162,7 +162,7 @@ The PCM device has accepted communication parameters and it is waiting for #snd_pcm_prepare() call to prepare the hardware for selected operation (playback or capture). -\par SND_PCM_STATE_PREPARE +\par SND_PCM_STATE_PREPARED The PCM device is prepared for operation. Application can use #snd_pcm_start() call, write or read data to start the operation. -- 1.7.12.4 (Apple Git-37)

2 1

[alsa-devel] [PATCH] ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.
by Dylan Reid 05 Apr '13

05 Apr '13

The Stumpy ChromeBox needs its pin configs fixed up. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0d9c58f..bd8d46c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -68,6 +68,7 @@ enum { enum { CS421X_CDB4210, CS421X_SENSE_B, + CS421X_STUMPY, }; /* Vendor-specific processing widget */ @@ -538,6 +539,7 @@ static int patch_cs420x(struct hda_codec *codec) /* CS4210 board names */ static const struct hda_model_fixup cs421x_models[] = { { .id = CS421X_CDB4210, .name = "cdb4210" }, + { .id = CS421X_STUMPY, .name = "stumpy" }, {} }; @@ -559,6 +561,17 @@ static const struct hda_pintbl cdb4210_pincfgs[] = { {} /* terminator */ }; +/* Stumpy ChromeBox */ +static const struct hda_pintbl stumpy_pincfgs[] = { + { 0x05, 0x022120f0 }, + { 0x06, 0x901700f0 }, + { 0x07, 0x02a120f0 }, + { 0x08, 0x77a70037 }, + { 0x09, 0x77a6003e }, + { 0x0a, 0x434510f0 }, + {} /* terminator */ +}; + /* Setup GPIO/SENSE for each board (if used) */ static void cs421x_fixup_sense_b(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -578,7 +591,11 @@ static const struct hda_fixup cs421x_fixups[] = { [CS421X_SENSE_B] = { .type = HDA_FIXUP_FUNC, .v.func = cs421x_fixup_sense_b, - } + }, + [CS421X_STUMPY] = { + .type = HDA_FIXUP_PINS, + .v.pins = stumpy_pincfgs, + }, }; static const struct hda_verb cs421x_coef_init_verbs[] = { -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
by Dylan Reid 05 Apr '13

05 Apr '13

Allow a codec to specify a delay in addition to the delay before samples are passed through DMA. Some codecs have large processing delays because of built-in DSPs that require codec-side buffers of many milliseconds. Default to zero delay if the codec doesn't specify a callback to get the processing delay. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 17 +++++++++++++++++ sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_intel.c | 2 ++ 3 files changed, 23 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..1d792dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3724,6 +3724,23 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, } EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); +/** + * snd_hda_codec_processing_delay - get the fixed delay through the codec + * @codec: HD-audio codec + * @stream: playback or capture + * + * This is needed for codecs that contain DSPs that cause long delays. + */ +unsigned int snd_hda_codec_processing_delay(struct hda_codec *codec, + unsigned int stream) +{ + if (codec->patch_ops.get_processing_delay) + return codec->patch_ops.get_processing_delay(codec, stream); + + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_processing_delay); + /* * supported power states check */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..23bd1e7 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -725,6 +725,8 @@ struct hda_codec_ops { int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); + unsigned int (*get_processing_delay)(struct hda_codec *codec, + unsigned int stream); }; /* record for amp information cache */ @@ -1063,6 +1065,8 @@ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); void snd_hda_bus_reboot_notify(struct hda_bus *bus); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); +unsigned int snd_hda_codec_processing_delay(struct hda_codec *codec, + unsigned int stream); int snd_hda_lock_devices(struct hda_bus *bus); void snd_hda_unlock_devices(struct hda_bus *bus); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 418bfc0..422bfaf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2349,6 +2349,7 @@ static unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev, bool with_check) { + struct azx_pcm *apcm = snd_pcm_substream_chip(azx_dev->substream); unsigned int pos; int stream = azx_dev->substream->stream; int delay = 0; @@ -2400,6 +2401,7 @@ static unsigned int azx_get_position(struct azx *chip, chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } azx_dev->substream->runtime->delay = + snd_hda_codec_processing_delay(apcm->codec, stream) + bytes_to_frames(azx_dev->substream->runtime, delay); } trace_azx_get_position(chip, azx_dev, pos, delay); -- 1.8.1.5

2 5

[alsa-devel] Handle large period size end-of-stream situation
by Patrick Lai 04 Apr '13

04 Apr '13

Hi, I am dealing with a use case requiring passing large buffer down to the DSP. Application in the user-space passes PCM through mmap interface. Per my understanding, ALSA is designed to work with DMA. Data is transferred in chunk equal to period size. At the end of playback, there may not be enough audio data left in the music stream to fill entire period. If remaining audio data only takes up very small chunk of period, playback takes longer to stop. Given that period size I have to deal with is quite large, this problem is observed easily. If my understanding is correct, what is the standard/recommended way of handling end of stream case? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 6

[alsa-devel] [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Apr '13

04 Apr '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> Tested-by: Daniel Mack <zonque(a)gmail.com> --- v5: add Daniel's Tested-by v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

3 6

[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by Wendelin Klimann 04 Apr '13

04 Apr '13

Hello Peter I am stuck again in my ASoc driver development and it would be really nice if you could give me an hint how i could solve this. Acctually i try to connect my PCM3168 Audio Codec to the McBSP on the BeagleBoard. My new driver loads fine and i got 2 soundcards: *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: omap3beagle [omap3beagle], device 0: TWL4030 twl4030-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: pcm3168 [pcm3168], device 0: PCM3168 pcm3168-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Your TWL4030 driver is still working perfect with: *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -Dhw:0,0 /home/root/fifi.wav* [ 1245.302551] omap-dma-engine omap-dma-engine: allocating channel for 33 Playing WAVE '/home/root/fifi.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but when i play the same file with my new driver i get an error (as codec master and also as codec slave): *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -Dhw:1,0 /home/root/fifi.wav* [ 1128.677337] omap-dma-engine omap-dma-engine: allocating channel for 17 Playing WAVE '/h[ 1128.689544] can't set codec DAI configuration - pcm3168 ome/root/fifi.wa[ 1128.695281] asoc: machine hw_params failed: -524 v' : Signed 16 b[ 1128.702514] omap-dma-engine omap-dma-engine: freeing channel for 17 it Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1145: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 5512 PERIOD_BYTES: 22048 PERIODS: 5 BUFFER_TIME: (624943 624944) BUFFER_SIZE: 27560 BUFFER_BYTES: 110240 TICK_TIME: 0 To check whether my machine driver (/omap-pcm3168.c/) is working i tried to implement the /twl4030.c/ codec driver into it (and deactivated the omap_twl4030_audio_init() function), which worked just fine. With this setup i changed the /.cpu_dai_name = "omap-mcbsp.2"/ to /.cpu_dai_name = "omap-mcbsp.3"/ and i adjusted the codec as slave (/SND_SOC_DAIFMT_CBS_CFS/) to test whether the McBSP3 setup is right (the PCM3168 codec was disconnected). I still got a clock signal on the McBSP2 but the McBSP2_DX had no signal anymore and on the McBSP3 i did not get any signal at all (measured with an osziloscope). I am a little bit confused as i am not shure why there is still the clk signal on the McBSP2 and furthermore i am not shure whether i test the McBSP3 properly or whether there is a proplem in the McBSP3 setup. The PinMUX for the McBSP3 is done in the kernel (/board-omap3beagle.c/) by setting: static void __init omap3_beagle_config_mcbsp3_mux(void) { omap_mux_init_signal("mcbsp3_fsx.mcbsp3_fsx", OMAP_PIN_INPUT); omap_mux_init_signal("uart2_cts.mcbsp3_dx", OMAP_PIN_OUTPUT); omap_mux_init_signal("uart2_rts.mcbsp3_dr", OMAP_PIN_INPUT); /* NOTE: Clock pins need to be in input mode */ omap_mux_init_signal("uart2_tx.mcbsp3_clkx", OMAP_PIN_INPUT); } It would be really nice if you could help me. Thanks Wendelin ****************************************************************************************************** * kernel_3.7.4+/sound/soc/omap/omap-pcm3168.c * ****************************************************************************************************** /* * PCM3168 ASoC driver for BeagleBoard. * * based on: * > omap3beagle.c -- SoC audio for OMAP3 Beagle * > Author: Steve Sakoman <steve(a)sakoman.com> * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" /* * Uncomment to test codec in slave mode or without actual codec. This makes * possible to test this driver by letting the OMAP to be DAI link master */ #define PCM3168_CODEC_SLAVE 1 static int pcm3168_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int fmt, div; int ret; #ifdef PCM3168_CODEC_SLAVE fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF; #else fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF; #endif /* Set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set codec DAI configuration - pcm3168\n"); return ret; } /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration - pcm3168\n"); return ret; } #ifdef PCM3168_CODEC_SLAVE ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 48000000, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("can't set McBSP sysclk - pcm3168\n"); return ret; } /* * Calculate McBSP SRG divisor in McBSP master mode */ div = 48000000 / params_rate(params) / params_channels(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: div /= 16; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: div /= 32; break; }; /* * Round to maximum divisor if needed. This means that extra bit-clock * cycles are transmitted when sample rate and number of bits in frame * (channels * sample bits) are low. */ if (div >= 256) div = 256; ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, div); if (ret < 0) { pr_err("can't set SRG clock divider - pcm3168\n"); return ret; } #endif return 0; } static struct snd_soc_ops pcm3168_ops = { .hw_params = pcm3168_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link pcm3168_dai = { .name = "PCM3168", .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp.3", .platform_name = "omap-pcm-audio", .codec_dai_name = "pcm3168-hifi", .codec_name = "pcm3168-codec.0", .ops = &pcm3168_ops, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_pcm3168 = { .name = "pcm3168", .owner = THIS_MODULE, .dai_link = &pcm3168_dai, .num_links = 1, }; struct platform_device pcm3168_codec = { .name = "pcm3168-codec", .id = 0, }; struct platform_device pcm3168_soc_audio = { .name = "pcm3168-soc-audio", .id = 0, }; static int __devinit pcm3168_soc_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_pcm3168; int ret; pr_info("OMAP3 Beagle - PCM3168 ASoC init\n"); card->dev = &pdev->dev; ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d - pcm3168\n", ret); return ret; } return 0; } static int __devexit pcm3168_soc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static struct platform_driver pcm3168_driver = { .driver = { .name = "pcm3168-soc-audio", .owner = THIS_MODULE, }, .probe = pcm3168_soc_probe, .remove = __devexit_p(pcm3168_soc_remove), }; static int __init pcm3168_soc_init(void) { platform_device_register(&pcm3168_codec); platform_device_register(&pcm3168_soc_audio); return platform_driver_register(&pcm3168_driver); } module_init(pcm3168_soc_init); static void __exit pcm3168_soc_exit(void) { platform_driver_unregister(&pcm3168_driver); platform_device_unregister(&pcm3168_soc_audio); platform_device_unregister(&pcm3168_codec); } module_exit(pcm3168_soc_exit); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:pcm3168-soc-audio"); ****************************************************************************************************** * kernel_3.7.4+/sound/soc/codecs/pcm3168.c * ****************************************************************************************************** /* * ALSA Soc PCM3168 codec support * * Author: Klimann Wendelin <wklimann(a)hotmail.com> * * based on: * > pcm3008.c * > Author: Hugo Villeneuve * > Copyright (C) 2008 Lyrtech inc * > * > Based on AC97 Soc codec, original copyright follow: * > Copyright 2005 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA */ #include <linux/init.h> #include <linux/kernel.h> #include <linux/device.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> //#include <sound/soc-dapm.h> #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <plat/mcbsp.h> #include "../omap/omap-mcbsp.h" #include "../omap/omap-pcm.h" #include "pcm3168.h" #define PCM3168_VERSION "0.1" #define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000)*/ #define PCM3168_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver pcm3168_dai = { .name = "pcm3168-hifi", .playback = { .stream_name = "PCM3168 Playback", .channels_min = 2, .channels_max = 8, .rates = PCM3168_RATES, .formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE .sig_bits = 24, }, .capture = { .stream_name = "PCM3168 Capture", .channels_min = 2, .channels_max = 6, .rates = PCM3168_RATES, .formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE .sig_bits = 24, }, }; static void pcm3168_gpio_free(struct pcm3168_setup_data *setup) { // gpio_free(setup->dem0_pin); // gpio_free(setup->dem1_pin); // gpio_free(setup->pdad_pin); // gpio_free(setup->pdda_pin); } static int pcm3168_soc_probe(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; int ret = 0; printk(KERN_ALERT "in SOC Probe codec -> kli \n"); printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION); return ret; gpio_err: // pcm3168_gpio_free(setup); return ret; } static int pcm3168_soc_remove(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; // pcm3168_gpio_free(setup); return 0; } #define pcm3168_soc_suspend NULL #define pcm3168_soc_resume NULL static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = { .probe = pcm3168_soc_probe, .remove = pcm3168_soc_remove, .suspend = pcm3168_soc_suspend, .resume = pcm3168_soc_resume, }; static int __devinit pcm3168_codec_probe(struct platform_device *pdev) { int ret; printk(KERN_ALERT "probe pcm3168 codec -> kli \n"); ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3168, &pcm3168_dai, 1); printk(KERN_ALERT "probe_after pcm3168 codec -> kli = %d \n", ret); return ret; } static int __devexit pcm3168_codec_remove(struct platform_device *pdev) { printk(KERN_ALERT "remove pcm3168 codec -> kli \n"); snd_soc_unregister_codec(&pdev->dev); return 0; } MODULE_ALIAS("platform:pcm3168-codec"); static struct platform_driver pcm3168_codec_driver = { .probe = pcm3168_codec_probe, .remove = __devexit_p(pcm3168_codec_remove), .driver = { .name = "pcm3168-codec", .owner = THIS_MODULE, }, }; static int __init pcm3168_modinit(void) { printk(KERN_ALERT "in init of 3168 -> kli \n"); return platform_driver_register(&pcm3168_codec_driver); } module_init(pcm3168_modinit); static void __exit pcm3168_exit(void) { printk(KERN_ALERT "in exit of 3168 -> kli \n"); platform_driver_unregister(&pcm3168_codec_driver); } module_exit(pcm3168_exit); MODULE_DESCRIPTION("Soc PCM3168 driver"); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_LICENSE("GPL");

2 5

[alsa-devel] [PATCH] ALSA: hda - fix typo in proc output
by David Henningsson 04 Apr '13

04 Apr '13

Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..e37b738 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg) "Line Out", "Speaker", "HP Out", "CD", "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" + "SPDIF In", "Digital In", "Reserved", "Other" }; return jack_types[(cfg & AC_DEFCFG_DEVICE) -- 1.7.9.5

2 1

[alsa-devel] [PATCHv2 00/10] UAC2: Automatic clock switching
by Eldad Zack 04 Apr '13

04 Apr '13

This is version 2 of the automatic clock switching. v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-March/060734.html No patches added or removed. Rebased patches with "ALSA: usb: Work around CM6631 sample rate change bug", commit 690a863ff03d9a29ace2b752b8f802fba78a842f . Clemens, Takashi and Torstein, thank you very much for reviewing and providing comments! Changes: * Fixed formatting * (#5) Fixed type cast: u32 -> __l32 * (#6) Added rational to commit message * (#7) Fixed assignment of int to unsigned char, return error code if encountered * (#8) Fixed the parameter name, annotated the default value * (#10) Avoid second read of frequency for read-only clocks Tested with Takashi's for-next. Applies against Takashi's for-next HEAD efc33ce197e4b6aaddf1eb2ba6293f51daf3c283 . Cheers, Eldad Zack (10): ALSA: usb-audio: convert list_for_each to entry variant ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement ALSA: usb-audio: neaten EXPORT_SYMBOLS placement ALSA: usb-audio: spelling correction ALSA: usb-audio: use endianness macros ALSA: usb-audio: UAC2: do clock validity check earlier ALSA: usb-audio: UAC2: try to find and switch to valid clock ALSA: usb-audio: UAC2: auto clock selection module param ALSA: usb-audio: show err in set_sample_rate_v2 debug ALSA: usb-audio: UAC2: support read-only freq control sound/usb/card.c | 11 ++-- sound/usb/clock.c | 153 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/clock.h | 3 +- sound/usb/endpoint.c | 18 +++--- sound/usb/endpoint.h | 2 +- sound/usb/format.c | 2 +- sound/usb/midi.c | 12 ++-- sound/usb/pcm.c | 32 ++++------- sound/usb/proc.c | 7 +-- sound/usb/stream.c | 12 ++-- sound/usb/usbaudio.h | 1 + 11 files changed, 157 insertions(+), 96 deletions(-) -- 1.8.1.5

2 11

[alsa-devel] [PATCH] Enabling Realtek ALC 671 codec
by Rainer Koenig 04 Apr '13

04 Apr '13

Changes: * Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig(a)ts.fujitsu.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7743775..fcbf4c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6663,7 +6663,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -7115,6 +7116,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, -- Dipl.-Inf. (FH) Rainer Koenig Project Manager Linux Clients Dept. PDG WPS R&D SW OSE Fujitsu Technology Solutions Bürgermeister-Ullrich-Str. 100 86199 Augsburg Germany Telephone: +49-821-804-3321 Telefax: +49-821-804-2131 Mail: mailto:Rainer.Koenig@ts.fujitsu.com Internet ts.fujtsu.com Company Details ts.fujitsu.com/imprint.html

2 1

[alsa-devel] How to use implicit feedback with full duplex?
by Clemens Ladisch 04 Apr '13

04 Apr '13

Hi, I thought I'd try to use implicit feedback with my simple audio device: Interface Descriptor: bInterfaceNumber 1 bAlternateSetting 0 Interface Descriptor: bInterfaceNumber 1 bAlternateSetting 1 AudioStreaming Interface Descriptor: ... Endpoint Descriptor: bEndpointAddress 0x01 EP 1 OUT Interface Descriptor: bInterfaceNumber 2 bAlternateSetting 0 Interface Descriptor: bInterfaceNumber 2 bAlternateSetting 1 AudioStreaming Interface Descriptor: ... Endpoint Descriptor: bEndpointAddress 0x82 EP 2 IN So I wrote a patch to configure it like those M-Audio devices: --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -391,6 +391,17 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; switch (subs->stream->chip->usb_id) { + case USB_ID(0x0582, 0x0018): + if (is_playback) { + implicit_fb = 1; + ep = 0x82; + iface = usb_ifnum_to_if(dev, 2); + if (!iface || iface->num_altsetting < 2) + return -EINVAL; + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ if (is_playback) { implicit_fb = 1; This works fine when playing something: kernel: setting usb interface 1:1 kernel: Creating new playback data endpoint #1 kernel: Creating new capture data endpoint #82 kernel: Setting params for ep #1 (type 0, 8 urbs), ret=0 kernel: match_endpoint_audioformats: (fmt @ffff8801fefa8900) score 2 kernel: Setting params for ep #82 (type 0, 8 urbs), ret=0 kernel: Starting data EP @ffff8801fd4e8000 kernel: Starting sync EP @ffff8801fd2ac000 But when I then try to record at the same time, the driver refuses to configure the input endpoint (to the only format, which is already set): kernel: setting usb interface 2:1 kernel: Re-using EP 82 in iface 2,1 @ffff8801fd2ac000 kernel: Unable to change format on ep #82: already in use And despite that "alreay in use" check, the input endpoint is affected so much that playback breaks. Is full duplex supposed to work? Does it work with other devices? Regards, Clemens

5 11

[alsa-devel] [PATCH v5] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 04 Apr '13

04 Apr '13

The C-Media CM6631 USB receiver doesn't respond to changes in sample rate while the interface is active. The same behavior is observed in other UAC2 hardware like the VIA VT1731. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all UAC2 devices, as it should not affect a standards compliant device, but it is only necessary for C-Media CM6631, VIA VT1731 and possibly others. Failure to read sample rate from the device is not handled as an error in set_sample_rate_v2(), as (permanent or intermittent) failure to read sample rate isn't essential for a successful sample rate set. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- Changes since v4: - Sample rate read failure is no longer fatal. Failure to read sample rate from the device is degraded from a fatal error to a warning message. The Schiit Bifrost has shown intermittent sample rate read failures, and this shouldn't be a reason to abort the initialization. - The reset is performed for all UAC2 devices. The number of devices affected by this issue seems to be larger than just the C-Media based devices, issues with the VIA VT1731 has been reported by Davor Herga to alsa-user [1]. The additional reset should not affect standards compliant devices and will hopefully improve the compatibility with future devices tested with the OS X UAC2 driver. This can cause problems for a theoretical device where sample rate read always fails, causing the reset to be performed every time the sample rate is set, *and* where repeated resets cause resume after pause to not resume properly, like the issues seen with CM6631 in the first iteration of this patch. [1] http://thread.gmane.org/gmane.linux.alsa.user/37312 Changes since v3: - Use given target rate instead of the new rate read from device when checking if a reset is needed. Some devices (at least Schiit Bifrost) doesn't report a reliable sample rate back right after setting a new rate. This appears to fix the issues with v3 of this patch with the Schiit, as reported by Chris Hermansen in http://thread.gmane.org/gmane.linux.alsa.user/36935/focus=37284 sound/usb/clock.c | 45 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 35 insertions(+), 10 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..9e2703a 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + prev_rate = 0; + } else { + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + } + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); - return err; + cur_rate = 0; + } else { + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + } return 0; } -- 1.7.10.4

5 8

Re: [alsa-devel] [PATCH 1/2] ASoC: Samsung: return error if drvdata is not set
by Mark Brown 03 Apr '13

03 Apr '13

On Tue, Apr 02, 2013 at 04:53:01PM +0530, Padmavathi Venna wrote: > From: Prathyush K <prathyush.k(a)samsung.com> > > This patch fixes a possible crash in case drvdata for the secondary > device is not set. Applied both, thanks.

1 0

[alsa-devel] [PATCH] Enable Realtek ALC671 Codec
by Rainer Koenig 03 Apr '13

03 Apr '13

Hi, working on a new mainboard generation with ALC671 codecs the first tests showed that the generic codec parser is just configuring the rear mic in for audio input, line-in and front-mic are missing. I tried to understand the code of patch_realtek.c and after compiling with CONFIG_SND_DEBUG I saw that this module is not used at all for the ALC671 codec because its device-ID was missing. So what I did is: - adding the device id and assigned the ALC662 patches for it. - Added the device id so that it enables 4 ports in alc662_parse_auto_config - changed the conditional in alc_subsystem_id because it looked so strange to me when I tried to understand what is happening. :-) Here is the patch: Signed-off-by: Rainer Koenig <Rainer.Koenig(a)ts.fujitsu.com> --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1283,7 +1283,7 @@ static int alc_subsystem_id(struct hda_codec *codec, } ass = codec->subsystem_id & 0xffff; - if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) + if ((ass == codec->bus->pci->subsystem_device) && (ass & 1)) goto do_sku; /* invalid SSID, check the special NID pin defcfg instead */ @@ -6663,7 +6663,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -7115,6 +7116,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, -- Dipl.-Inf. (FH) Rainer Koenig Project Manager Linux Clients Dept. PDG WPS R&D SW OSE Fujitsu Technology Solutions Bürgermeister-Ullrich-Str. 100 86199 Augsburg Germany Telephone: +49-821-804-3321 Telefax: +49-821-804-2131 Mail: mailto:Rainer.Koenig@ts.fujitsu.com Internet ts.fujtsu.com Company Details ts.fujitsu.com/imprint.html

2 5

Re: [alsa-devel] [PATCH] sound: soc: tegra: do not provide PAUSE/RESUME in hardware info
by Lars-Peter Clausen 03 Apr '13

03 Apr '13

On 04/03/2013 11:19 AM, Laxman Dewangan wrote: > Tegra PCM driver uses the dmaengine based DMA driver and Tegra DMA > driver does not support the pause/resume functionality as hardware > does not support it. > > ALSA driver support the sw based pause/resume functionality if hardware > does not support it and in this case, hw driver should not provide > the hardware info with PAUSE/RESUME bit enable. > > Remmoving the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME from hardware > info and using the snd_dmaengine_pcm_trigger() for pcm_trigger callback. > > Signed-off-by: Laxman Dewangan <ldewangan(a)nvidia.com> Hm, I sent out the same patch just half an our or so ago. - Lars > --- > Lars reported that if driver/chip dma driver does not support the pause/resume > then need not to export this info through the hw capability and leave this > implementation to alsa driver. > > Making the change accordingly. > > sound/soc/tegra/tegra_pcm.c | 24 +----------------------- > 1 files changed, 1 insertions(+), 23 deletions(-) > > diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c > index e67af0b..117cccc 100644 > --- a/sound/soc/tegra/tegra_pcm.c > +++ b/sound/soc/tegra/tegra_pcm.c > @@ -43,8 +43,6 @@ > static const struct snd_pcm_hardware tegra_pcm_hardware = { > .info = SNDRV_PCM_INFO_MMAP | > SNDRV_PCM_INFO_MMAP_VALID | > - SNDRV_PCM_INFO_PAUSE | > - SNDRV_PCM_INFO_RESUME | > SNDRV_PCM_INFO_INTERLEAVED, > .formats = SNDRV_PCM_FMTBIT_S16_LE, > .channels_min = 2, > @@ -121,26 +119,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) > return 0; > } > > -static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > -{ > - switch (cmd) { > - case SNDRV_PCM_TRIGGER_START: > - case SNDRV_PCM_TRIGGER_RESUME: > - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > - return snd_dmaengine_pcm_trigger(substream, > - SNDRV_PCM_TRIGGER_START); > - > - case SNDRV_PCM_TRIGGER_STOP: > - case SNDRV_PCM_TRIGGER_SUSPEND: > - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > - return snd_dmaengine_pcm_trigger(substream, > - SNDRV_PCM_TRIGGER_STOP); > - default: > - return -EINVAL; > - } > - return 0; > -} > - > static int tegra_pcm_mmap(struct snd_pcm_substream *substream, > struct vm_area_struct *vma) > { > @@ -158,7 +136,7 @@ static struct snd_pcm_ops tegra_pcm_ops = { > .ioctl = snd_pcm_lib_ioctl, > .hw_params = tegra_pcm_hw_params, > .hw_free = tegra_pcm_hw_free, > - .trigger = tegra_pcm_trigger, > + .trigger = snd_dmaengine_pcm_trigger, > .pointer = snd_dmaengine_pcm_pointer, > .mmap = tegra_pcm_mmap, > };

1 0

[alsa-devel] [PATCH] salsa-lib: Add seq_midi_event.h
by Mihail Zenkov 03 Apr '13

03 Apr '13

--- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/configure.ac +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/configure.ac @@ -273,6 +273,7 @@ test "$sndseq" = "yes" && echo "#include \"seq_event.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seq.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seqmid.h\"" >> src/asoundlib.h +test "$sndseq" = "yes" && echo "#include \"seq_midi_event.h\"" >> src/asoundlib.h cat "$srcdir"/src/asoundlib-tail.h >> src/asoundlib.h echo "SALSA-Lib build configuration:" --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/Makefile.am +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/Makefile.am @@ -60,7 +60,7 @@ endif if BUILD_SEQ alsainclude_HEADERS += \ - seq.h seq_event.h seqmid.h + seq.h seq_event.h seqmid.h seq_midi_event.h endif noinst_HEADERS = local.h --- /dev/null +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/seq_midi_event.h @@ -0,0 +1,78 @@ +#ifndef __ALSA_SEQ_MIDI_EVENT_H +#define __ALSA_SEQ_MIDI_EVENT_H + +#ifdef __cplusplus +extern "C" { +#endif + +/** + * \defgroup MIDI_Event Sequencer event <-> MIDI byte stream coder + * \ingroup Sequencer + * Sequencer event <-> MIDI byte stream coder + * \{ + */ + +/** container for sequencer midi event parsers */ +typedef struct snd_midi_event { + size_t qlen; + size_t read; + int type; + unsigned char lastcmd; + unsigned char nostat; + size_t bufsize; + unsigned char *buf; +} snd_midi_event_t; + +__SALSA_EXPORT_FUNC +int snd_midi_event_new(size_t bufsize, snd_midi_event_t **rdev) +{ + return 0; +} +__SALSA_EXPORT_FUNC +int snd_midi_event_resize_buffer(snd_midi_event_t *dev, size_t bufsize) +{ + return 0; +} +__SALSA_EXPORT_FUNC +void snd_midi_event_free(snd_midi_event_t *dev) +{ +} +__SALSA_EXPORT_FUNC +void snd_midi_event_init(snd_midi_event_t *dev) +{ +} +__SALSA_EXPORT_FUNC +void snd_midi_event_reset_encode(snd_midi_event_t *dev) +{ +} +__SALSA_EXPORT_FUNC +void snd_midi_event_reset_decode(snd_midi_event_t *dev) +{ +} +__SALSA_EXPORT_FUNC +void snd_midi_event_no_status(snd_midi_event_t *dev, int on) +{ +} +__SALSA_EXPORT_FUNC +long snd_midi_event_encode(snd_midi_event_t *dev, const unsigned char *buf, long count, snd_seq_event_t *ev) +{ + return 0; +} +__SALSA_EXPORT_FUNC +int snd_midi_event_encode_byte(snd_midi_event_t *dev, int c, snd_seq_event_t *ev) +{ + return 0; +} +__SALSA_EXPORT_FUNC +long snd_midi_event_decode(snd_midi_event_t *dev, unsigned char *buf, long count, const snd_seq_event_t *ev) +{ + return 0; +} + + +#ifdef __cplusplus +} +#endif + +#endif /* __ALSA_SEQ_MIDI_EVENT_H */ +

2 6

[alsa-devel] [PATCH] salsa-lib: Fix error on jack compilation with salsa
by Mihail Zenkov 03 Apr '13

03 Apr '13

Fix error on jack compilation with salsa: /usr/include/alsa/ctl_macros.h:251:48: error: invalid conversion from 'void*' to 'snd_ctl_elem_id_t* {aka snd_ctl_elem_id*}' [-fpermissive] --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/ctl_macros.h +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/ctl_macros.h @@ -248,7 +248,7 @@ unsigned int entries) { free(obj->pids); - obj->pids = calloc(entries, sizeof(*obj->pids)); + obj->pids = (snd_ctl_elem_id_t*)calloc(entries, sizeof(*obj->pids)); if (!obj->pids) { obj->space = 0; return -ENOMEM;

3 5

[alsa-devel] [PATCH 0/4] ALSA: Updates for the CA0132 codec
by Ian Minett 03 Apr '13

03 Apr '13

From: Ian Minett <ian_minett(a)creativelabs.com> Hi, This series contains the following updates to CA0132: 1: Add firmware loading for the SpeakerEQ DSP effect. 2: Improve DSP transfer timeout calculations. 3: Add stream and effect latency monitoring routines 4: Update hp unsolicited event handler to manage queued work. These were based on the latest sound.git, but please let me know if they should be against sound-unstable tree instead. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com>

3 19

[alsa-devel] Audio capture on embedded system (via USB port)
by Israel Alejandro Arriaga Trejo 03 Apr '13

03 Apr '13

Hello, I am writing to kindly ask for assistance with ALSA on an embedded platform, in particular to record audio via a USB headset. The embedded platform I am currently using has analog inputs and outputs for audio (mic and headphone). So far I have been able to play and capture audio using the analog audio inputs/outputs with the commands aplay and arecord respectively from ALSA without any problems. However this is not the case when I am trying to use a USB headset via the USB port, at least while trying to record audio (playing audio is not a problem, it works perfectly fine for different rates and formats and the devices are recognized when I type aplay –l or arecord -l). However, when I try to capture audio, using the ALSA command line arecord as in: $ arecord –d 10 –r 16000 –f S16_LE –t wav –Dhw:1,0 test.wav even though the following message appears: $ Recording WAVE ‘test.wav’ : Signed 16 bit Little Endian, Rate 16000 Hz, Mono the application seems “to freeze”. In fact, the recording does not stop in the indicated 10 seconds, instead it continues “working” for a long period of time and creates the file test.wav which after is played it contains fragments of what actually was recorded via the microphone nonetheless, its content is unintelligible. In some cases the following error also appears: $ arecord: pcm_read:1617: read error: Input/Output error As mentioned previously this behavior is only observed during capture mode (while attempting to record using the microphone) since the playback mode using the USB headphone works properly. After browsing for a while I have found similar situations reported, however I have not been able to find a solution to my problem. I would highly appreciate for any hint that could help me to solve this issue. Best regards, Alejandro

2 1

Re: [alsa-devel] [PATCH 1/2] dmaengine: add const for name parameter
by Vinod Koul 02 Apr '13

02 Apr '13

On Fri, Mar 15, 2013 at 09:53:46AM +0100, Markus Pargmann wrote: > Hi, > > I already sent a nearly identical patch: > https://patchwork.kernel.org/patch/2179321/ I applied that one a while ago so Markus can you check my tree and see if anything is remaining, looks most of cases are covered > > Regards, > > Markus > > On Fri, Mar 15, 2013 at 11:36:40AM +0800, Shawn Guo wrote: > > Mark name parameter of generic device tree helpers as const, as it > > shouldn't be changed anyway. > > > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > > --- > > drivers/dma/dmaengine.c | 2 +- > > drivers/dma/of-dma.c | 6 +++--- > > include/linux/dmaengine.h | 7 ++++--- > > include/linux/of_dma.h | 2 +- > > 4 files changed, 9 insertions(+), 8 deletions(-) > > > > diff --git a/drivers/dma/dmaengine.c b/drivers/dma/dmaengine.c > > index b2728d6..2cbfefe 100644 > > --- a/drivers/dma/dmaengine.c > > +++ b/drivers/dma/dmaengine.c > > @@ -555,7 +555,7 @@ EXPORT_SYMBOL_GPL(__dma_request_channel); > > * @dev: pointer to client device structure > > * @name: slave channel name > > */ > > -struct dma_chan *dma_request_slave_channel(struct device *dev, char *name) > > +struct dma_chan *dma_request_slave_channel(struct device *dev, const char *name) > > { > > /* If device-tree is present get slave info from here */ > > if (dev->of_node) > > diff --git a/drivers/dma/of-dma.c b/drivers/dma/of-dma.c > > index 69d04d2..6036cd0 100644 > > --- a/drivers/dma/of-dma.c > > +++ b/drivers/dma/of-dma.c > > @@ -172,8 +172,8 @@ EXPORT_SYMBOL_GPL(of_dma_controller_free); > > * specifiers, matches the name provided. Returns 0 if the name matches and > > * a valid pointer to the DMA specifier is found. Otherwise returns -ENODEV. > > */ > > -static int of_dma_match_channel(struct device_node *np, char *name, int index, > > - struct of_phandle_args *dma_spec) > > +static int of_dma_match_channel(struct device_node *np, const char *name, > > + int index, struct of_phandle_args *dma_spec) > > { > > const char *s; > > > > @@ -198,7 +198,7 @@ static int of_dma_match_channel(struct device_node *np, char *name, int index, > > * Returns pointer to appropriate dma channel on success or NULL on error. > > */ > > struct dma_chan *of_dma_request_slave_channel(struct device_node *np, > > - char *name) > > + const char *name) > > { > > struct of_phandle_args dma_spec; > > struct of_dma *ofdma; > > diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h > > index 91ac8da..68b4252 100644 > > --- a/include/linux/dmaengine.h > > +++ b/include/linux/dmaengine.h > > @@ -968,7 +968,8 @@ enum dma_status dma_sync_wait(struct dma_chan *chan, dma_cookie_t cookie); > > enum dma_status dma_wait_for_async_tx(struct dma_async_tx_descriptor *tx); > > void dma_issue_pending_all(void); > > struct dma_chan *__dma_request_channel(dma_cap_mask_t *mask, dma_filter_fn fn, void *fn_param); > > -struct dma_chan *dma_request_slave_channel(struct device *dev, char *name); > > +struct dma_chan *dma_request_slave_channel(struct device *dev, > > + const char *name); > > void dma_release_channel(struct dma_chan *chan); > > #else > > static inline enum dma_status dma_wait_for_async_tx(struct dma_async_tx_descriptor *tx) > > @@ -984,7 +985,7 @@ static inline struct dma_chan *__dma_request_channel(dma_cap_mask_t *mask, > > return NULL; > > } > > static inline struct dma_chan *dma_request_slave_channel(struct device *dev, > > - char *name) > > + const char *name) > > { > > return NULL; > > } > > @@ -1007,7 +1008,7 @@ struct dma_chan *net_dma_find_channel(void); > > static inline struct dma_chan > > *__dma_request_slave_channel_compat(dma_cap_mask_t *mask, dma_filter_fn fn, > > void *fn_param, struct device *dev, > > - char *name) > > + const char *name) > > { > > struct dma_chan *chan; > > > > diff --git a/include/linux/of_dma.h b/include/linux/of_dma.h > > index d15073e..0c4a82e 100644 > > --- a/include/linux/of_dma.h > > +++ b/include/linux/of_dma.h > > @@ -40,7 +40,7 @@ extern int of_dma_controller_register(struct device_node *np, > > void *data); > > extern int of_dma_controller_free(struct device_node *np); > > extern struct dma_chan *of_dma_request_slave_channel(struct device_node *np, > > - char *name); > > + const char *name); > > extern struct dma_chan *of_dma_simple_xlate(struct of_phandle_args *dma_spec, > > struct of_dma *ofdma); > > #else > > -- > > 1.7.9.5 > > > > > > > > -- > Pengutronix e.K. | | > Industrial Linux Solutions | http://www.pengutronix.de/ | > Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | > Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

4 6

[alsa-devel] BUG: unable to handle kernel NULL pointer dereference at 0000000000000010, IP: [<ffffffff819865fb>] snd_info_get_line+0x3b/0xf0
by Tommi Rantala 02 Apr '13

02 Apr '13

Hello, Saw this while fuzzing with trinity in a qemu virtual machine: [6026] Random reseed: 625996896 [watchdog] 7640111 iterations. [F:6650299 S:989808] [6026] Random reseed: 472197910 [ 459.365417] misc rfkill: uevent: unknown action-string [ 459.521953] BUG: unable to handle kernel NULL pointer dereference at 0000000000000010 [ 459.522027] IP: [<ffffffff819865fb>] snd_info_get_line+0x3b/0xf0 [ 459.522027] PGD 52deb067 PUD 50bbc067 PMD 0 [ 459.522027] Oops: 0000 [#1] SMP [ 459.522027] CPU 0 [ 459.522027] Pid: 7781, comm: trinity-child53 Tainted: G W 3.9.0-rc1+ #103 Bochs Bochs [ 459.522027] RIP: 0010:[<ffffffff819865fb>] [<ffffffff819865fb>] snd_info_get_line+0x3b/0xf0 [ 459.522027] RSP: 0018:ffff880065455a88 EFLAGS: 00010202 [ 459.522027] RAX: 0000000000000001 RBX: ffff8800508e1760 RCX: 0000000000000000 [ 459.522027] RDX: ffff880065455b57 RSI: ffff880065455ad8 RDI: ffff8800599ceba0 [ 459.522027] RBP: ffff880065455a88 R08: 0000000000000010 R09: 0000000000000001 [ 459.522027] R10: ffff8800461f5630 R11: 0000000000000000 R12: ffff88007a470960 [ 459.522027] R13: ffff88007a43b980 R14: ffff88007a470000 R15: ffff88007a43b900 [ 459.522027] FS: 00007f9c07c94700(0000) GS:ffff88007f800000(0000) knlGS:0000000000000000 [ 459.522027] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 459.522027] CR2: 0000000000000010 CR3: 0000000060ee2000 CR4: 00000000000006f0 [ 459.522027] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 459.522027] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 459.522027] Process trinity-child53 (pid: 7781, threadinfo ffff880065454000, task ffff8800797c2290) [ 459.522027] Stack: [ 459.522027] ffff880065455b88 ffffffff819a3f7e ffff880065455ab8 ffff8800599ceba0 [ 459.522027] ffff8800508e1220 ffff8800508e1200 ffff880065455b48 ffffffff8136eafc [ 459.522027] ffff880065455ae8 ffffffff81091198 ffff88007cc01600 ffff8800508e1200 [ 459.522027] Call Trace: [ 459.522027] [<ffffffff819a3f7e>] snd_mixer_oss_proc_write+0x1ee/0x250 [ 459.522027] [<ffffffff8136eafc>] ? debug_check_no_obj_freed+0x17c/0x230 [ 459.522027] [<ffffffff81091198>] ? kmemcheck_shadow_lookup+0x28/0x80 [ 459.522027] [<ffffffff81985a38>] ? snd_info_entry_release+0x48/0xd0 [ 459.522027] [<ffffffff81199c0c>] ? kfree+0x1ac/0x1e0 [ 459.522027] [<ffffffff81985a50>] snd_info_entry_release+0x60/0xd0 [ 459.522027] [<ffffffff81211022>] proc_reg_release+0x132/0x180 [ 459.522027] [<ffffffff819859f0>] ? snd_info_entry_llseek+0x100/0x100 [ 459.522027] [<ffffffff811a807a>] __fput+0x11a/0x220 [ 459.522027] [<ffffffff811a8189>] ____fput+0x9/0x10 [ 459.522027] [<ffffffff810bcbbe>] task_work_run+0x9e/0xc0 [ 459.522027] [<ffffffff8109d85c>] do_exit+0x3ac/0xad0 [ 459.522027] [<ffffffff810afc29>] ? get_signal_to_deliver+0xf9/0x810 [ 459.522027] [<ffffffff8109e044>] do_group_exit+0x84/0xd0 [ 459.522027] [<ffffffff810b0269>] get_signal_to_deliver+0x739/0x810 [ 459.522027] [<ffffffff810583b2>] do_signal+0x52/0x570 [ 459.522027] [<ffffffff810cbaf3>] ? finish_task_switch+0x43/0x120 [ 459.522027] [<ffffffff810f3366>] ? trace_hardirqs_on_caller+0x16/0x1f0 [ 459.522027] [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [ 459.522027] [<ffffffff81fad85b>] ? _raw_spin_unlock_irq+0x2b/0x40 [ 459.522027] [<ffffffff810cbb30>] ? finish_task_switch+0x80/0x120 [ 459.522027] [<ffffffff810cbaf3>] ? finish_task_switch+0x43/0x120 [ 459.522027] [<ffffffff810588f7>] do_notify_resume+0x27/0x70 [ 459.522027] [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [ 459.522027] [<ffffffff81faf0a2>] int_signal+0x12/0x17 [ 459.522027] Code: 00 8b 4f 14 85 c9 0f 85 cf 00 00 00 44 8b 5f 18 45 85 db 74 6b e9 c1 00 00 00 66 0f 1f 44 00 00 8b 47 08 4c 8b 07 89 c1 83 c0 01 <41> 0f b6 0c 08 89 47 08 44 0f be c1 41 83 f8 0a 75 2b 39 47 0c [ 459.522027] RIP [<ffffffff819865fb>] snd_info_get_line+0x3b/0xf0 [ 459.522027] RSP <ffff880065455a88> [ 459.522027] CR2: 0000000000000010 [ 459.652998] ---[ end trace a9ded493128a1bce ]--- [ 459.654880] Fixing recursive fault but reboot is needed! [6026] Random reseed: 807260652 [watchdog] pid 8469 hasn't made progress in 30 seconds! (last:2725563185 now:2725563215 diff:30). Stuck in syscall 70:msgrcv. Sending SIGKILL. [watchdog] 7677069 iterations. [F:6682605 S:994461] [6026] Random reseed: 575363686 [ 461.032969] misc ashmem: uevent: unknown action-string [6026] Random reseed: 1095114151 [6026] Regenerating random pages, fd's etc. Tommi

2 1

[alsa-devel] How to create and verify verb tables for coreboot?
by Paul Menzel 02 Apr '13

02 Apr '13

Dear ALSA folks, coreboot is a free replacement for the traditional BIOS [1]. Therefore coreboot is also responsible for providing the correct verb table [2]. According to Jens the Verb Table format is not documented and is created by for example Realtek [3]. Could you give some more information about the verb table and how this can be created from scratch? Our developer Rudolf suggests to take the values from /sys/class/sound/hwC1D0/init_pin_configs when running the vendor BIOS [2]. How can a correct verb table be verified? As for the ASRock E350M1 [4][5], I attach the output of `alsa-info.sh` [6] when running coreboot. The one running with the vendor BIOS is uploaded to the server [7] due to the size limit of this mailing list. $ wdiff vendorbios coreboot # init_pin_configs 0x11 [-0x411111f0-] {+0x411110f0+} 0x12 0x411111f0 0x14 [-0x01014010-] {+0x01014030+} 0x15 [-0x01011012-] {+0x01011031+} 0x16 [-0x01016011-] {+0x01016032+} 0x17 [-0x411111f0-] {+0x01012033+} 0x18 [-0x01a19840-] {+0x01a19850+} 0x19 [-0x02a19950-] {+0x02a19c80+} 0x1a [-0x0181304f-] {+0x01813051+} 0x1b [-0x02214120-] {+0x02214c40+} 0x1c [-0x411111f0-] {+0x9933105f+} 0x1d [-0x4005e601-] {+0x00000100+} 0x1e [-0x01452130-] {+0x01441070+} 0x1f [-0x411111f0-] {+0x41c46060+} Thanks, Paul [1] http://www.coreboot.org/ [2] http://www.coreboot.org/pipermail/coreboot/2013-March/075547.html [3] http://review.coreboot.org/#/c/2553/1/src/mainboard/lippert/frontrunner-af/… [4] http://www.coreboot.org/ASRock_E350M1 [5] http://www.coreboot.org/pipermail/coreboot/2013-March/075557.html [6] http://alsa-project.org/main/index.php/Help_To_Debug [7] http://www.alsa-project.org/db/?f=96ff75103a48ab8e58a216bd12f65d2755679cdc

2 1

[alsa-devel] [PATCH v2 2/2] ALSA: hda - bug fix on HDMI ELD debug message
by mengdong.lin＠intel.com 02 Apr '13

02 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..de8ac5c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, -- 1.7.10.4

3 2

[alsa-devel] [PATCH v2 1/2] ALSA: hda - bug fix on return value when getting HDMI ELD info
by mengdong.lin＠intel.com 02 Apr '13

02 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7dd8463..d0d7ac1 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { int i; - int ret; + int ret = 0; int size; /* -- 1.7.10.4

3 2

[alsa-devel] ASoC updates for v3.9
by Mark Brown 02 Apr '13

02 Apr '13

The following changes since commit 8bb9660418e05bb1845ac1a2428444d78e322cc7: Linux 3.9-rc4 (2013-03-23 16:52:44 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.9-rc4 for you to fetch changes up to 40bac28eb1e39f229a94ad72e2ca8cdeef3ba31a: Merge remote-tracking branch 'asoc/fix/spear' into asoc-next (2013-03-26 14:08:07 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.9 A few more fixes here and there, including quite a few nasty driver specific ones, but nothing that has a major general impact. ---------------------------------------------------------------- Axel Lin (1): ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case Joe Perches (1): ASoC:: max98090: Remove executable bit Lars-Peter Clausen (2): ASoC: spear_pcm: Update to new pcm_new() API ASoC: dma-sh7760: Fix compile error Mark Brown (9): ASoC: dapm: Fix handling of loops Merge remote-tracking branch 'asoc/fix/adsp' into asoc-next Merge remote-tracking branch 'asoc/fix/core' into asoc-next Merge remote-tracking branch 'asoc/fix/dapm' into asoc-next Merge remote-tracking branch 'asoc/fix/fsl' into asoc-next Merge remote-tracking branch 'asoc/fix/max98090' into asoc-next Merge remote-tracking branch 'asoc/fix/sh' into asoc-next Merge remote-tracking branch 'asoc/fix/si476x' into asoc-next Merge remote-tracking branch 'asoc/fix/spear' into asoc-next Markus Pargmann (1): ASoC: pcm030 audio fabric: remove __init from probe Peter Ujfalusi (1): ASoC: dapm: Fix pointer dereference in is_connected_output_ep() Sascha Hauer (1): ASoC: imx-ssi: Fix occasional AC97 reset failure Silviu-Mihai Popescu (1): ASoC: core: fix invalid free of devm_ allocated data Wei Yongjun (2): ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff() ASoC: core: fix possible memory leak in snd_soc_bytes_put() include/sound/soc-dapm.h | 1 + sound/soc/codecs/si476x.c | 1 + sound/soc/codecs/wm_adsp.c | 5 +++-- sound/soc/fsl/imx-ssi.c | 5 +++++ sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/sh/dma-sh7760.c | 4 ++-- sound/soc/soc-core.c | 8 ++++---- sound/soc/soc-dapm.c | 14 ++++++++++++++ sound/soc/spear/spear_pcm.c | 12 ++++++------ 9 files changed, 37 insertions(+), 15 deletions(-) mode change 100755 => 100644 include/sound/max98090.h mode change 100755 => 100644 sound/soc/codecs/max98090.c mode change 100755 => 100644 sound/soc/codecs/max98090.h

2 2

[alsa-devel] [PATCH] ALSA: hda/ca0132 - Delay HP amp turnon.
by Dylan Reid 02 Apr '13

02 Apr '13

From: Chih-Chung Chang <chihchung(a)chromium.org> Turing on the headphone amp interferes with the impedance measurement used to detect a TRRS style headset microphone. Delay the HP turn on until 500ms after the jack is detected, allowing the mic detection state machine to run to completion. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_ca0132.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0792b57..12eb21a 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -741,6 +741,9 @@ struct ca0132_spec { long voicefx_val; long cur_mic_boost; + struct hda_codec *codec; + struct delayed_work unsol_hp_work; + #ifdef ENABLE_TUNING_CONTROLS long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif @@ -3227,6 +3230,14 @@ exit: return err < 0 ? err : 0; } +static void ca0132_unsol_hp_delayed(struct work_struct *work) +{ + struct ca0132_spec *spec = container_of( + to_delayed_work(work), struct ca0132_spec, unsol_hp_work); + ca0132_select_out(spec->codec); + snd_hda_jack_report_sync(spec->codec); +} + static void ca0132_set_dmic(struct hda_codec *codec, int enable); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); @@ -4399,8 +4410,7 @@ static void ca0132_process_dsp_response(struct hda_codec *codec) static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res) { - snd_printdd(KERN_INFO "ca0132_unsol_event: 0x%x\n", res); - + struct ca0132_spec *spec = codec->spec; if (((res >> AC_UNSOL_RES_TAG_SHIFT) & 0x3f) == UNSOL_TAG_DSP) { ca0132_process_dsp_response(codec); @@ -4412,8 +4422,13 @@ static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res) switch (res) { case UNSOL_TAG_HP: - ca0132_select_out(codec); - snd_hda_jack_report_sync(codec); + /* Delay enabling the HP amp, to let the mic-detection + * state machine run. + */ + cancel_delayed_work_sync(&spec->unsol_hp_work); + queue_delayed_work(codec->bus->workq, + &spec->unsol_hp_work, + msecs_to_jiffies(500)); break; case UNSOL_TAG_AMIC1: ca0132_select_mic(codec); @@ -4588,6 +4603,7 @@ static void ca0132_free(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; + cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); snd_hda_sequence_write(codec, spec->base_exit_verbs); ca0132_exit_chip(codec); @@ -4653,6 +4669,7 @@ static int patch_ca0132(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + spec->codec = codec; spec->num_mixers = 1; spec->mixers[0] = ca0132_mixer; @@ -4663,6 +4680,8 @@ static int patch_ca0132(struct hda_codec *codec) spec->init_verbs[1] = ca0132_init_verbs1; spec->num_init_verbs = 2; + INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed); + ca0132_init_chip(codec); ca0132_config(codec); -- 1.8.1.3.605.g02339dd

2 1

Re: [alsa-devel] [PATCH] sound: oss: sb_common: Used kmemdup instead of kmalloc and memcpy
by Takashi Iwai 02 Apr '13

02 Apr '13

At Mon, 25 Mar 2013 15:50:34 +0200, Alexandru Gheorghiu wrote: > > Used kmemdup instead of replicating it's behaviour with kmalloc followed > by memcpy. > Patch found using coccinelle. > > Signed-off-by: Alexandru Gheorghiu <gheorghiuandru(a)gmail.com> Thanks, applied. Takashi > --- > sound/oss/sb_common.c | 3 +-- > 1 file changed, 1 insertion(+), 2 deletions(-) > > diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c > index 7d42c54..851a1da 100644 > --- a/sound/oss/sb_common.c > +++ b/sound/oss/sb_common.c > @@ -626,13 +626,12 @@ int sb_dsp_detect(struct address_info *hw_config, int pci, int pciio, struct sb_ > */ > > > - detected_devc = kmalloc(sizeof(sb_devc), GFP_KERNEL); > + detected_devc = kmemdup(devc, sizeof(sb_devc), GFP_KERNEL); > if (detected_devc == NULL) > { > printk(KERN_ERR "sb: Can't allocate memory for device information\n"); > return 0; > } > - memcpy(detected_devc, devc, sizeof(sb_devc)); > MDB(printk(KERN_INFO "SB %d.%02d detected OK (%x)\n", devc->major, devc->minor, hw_config->io_base)); > return 1; > } > -- > 1.7.9.5 >

1 0

Re: [alsa-devel] [PATCH] sound: oss: uart401: Used kmemdup instead of kmalloc and memcpy
by Takashi Iwai 02 Apr '13

02 Apr '13

At Mon, 25 Mar 2013 15:46:49 +0200, Alexandru Gheorghiu wrote: > > Used kmemdup instead of replicating it's behaviour with kmalloc followed > by memcpy. > Patch found using coccinelle. > > Signed-off-by: Alexandru Gheorghiu <gheorghiuandru(a)gmail.com> Thanks, applied. Takashi > --- > sound/oss/uart401.c | 11 +++++++---- > 1 file changed, 7 insertions(+), 4 deletions(-) > > diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c > index 8e514a6..5433c6f 100644 > --- a/sound/oss/uart401.c > +++ b/sound/oss/uart401.c > @@ -352,23 +352,26 @@ int probe_uart401(struct address_info *hw_config, struct module *owner) > goto cleanup_irq; > } > conf_printf(name, hw_config); > - midi_devs[devc->my_dev] = kmalloc(sizeof(struct midi_operations), GFP_KERNEL); > + midi_devs[devc->my_dev] = kmemdup(&uart401_operations, > + sizeof(struct midi_operations), > + GFP_KERNEL); > if (!midi_devs[devc->my_dev]) { > printk(KERN_ERR "uart401: Failed to allocate memory\n"); > goto cleanup_unload_mididev; > } > - memcpy(midi_devs[devc->my_dev], &uart401_operations, sizeof(struct midi_operations)); > > if (owner) > midi_devs[devc->my_dev]->owner = owner; > > midi_devs[devc->my_dev]->devc = devc; > - midi_devs[devc->my_dev]->converter = kmalloc(sizeof(struct synth_operations), GFP_KERNEL); > + midi_devs[devc->my_dev]->converter = kmemdup(&std_midi_synth, > + sizeof(struct synth_operations), > + GFP_KERNEL); > + > if (!midi_devs[devc->my_dev]->converter) { > printk(KERN_WARNING "uart401: Failed to allocate memory\n"); > goto cleanup_midi_devs; > } > - memcpy(midi_devs[devc->my_dev]->converter, &std_midi_synth, sizeof(struct synth_operations)); > strcpy(midi_devs[devc->my_dev]->info.name, name); > midi_devs[devc->my_dev]->converter->id = "UART401"; > midi_devs[devc->my_dev]->converter->midi_dev = devc->my_dev; > -- > 1.7.9.5 >

1 0

[alsa-devel] [PATCH] USB-audio: Add "HP Digital Stereo Headset" to SPDIF blacklist
by David Henningsson 02 Apr '13

02 Apr '13

Yet another headset without digital I/O. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 177a7af..71c2971 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -39,7 +39,8 @@ USB-Audio.pcm.iec958_device { # The below don't have digital in/out, so prevent them from being opened. "Blue Snowball" 999 - "Logitech Speaker Lapdesk N700" 999 + "HP Digital Stereo Headset" 999 + "Logitech Speaker Lapdesk N700" 999 "Logitech USB Headset" 999 "Logitech Wireless Headset" 999 "Plantronics GameCom 780" 999 -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: codecs: remove hidden prompt
by Mark Brown 02 Apr '13

02 Apr '13

On Tue, Mar 12, 2013 at 09:08:38PM +0100, Paul Bolle wrote: > The Kconfig symbol SND_SOC_OF_SIMPLE got removed in commit > f0fba2ad1b6b53d5360125c41953b7afcd6deff0 ("ASoC: multi-component - ASoC > Multi-Component Support"). But that commit missed one instance. Remove > it now, together with the prompt it has effectively hidden ever since. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: dapm: Remove redundant clear_walk() for supply widgets
by Mark Brown 01 Apr '13

01 Apr '13

From: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> We already clear the walked state in dapm_widget_power_check(), no need to do it again. Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfosonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1896a70..0a395eb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1179,8 +1179,6 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) return 1; } - dapm_clear_walk(w->dapm); - return 0; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: dapm: Only clear paths we've walked
by Mark Brown 01 Apr '13

01 Apr '13

From: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> When clearing the walked flags there is no need to clear all paths, we only need to clear the paths we actually walked. This means we can split dapm_clear_walk() into input and output versions and rather than going through all DAPM paths we can recurse down the path until we encounter paths we have not yet walked. This reduces the number of operations we need to perform and improves cache locality. [Pulled out of the vendor tree that the patch was originally generated for by me, any bugs were introduced in that process -- broonie] Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui(a)wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 47 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 35 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a395eb..93def6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -707,14 +707,33 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) } /* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) +static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, + struct list_head *sink) { struct snd_soc_dapm_path *p; - list_for_each_entry(p, &dapm->card->paths, list) - p->walked = 0; + list_for_each_entry(p, sink, list_source) { + if (p->walked) { + p->walked = 0; + dapm_clear_walk_output(dapm, &p->sink->sinks); + } + } +} + +static void dapm_clear_walk_input(struct snd_soc_dapm_context *dapm, + struct list_head *source) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, source, list_sink) { + if (p->walked) { + p->walked = 0; + dapm_clear_walk_input(dapm, &p->source->sources); + } + } } + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -997,13 +1016,17 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); dapm_reset(card); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { paths = is_connected_output_ep(dai->playback_widget, list); - else + dapm_clear_walk_output(&card->dapm, + &dai->playback_widget->sinks); + } else { paths = is_connected_input_ep(dai->capture_widget, list); + dapm_clear_walk_input(&card->dapm, + &dai->capture_widget->sources); + } trace_snd_soc_dapm_connected(paths, stream); - dapm_clear_walk(&card->dapm); mutex_unlock(&card->dapm_mutex); return paths; @@ -1106,9 +1129,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0 && in != 0; } @@ -1131,7 +1154,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) if (w->active) { in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); return in != 0; } else { return dapm_generic_check_power(w); @@ -1147,7 +1170,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) if (w->active) { out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0; } else { return dapm_generic_check_power(w); @@ -1775,9 +1798,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, return -ENOMEM; in = is_connected_input_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_input(w->dapm, &w->sources); out = is_connected_output_ep(w, NULL); - dapm_clear_walk(w->dapm); + dapm_clear_walk_output(w->dapm, &w->sinks); ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", -- 1.7.10.4

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[alsa-devel] [PATCH 00/13] mach-mxs header files cleanup
by Shawn Guo 01 Apr '13

01 Apr '13

The series cleans up mach-mxs headers files in order to support multiplatform build on mxs. Andrew, Jonathan, Mark, Can you please give your ACK on the patch you are copied, so that the series can go via arm-soc as a whole? Thanks. Shawn Shawn Guo (13): rtc: stmp3xxx: use stmp_reset_block() instead iio: mxs-lradc: remove unneeded mach header inclusion ASoC: mxs-saif: remove mach header inclusion ARM: mxs: remove empty hardware.h ARM: mxs: get reset address from device tree ARM: mxs: remove system.c ARM: mxs: get ocotp base address from device tree ARM: mxs: use debug_ll_io_init for low-level debug ARM: mxs: remove mm.c ARM: mxs: move mxs_get_ocotp() into mach-mxs.c ARM: mxs: remove common.h ARM: mxs: merge imx23 and imx28 into one machine_desc ARM: mxs: remove unused headers arch/arm/boot/dts/imx23.dtsi | 3 +- arch/arm/boot/dts/imx28.dtsi | 3 +- arch/arm/mach-mxs/Makefile | 4 - arch/arm/mach-mxs/include/mach/common.h | 25 --- arch/arm/mach-mxs/include/mach/debug-macro.S | 9 +- arch/arm/mach-mxs/include/mach/digctl.h | 22 --- arch/arm/mach-mxs/include/mach/hardware.h | 23 --- arch/arm/mach-mxs/include/mach/mx23.h | 169 ------------------- arch/arm/mach-mxs/include/mach/mx28.h | 225 -------------------------- arch/arm/mach-mxs/include/mach/mxs.h | 116 ------------- arch/arm/mach-mxs/mach-mxs.c | 182 +++++++++++++++++---- arch/arm/mach-mxs/mm.c | 52 ------ arch/arm/mach-mxs/ocotp.c | 93 ----------- arch/arm/mach-mxs/system.c | 139 ---------------- drivers/rtc/rtc-stmp3xxx.c | 6 +- drivers/staging/iio/adc/mxs-lradc.c | 3 - include/linux/clk/mxs.h | 16 ++ sound/soc/mxs/mxs-saif.c | 5 +- 18 files changed, 180 insertions(+), 915 deletions(-) delete mode 100644 arch/arm/mach-mxs/include/mach/common.h delete mode 100644 arch/arm/mach-mxs/include/mach/digctl.h delete mode 100644 arch/arm/mach-mxs/include/mach/hardware.h delete mode 100644 arch/arm/mach-mxs/include/mach/mx23.h delete mode 100644 arch/arm/mach-mxs/include/mach/mx28.h delete mode 100644 arch/arm/mach-mxs/include/mach/mxs.h delete mode 100644 arch/arm/mach-mxs/mm.c delete mode 100644 arch/arm/mach-mxs/ocotp.c delete mode 100644 arch/arm/mach-mxs/system.c create mode 100644 include/linux/clk/mxs.h -- 1.7.9.5

2 2

[alsa-devel] [PATCH V2] ASoC: Implement mixer control sharing
by Stephen Warren 31 Mar '13

31 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This is the equivalent of commit af46800 "ASoC: Implement mux control sharing", but applied to mixers instead of muxes. This allows a single control to affect multiple mixer widgets at once, which is useful when there is a single set of register bits that affects multiple mixers in HW, for example both the L and R mixers of a stereo path. Without this, you either: 1) End up with multiple controls that affect the same register bits, but whose DAPM state falls out of sync with HW, since the DAPM state is only updated for the specific control that is modified, and not for other paths that are affected by the register bit(s). 2) False paths through DAPM, since you end up merging unconnected stereo paths together into a single widget which hosts the single control, and then branching back out again, thus conjoining the enable states of the two input paths. Now that the kcontrol creation logic is split out into a separate function, dapm_create_or_share_mixmux_kcontrol(), also use that to replace most of the body of dapm_new_mux(). This should produce no functional change, but simply eliminates some mostly duplicated code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Split dapm_new_mixer()'s loop body into a separate function, dapm_create_or_share_mixmux_kcontrol, to avoid a nasty goto. Also use this for dapm_new_mux(). --- sound/soc/soc-dapm.c | 270 +++++++++++++++++++++++++------------------------- 1 file changed, 136 insertions(+), 134 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 33acd8b..c7cacf9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -504,17 +504,27 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, return 0; } -/* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_widget *w) +/* + * Determine if a kcontrol is shared. If it is, look it up. If it isn't, + * create it. Either way, add the widget into the control's widget list + */ +static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, + int kci, struct snd_soc_dapm_path *path) { struct snd_soc_dapm_context *dapm = w->dapm; - int i, ret = 0; - size_t name_len, prefix_len; - struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; + size_t prefix_len; + int shared; + struct snd_kcontrol *kcontrol; struct snd_soc_dapm_widget_list *wlist; + int wlistentries; size_t wlistsize; + bool wname_in_long_name, kcname_in_long_name; + size_t name_len; + char *long_name; + const char *name; + int ret; if (dapm->codec) prefix = dapm->codec->name_prefix; @@ -526,103 +536,141 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) else prefix_len = 0; - /* add kcontrol */ - for (i = 0; i < w->num_kcontrols; i++) { + shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[kci], + &kcontrol); - /* match name */ - list_for_each_entry(path, &w->sources, list_sink) { + if (kcontrol) { + wlist = kcontrol->private_data; + wlistentries = wlist->num_widgets + 1; + } else { + wlist = NULL; + wlistentries = 1; + } - /* mixer/mux paths name must match control name */ - if (path->name != (char *)w->kcontrol_news[i].name) - continue; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + wlist = krealloc(wlist, wlistsize, GFP_KERNEL); + if (wlist == NULL) { + dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist->num_widgets = wlistentries; + wlist->widgets[wlistentries - 1] = w; - if (w->kcontrols[i]) { - path->kcontrol = w->kcontrols[i]; - continue; + if (!kcontrol) { + if (shared) { + wname_in_long_name = false; + kcname_in_long_name = true; + } else { + switch (w->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + wname_in_long_name = true; + kcname_in_long_name = true; + break; + case snd_soc_dapm_mixer_named_ctl: + wname_in_long_name = false; + kcname_in_long_name = true; + break; + case snd_soc_dapm_mux: + case snd_soc_dapm_virt_mux: + case snd_soc_dapm_value_mux: + wname_in_long_name = true; + kcname_in_long_name = false; + break; + default: + kfree(wlist); + return -EINVAL; } + } + + if (wname_in_long_name && kcname_in_long_name) { + name_len = strlen(w->name) - prefix_len + 1 + + strlen(w->kcontrol_news[kci].name) + 1; - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - sizeof(struct snd_soc_dapm_widget *), - wlist = kzalloc(wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, - "ASoC: can't allocate widget list for %s\n", - w->name); + long_name = kmalloc(name_len, GFP_KERNEL); + if (long_name == NULL) { + kfree(wlist); return -ENOMEM; } - wlist->num_widgets = 1; - wlist->widgets[0] = w; - - /* add dapm control with long name. - * for dapm_mixer this is the concatenation of the - * mixer and kcontrol name. - * for dapm_mixer_named_ctl this is simply the - * kcontrol name. + + /* + * The control will get a prefix from the control + * creation process but we're also using the same + * prefix for widgets so cut the prefix off the + * front of the widget name. */ - name_len = strlen(w->kcontrol_news[i].name) + 1; - if (w->id != snd_soc_dapm_mixer_named_ctl) - name_len += 1 + strlen(w->name); + snprintf(long_name, name_len, "%s %s", + w->name + prefix_len, + w->kcontrol_news[kci].name); + long_name[name_len - 1] = '\0'; + + name = long_name; + } else if (wname_in_long_name) { + long_name = NULL; + name = w->name + prefix_len; + } else { + long_name = NULL; + name = w->kcontrol_news[kci].name; + } - path->long_name = kmalloc(name_len, GFP_KERNEL); + kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], wlist, name, + prefix); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, name, ret); + kfree(wlist); + kfree(long_name); + return ret; + } - if (path->long_name == NULL) { - kfree(wlist); - return -ENOMEM; - } + path->long_name = long_name; + } - switch (w->id) { - default: - /* The control will get a prefix from - * the control creation process but - * we're also using the same prefix - * for widgets so cut the prefix off - * the front of the widget name. - */ - snprintf((char *)path->long_name, name_len, - "%s %s", w->name + prefix_len, - w->kcontrol_news[i].name); - break; - case snd_soc_dapm_mixer_named_ctl: - snprintf((char *)path->long_name, name_len, - "%s", w->kcontrol_news[i].name); - break; - } + kcontrol->private_data = wlist; + w->kcontrols[kci] = kcontrol; + path->kcontrol = kcontrol; - ((char *)path->long_name)[name_len - 1] = '\0'; + return 0; +} - path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], - wlist, path->long_name, - prefix); - ret = snd_ctl_add(card, path->kcontrol); - if (ret < 0) { - dev_err(dapm->dev, "ASoC: failed to add widget" - " %s dapm kcontrol %s: %d\n", - w->name, path->long_name, ret); - kfree(wlist); - kfree(path->long_name); - path->long_name = NULL; - return ret; +/* create new dapm mixer control */ +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_soc_dapm_path *path; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + /* match name */ + list_for_each_entry(path, &w->sources, list_sink) { + /* mixer/mux paths name must match control name */ + if (path->name != (char *)w->kcontrol_news[i].name) + continue; + + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; } - w->kcontrols[i] = path->kcontrol; + + ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + if (ret < 0) + return ret; } } - return ret; + + return 0; } /* create new dapm mux control */ static int dapm_new_mux(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_dapm_path *path = NULL; - struct snd_kcontrol *kcontrol; - struct snd_card *card = dapm->card->snd_card; - const char *prefix; - size_t prefix_len; + struct snd_soc_dapm_path *path; int ret; - struct snd_soc_dapm_widget_list *wlist; - int shared, wlistentries; - size_t wlistsize; - const char *name; if (w->num_kcontrols != 1) { dev_err(dapm->dev, @@ -631,65 +679,19 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0], - &kcontrol); - if (kcontrol) { - wlist = kcontrol->private_data; - wlistentries = wlist->num_widgets + 1; - } else { - wlist = NULL; - wlistentries = 1; - } - wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - wlistentries * sizeof(struct snd_soc_dapm_widget *), - wlist = krealloc(wlist, wlistsize, GFP_KERNEL); - if (wlist == NULL) { - dev_err(dapm->dev, - "ASoC: can't allocate widget list for %s\n", w->name); - return -ENOMEM; - } - wlist->num_widgets = wlistentries; - wlist->widgets[wlistentries - 1] = w; - - if (!kcontrol) { - if (dapm->codec) - prefix = dapm->codec->name_prefix; - else - prefix = NULL; - - if (shared) { - name = w->kcontrol_news[0].name; - prefix_len = 0; - } else { - name = w->name; - if (prefix) - prefix_len = strlen(prefix) + 1; - else - prefix_len = 0; - } - - /* - * The control will get a prefix from the control creation - * process but we're also using the same prefix for widgets so - * cut the prefix off the front of the widget name. - */ - kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist, - name + prefix_len, prefix); - ret = snd_ctl_add(card, kcontrol); - if (ret < 0) { - dev_err(dapm->dev, "ASoC: failed to add kcontrol %s: %d\n", - w->name, ret); - kfree(wlist); - return ret; - } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + if (!path) { + dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); + return -EINVAL; } - kcontrol->private_data = wlist; - - w->kcontrols[0] = kcontrol; + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + if (ret < 0) + return ret; list_for_each_entry(path, &w->sources, list_sink) - path->kcontrol = kcontrol; + path->kcontrol = w->kcontrols[0]; return 0; } -- 1.7.10.4

2 1

[alsa-devel] Guide to USB debugging in Windows?
by Martin Cracauer 30 Mar '13

30 Mar '13

Does anyone have instructions on how to use a USB traffic monitor on windows to extract some bits needed to control a sound device under ALSA? As in, for people not normally developing in windows? I have here a Sound Devices USBPre v1. It is mostly USB 1.1 sound interface compatible and it accepts a stream played from ALSA under Linux just fine. The problem is that the manufacturer has decided to, in addition to the standard usb sound interface, have all the different inputs on the device behind on/off switches, all of them controlled by secret bits in the windows and mac drivers, all of them default to off. So thingie is a brick as-is. I don't blame them too hard, they now make a v2 of this devices and fixed it. Just so that I don't appear more insane than usual, let me say why I want to bother. The windows drivers for this are only available in 32 bit, and the mac drivers are 32 bit only as well, in addition the mac drivers never went > 16bit/48KHz. The inability to use this on modern macs and windows machines has pushed the price of these things very low on Ebay, considering it is a fully professional device with the best mic preamps you can get. If would be really cool if this could work under ALSA, and as you see I figured it needs just a couple of bits that you need to flip with some magic to at least get it to run up to 16bit/48KHz. Martin -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer <cracauer(a)cons.org> http://www.cons.org/cracauer/

3 2

[alsa-devel] [PATCH v2] ASoC: mid-x86 - add support for meaadata apis
by Vinod Koul 29 Mar '13

29 Mar '13

while at it, update the copyright timeline too Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v2: fix the missing header inclusion in patch sound/soc/mid-x86/sst_platform.c | 12 +++++++++++- sound/soc/mid-x86/sst_platform.h | 2 ++ 2 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a263cbe..968656c 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -1,7 +1,7 @@ /* * sst_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2012 Intel Corp + * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul(a)intel.com> * Author: Harsha Priya <priya.harsha(a)intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -652,11 +652,21 @@ static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream, return stream->compr_ops->get_codec_caps(codec); } +static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct sst_runtime_stream *stream = + cstream->runtime->private_data; + + return stream->compr_ops->set_metadata(stream->id, metadata); +} + static struct snd_compr_ops sst_platform_compr_ops = { .open = sst_platform_compr_open, .free = sst_platform_compr_free, .set_params = sst_platform_compr_set_params, + .set_metadata = sst_platform_compr_set_metadata, .trigger = sst_platform_compr_trigger, .pointer = sst_platform_compr_pointer, .ack = sst_platform_compr_ack, diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h index d61c5d5..cacc906 100644 --- a/sound/soc/mid-x86/sst_platform.h +++ b/sound/soc/mid-x86/sst_platform.h @@ -124,6 +124,8 @@ struct compress_sst_ops { int (*close) (unsigned int str_id); int (*get_caps) (struct snd_compr_caps *caps); int (*get_codec_caps) (struct snd_compr_codec_caps *codec); + int (*set_metadata) (unsigned int str_id, + struct snd_compr_metadata *mdata); }; -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls
by Mark Brown 29 Mar '13

29 Mar '13

Now that we have regular register mapped controls we should be splitting the control sets for ADSP1 and ADSP2 as the register maps are not identical. Do that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm_adsp.c | 26 +++++++++++++++++++------- sound/soc/codecs/wm_adsp.h | 3 ++- 4 files changed, 23 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index ddc98f0..57ba315 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1565,7 +1565,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d1b43eb..cb03cc4 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1572,7 +1572,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3a481fd..bc03bae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -247,7 +247,18 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -static const struct soc_enum wm_adsp_rate_enum[] = { +const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { + SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], + wm_adsp_fw_get, wm_adsp_fw_put), +}; +EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); + +#if IS_ENABLED(CONFIG_SND_SOC_ARIZONA) +static const struct soc_enum wm_adsp2_rate_enum[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, @@ -266,21 +277,22 @@ static const struct soc_enum wm_adsp_rate_enum[] = { arizona_rate_text, arizona_rate_val), }; -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp2_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP1 Rate", wm_adsp_rate_enum[0]), + SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP2 Rate", wm_adsp_rate_enum[1]), + SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP3 Rate", wm_adsp_rate_enum[2]), + SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP4 Rate", wm_adsp_rate_enum[3]), + SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), }; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp2_fw_controls); +#endif static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a..9f90c9f 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -65,7 +65,8 @@ struct wm_adsp { .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *adsp); int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: Implement mixer control sharing
by Stephen Warren 29 Mar '13

29 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> This is the equivalent of commit af46800 "ASoC: Implement mux control sharing", but applied to mixers instead of muxes. This allows a single control to affect multiple mixer widgets at once, which is useful when there is a single set of register bits that affects both multiple mixers in HW, for example both the L and R mixers of a stereo path. Without this, you either: 1) End up with multiple controls that affect the same register bits, but whose DAPM state falls out of sync with HW, since the DAPM state is only updated for the specific control that is modified, and not for other paths that are affected only by the other control. 2) False paths through DAPM, since you end up merging unconnected stereo paths together into a single widget which hosts the single control, and then branching back out again, thus conjoining the enable states of the two input paths. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, the goto here might be considered icky. It avoids heavy indent levels. Should I separate out the control creation part of dapm_new_mixer() into a separate function to avoid this? Also, this commit has the potential for user-space impact, since control names could change for existing drivers. However, in this case I suspect the affected controls wouldn't have worked very well anyway, since they'd be affected by one of the problems I mention in the commit description above anyway, so I doubt this will be a practical problem. FYI, this feature for muxes will be useful for the rt5640.c driver, which has a single register for the HPO MUX, which separately affects both a L and R path to the HP outputs. sound/soc/soc-dapm.c | 57 +++++++++++++++++++++++++++++++------------------- 1 file changed, 36 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 33acd8b..9a87aa5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -508,13 +508,15 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { struct snd_soc_dapm_context *dapm = w->dapm; - int i, ret = 0; + int i, ret; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; struct snd_soc_dapm_widget_list *wlist; + int shared, wlistentries; size_t wlistsize; + struct snd_kcontrol *kcontrol; if (dapm->codec) prefix = dapm->codec->name_prefix; @@ -531,7 +533,6 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) /* match name */ list_for_each_entry(path, &w->sources, list_sink) { - /* mixer/mux paths name must match control name */ if (path->name != (char *)w->kcontrol_news[i].name) continue; @@ -541,26 +542,41 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; } + shared = dapm_is_shared_kcontrol(dapm, w, + &w->kcontrol_news[i], &kcontrol); + + if (kcontrol) { + wlist = kcontrol->private_data; + wlistentries = wlist->num_widgets + 1; + } else { + wlist = NULL; + wlistentries = 1; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - sizeof(struct snd_soc_dapm_widget *), - wlist = kzalloc(wlistsize, GFP_KERNEL); + wlistentries * + sizeof(struct snd_soc_dapm_widget *), + wlist = krealloc(wlist, wlistsize, GFP_KERNEL); if (wlist == NULL) { dev_err(dapm->dev, "ASoC: can't allocate widget list for %s\n", w->name); return -ENOMEM; } - wlist->num_widgets = 1; - wlist->widgets[0] = w; + wlist->num_widgets = wlistentries; + wlist->widgets[wlistentries - 1] = w; + + if (kcontrol) + goto skip_new_control; /* add dapm control with long name. * for dapm_mixer this is the concatenation of the * mixer and kcontrol name. - * for dapm_mixer_named_ctl this is simply the - * kcontrol name. + * for dapm_mixer_named_ctl or shared controls, this + * is simply the kcontrol name. */ name_len = strlen(w->kcontrol_news[i].name) + 1; - if (w->id != snd_soc_dapm_mixer_named_ctl) + if (!shared && w->id != snd_soc_dapm_mixer_named_ctl) name_len += 1 + strlen(w->name); path->long_name = kmalloc(name_len, GFP_KERNEL); @@ -570,8 +586,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) return -ENOMEM; } - switch (w->id) { - default: + if (!shared && w->id != snd_soc_dapm_mixer_named_ctl) /* The control will get a prefix from * the control creation process but * we're also using the same prefix @@ -581,19 +596,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) snprintf((char *)path->long_name, name_len, "%s %s", w->name + prefix_len, w->kcontrol_news[i].name); - break; - case snd_soc_dapm_mixer_named_ctl: + else snprintf((char *)path->long_name, name_len, "%s", w->kcontrol_news[i].name); - break; - } ((char *)path->long_name)[name_len - 1] = '\0'; - path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], - wlist, path->long_name, - prefix); - ret = snd_ctl_add(card, path->kcontrol); + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], wlist, + path->long_name, prefix); + ret = snd_ctl_add(card, kcontrol); if (ret < 0) { dev_err(dapm->dev, "ASoC: failed to add widget" " %s dapm kcontrol %s: %d\n", @@ -603,10 +614,14 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) path->long_name = NULL; return ret; } - w->kcontrols[i] = path->kcontrol; + +skip_new_control: + path->kcontrol = kcontrol; + kcontrol->private_data = wlist; + w->kcontrols[i] = kcontrol; } } - return ret; + return 0; } /* create new dapm mux control */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm0010: Constify usage of firmware filenames
by Charles Keepax 29 Mar '13

29 Mar '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 40256b5..a5cd51f 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -342,7 +342,7 @@ static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); } -static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec) +static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) { struct spi_device *spi = to_spi_device(codec->dev); struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); -- 1.7.2.5

2 2

Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Jussi Laako 28 Mar '13

28 Mar '13

Hi Andreas,> Now of course, with your heroic effort in creating a generic driver that > would help everyone, we have a mess. Some DACs expect little endian > (mine and others as well) and others (all that are based on Rigisystems > modules) big endian positions. OK, we actually have two separate endianess issues here, but only one with word lengths longer than 8-bit. We have the traditional byte order endianess and the bit order endianess. Of course from computer point of view order of bits in a byte is sort of separate issue. And now the two file formats used for DSD use opposite bit ordering, but being byte-based don't have the byte ordering issue. At worst we have four different sample format possibilities for word lengths longer than a byte... Either way, application will have to be able to reshuffle the bits for the other format. - Jussi

2 2

Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Daniel Mack 28 Mar '13

28 Mar '13

On 28.03.2013 05:58, Andreas Koch wrote: > >> >> Andreas, >> >> Am I right to assume that in your case it's the same as for DoP, >> IOW, oldest in MSB? > > Good point - thanks for bringing it up - another can of worms. > With Windows having a little endian philosophy we started early on > with the first DSD workstation by putting the oldest sample in the > LSB position. ASIO came along and adopted the same thing, but then > extended support for the opposite as well. Rigisystems (offering OEM > modules for USB audio) came along about a year ago and insisted that > there is only their way, i.e. putting the oldest bit in the MSB > position. I tried to explain to them that there are already hundreds > of products out in the market that do it differently, but that didn't > matter to them. The result is that now we have implementations with > both. Up until now that was not a huge issue, because there is no > generic Windows driver and every hardware manufacturer had to have a > custom driver. > Now of course, with your heroic effort in creating a generic driver > that would help everyone, we have a mess. Some DACs expect little > endian (mine and others as well) and others (all that are based on > Rigisystems modules) big endian positions. > > Since the time I had the discussions with Rigisystems I didn't have > any further communication with them. Maybe it is a good idea to touch > base with them again and find out what the latest is. I will try my > contact there and let you know. Eek, what an utter mess. Just let me know what you find out. We then need to augment the number of DSD formats, but then again, we can also move the sample rate calculation table inside the ALSA kernel core instead. Takashi et al - please don't apply any of these patches yet (expecpt for the the indentation fixup) :) I guess we need some more spins on this. Thanks, Daniel

1 0

[alsa-devel] [PATCH] ALSA: hda - bug fix on return value when getting HDMI ELD info
by mengdong.lin＠intel.com 28 Mar '13

28 Mar '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. This patch also let debug message show 'pin_eld->monitor_present' which is the real pin response to verb GET_PIN_SENSE. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7dd8463..d0d7ac1 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { int i; - int ret; + int ret = 0; int size; /* diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..de8ac5c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, -- 1.7.10.4

3 3

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 28 Mar '13

28 Mar '13

On 03/24/2013 03:50 AM, Andreas Koch wrote: > rate for PCM. If you plug that DAC into a Mac, the Mac will > automatically select 705.6kHz for PCM and then nothing will work because > the DAC doesn't support that in PCM. So then, the DAC would have to > define 2 complete interfaces: one for PCM and one for DSD. That OK, the difference is that I don't use UAC2 at all, but completely different protocol. So what I'm talking about is only how it looks from the driver towards the application... DoP could be implemented similar way inside the UAC driver using for example snd_pcm_hw_rule_add() between SNDRV_PCM_HW_PARAM_FORMAT and SNDRV_PCM_HW_PARAM_RATE. And maybe enabled as quirks? There are many devices having constraints between rate and channels. In this case we just have a constraint between rate and formats. Looking at the UAC driver there seems to be something a bit related: /* FIXME: there is no AC3 format defined yet */ // fp->formats = SNDRV_PCM_FMTBIT_AC3; fp->formats = SNDRV_PCM_FMTBIT_U8; /* temporary hack to receive byte streams */ - Jussi

2 2

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 28 Mar '13

28 Mar '13

Hi Andreas, On 24.03.2013 02:50, Andreas Koch wrote: > At 01:01 PM 3/23/2013, Jussi Laako wrote: >> On 03/22/2013 06:05 PM, Andreas Koch wrote: >>> The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for >>> 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz on >>> the DAC, >> >> I see, this is different from how I've been dealing with it. Since I >> always use uint8 format, but instead vary the rate to 352.8/705.6/etc... >> >> Just because it's much more straightforward form the application's >> perspective -> output rate is always bit rate divided by eight >> (=packet rate). And switch between uint8 / uint32 sample format >> equals to switch between DSD <-> PCM. > > Since the sample rate is defined on the interface level, not on the > alternate interface level, it becomes a problem to vary the sample > rate. For instance, for 5.6MHz DSD a sample rate of 705.6kHz would be > necessary with 8-bit data. But most likely the DAC will not support > that rate for PCM. If you plug that DAC into a Mac, the Mac will > automatically select 705.6kHz for PCM and then nothing will work > because the DAC doesn't support that in PCM. So then, the DAC would > have to define 2 complete interfaces: one for PCM and one for DSD. > That increases the size of the descriptor file (which can be a > problem with some hardware implementations) and increases the > overhead a bit during enumeration and sample rate queries. For the > hardware it is easier to bundle all into one single interface and > then define alternate interfaces for PCM and DSD. > Just my 2 cents from the hardware perspective. Ok. So we could define 3 DSD formats, for 8-bit, 16-bit and 32-bit. On that three formats, arbitrary samples can be transported, and the mapping between the configured sample rate and the resulting DSD rates would be as follows (please tell me if I'm correct): 352.8kHz 705.6KHz 1411.2KHz 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz 32-bit 11.2MHz So driver would mark a stream as DSD capable (8, 16, 32bit, depending on the USB descriptors), and then leave it up to the application to do the sample frequency determination according to the above table, and also as stated in the DoP document. Other drivers for more sound cards (presumable on other bus types) might follow. I think it's ok to handle it like this, because after all, what an application sets as sample rate for the stream is the number of actual words that are transported, where as the DSD rate is referring to bits/s. Jussi, for your application, that means that you'd have to add these 3 new formats, and teach your application that they can be used for DSD right away. The question is whether all DSD-capable USB DACs behave correctly in order to make this logic work. Andreas? Thanks, Daniel

2 1

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 28 Mar '13

28 Mar '13

On 03/27/2013 05:08 PM, Andreas Koch wrote: > Clearly, main stream is 2.8MHz DSD. But there are new A/D and D/A chips > becoming available that support up to 11.2MHz. So far I haven't really And many DACs actually support also the 48k-base rates of 3.0/6.1/12.2 MHz. But of course from API point of view that's trivial. - Jussi

1 0

[alsa-devel] [PATCH 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 28 Mar '13

28 Mar '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: add DSD formats
by Daniel Mack 28 Mar '13

28 Mar '13

This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. In order to provide a compatibility way for pushing samples of DSD type through ordinary PCM channels, the "DoP open Standard" was created. See http://www.dsd-guide.com for a copy of the documentation. The two new types describe streams that are capable of handling DSD samples in 8-bit or in 16-bit (or at a single or double data rate, respectively). For applications, the following mapping table is used to configure the DSD mode by selecting a combination of PCM sample rates and word lengths: 352.8kHz 705.6KHz 1411.2KHz 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index aa7b0a8..2dffd55 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8) +#define SNDRV_PCM_FMTBIT_DSD_U16 _SNDRV_PCM_FMTBIT(DSD_U16) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..f9b0d3f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD in 1 byte */ +#define SNDRV_PCM_FORMAT_DSD_U16 ((__force snd_pcm_format_t) 49) /* DSD in 2 bytes */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_16U #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 578327e..433efdb 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_U8), + FORMAT(DSD_U16), }; const char *snd_pcm_format_name(snd_pcm_format_t format) -- 1.8.1.4

1 1

[alsa-devel] [PATCH 1/2 resend] ASoC: cs4271: switch to mute_stream
by Daniel Mack 28 Mar '13

28 Mar '13

Use the newly introduced mute_stream DAI operation, and don't mute the codec if it's called for the _CAPTURE stream. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Acked-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ac0d3b4..03036b3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -388,7 +388,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, return cs4271_set_deemph(codec); } -static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_codec *codec = dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -396,6 +396,9 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) int val_a = 0; int val_b = 0; + if (stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + if (mute) { val_a = CS4271_VOLA_MUTE; val_b = CS4271_VOLB_MUTE; @@ -442,7 +445,7 @@ static const struct snd_soc_dai_ops cs4271_dai_ops = { .hw_params = cs4271_hw_params, .set_sysclk = cs4271_set_dai_sysclk, .set_fmt = cs4271_set_dai_fmt, - .digital_mute = cs4271_digital_mute, + .mute_stream = cs4271_mute_stream, }; static struct snd_soc_dai_driver cs4271_dai = { -- 1.8.1.4

2 8

[alsa-devel] [PATCH v7] ASoC: soc-compress: Cancel delayed power down if needed
by Charles Keepax 28 Mar '13

28 Mar '13

When a new stream is being opened it is necessary to cancel any delayed power down of the audio. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- ARGH! White space error. Charles sound/soc/soc-compress.c | 15 ++++++++++++--- 1 files changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index f9b2197..0f4cfce 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -198,6 +198,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -211,19 +212,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH 1/3] ASoC: Constify the 'driver' field of snd_soc_platform
by Lars-Peter Clausen 28 Mar '13

28 Mar '13

The ASoC core does no not modify the driver of a platform. Making it const allows ASoC platform drivers to declare the snd_soc_platform_driver struct as const. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ++-- sound/soc/soc-core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1ffe9d4..e666119 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -373,7 +373,7 @@ int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); int snd_soc_register_platform(struct device *dev, - struct snd_soc_platform_driver *platform_drv); + const struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -829,7 +829,7 @@ struct snd_soc_platform { const char *name; int id; struct device *dev; - struct snd_soc_platform_driver *driver; + const struct snd_soc_platform_driver *driver; struct mutex mutex; unsigned int suspended:1; /* platform is suspended */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 561d589..7bf21a1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3905,7 +3905,7 @@ static void snd_soc_unregister_dais(struct device *dev, size_t count) * @platform: platform to register */ int snd_soc_register_platform(struct device *dev, - struct snd_soc_platform_driver *platform_drv) + const struct snd_soc_platform_driver *platform_drv) { struct snd_soc_platform *platform; -- 1.8.0

2 3

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Mark Brown 28 Mar '13

28 Mar '13

On Tue, Mar 26, 2013 at 05:35:38PM -0600, Stephen Warren wrote: > Mark, a couple questions: > * Is the custom index_reg device attr file OK, or should I remove that? > * Do new CODEC drivers have to use regmap directly, or is using the ASoC > IO system still OK? Convert to regmap please - it looks like this ought to be using the paging support from a quick glance at what the register I/O stuff is doing. The device file appears to duplicate the register dump stuff and isn't suitable for sysfs anyway as it's not one value per file, it ought to be in debugfs. > +/* IN1/IN2 Input Type */ > +static const char * const rt5640_input_mode[] = { > + "Single ended", "Differential"}; This looks like platform data. > +static const char * const rt5640_data_select[] = { > + "Normal", "left copy to right", "right copy to left", "Swap"}; DAPM? > +/* DMIC */ > +static const char * const rt5640_dmic_mode[] = {"Disable", "DMIC1", "DMIC2"}; > + > +static const SOC_ENUM_SINGLE_DECL(rt5640_dmic_enum, 0, 0, rt5640_dmic_mode); DAPM? > +static int rt5640_vol_rescale_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct soc_mixer_control *mc = > + (struct soc_mixer_control *)kcontrol->private_value; > + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); > + unsigned int val = snd_soc_read(codec, mc->reg); > + > + ucontrol->value.integer.value[0] = VOL_RESCALE_MAX_VOL - > + ((val & RT5640_L_VOL_MASK) >> mc->shift); > + ucontrol->value.integer.value[1] = VOL_RESCALE_MAX_VOL - > + (val & RT5640_R_VOL_MASK); This looks like a range control. > +static int hp_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + switch (event) { > + case SND_SOC_DAPM_POST_PMU: > + break; > + > + case SND_SOC_DAPM_PRE_PMD: > + break; > + > + default: > + return 0; > + } > + return 0; > +} This does nothing, it should be removed. > +static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + > + switch (event) { > + case SND_SOC_DAPM_PRE_PMU: > + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, > + RT5640_GP2_PIN_MASK | RT5640_GP3_PIN_MASK, > + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP3_PIN_DMIC1_SDA); > + snd_soc_update_bits(codec, RT5640_DMIC, > + RT5640_DMIC_1L_LH_MASK | RT5640_DMIC_1R_LH_MASK | > + RT5640_DMIC_1_DP_MASK, > + RT5640_DMIC_1L_LH_FALLING | RT5640_DMIC_1R_LH_RISING | > + RT5640_DMIC_1_DP_IN1P); > + snd_soc_update_bits(codec, RT5640_DMIC, > + RT5640_DMIC_1_EN_MASK, RT5640_DMIC_1_EN); > + default: > + return 0; The last write there looks awfully like an enable but there's not any corresponding disable... > +static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, > + struct snd_kcontrol *kcontrol, int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + > + switch (event) { > + case SND_SOC_DAPM_PRE_PMU: > + snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, > + RT5640_GP2_PIN_MASK | RT5640_GP4_PIN_MASK, > + RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP4_PIN_DMIC2_SDA); There's a degree of overlap between these two functions... > +static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { > + {"IN1P", NULL, "LDO2"}, > + {"IN2P", NULL, "LDO2"}, > + > + {"IN1P", NULL, "MIC1"}, > + {"IN1N", NULL, "MIC1"}, > + {"IN2P", NULL, "MIC2"}, > + {"IN2N", NULL, "MIC2"}, Are MIC1 and MIC2 pins on the device? It's a bit odd to have a single physical input feeding both sides of a differential path internally. > + {"DMIC L1", NULL, "DMIC CLK"}, > + {"DMIC L2", NULL, "DMIC CLK"}, Shouldn't the DMIC CLK widget be handling the clock enables/muxes from the DMIC event above? > +static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) > +{ > + int ret = 0, val = snd_soc_read(codec, RT5640_I2S1_SDP); > + > + if (codec == NULL) > + return -EINVAL; No point in the NULL check immediately after a dereference. > + val = (val & RT5640_I2S_IF_MASK) >> RT5640_I2S_IF_SFT; > + switch (dai_id) { > + case RT5640_AIF1: > + if (val == RT5640_IF_123 || val == RT5640_IF_132 || > + val == RT5640_IF_113) > + ret |= RT5640_U_IF1; > + if (val == RT5640_IF_312 || val == RT5640_IF_213 || > + val == RT5640_IF_113) > + ret |= RT5640_U_IF2; > + if (val == RT5640_IF_321 || val == RT5640_IF_231) > + ret |= RT5640_U_IF3; The tree of if ( || || ) looks like a switch statement and should probably be written as such. > + frame_size = snd_soc_params_to_frame_size(params); > + if (frame_size < 0) { > + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); > + return -EINVAL; > + } The error message here is confusing as a negative code is an errno not a frame size and it probably ought to be passed to the caller. > + bclk_ms = frame_size > 32 ? 1 : 0; Grumble. > +static int rt5640_prepare(struct snd_pcm_substream *substream, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_codec *codec = rtd->codec; > + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); > + > + rt5640->aif_pu = dai->id; > + return 0; > +} This looks like only one DAI can be active at once, shouldn't there be some sort of checking for busy here? > + dai_sel = get_sdp_info(codec, dai->id); > + if (dai_sel < 0) { > + dev_err(codec->dev, "Failed to get sdp info: %d\n", dai_sel); > + return -EINVAL; > + } I suspect using dai->base (or just picking a suitable value for dai->id even) would be simpler. > + if (dai_sel & RT5640_U_IF1) { > + snd_soc_update_bits(codec, RT5640_I2S1_SDP, > + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | > + RT5640_I2S_DF_MASK, reg_val); > + } > + if (dai_sel & RT5640_U_IF2) { > + snd_soc_update_bits(codec, RT5640_I2S2_SDP, > + RT5640_I2S_MS_MASK | RT5640_I2S_BP_MASK | > + RT5640_I2S_DF_MASK, reg_val); > + } This looks like switch statements again, though (not having checked) I'd expect that there's just blocks of registers for each DAI and we could just work in terms of a base? > +/** > + * rt5640_pll_calc - Calcualte PLL M/N/K code. Typo. > + snd_soc_add_codec_controls(codec, rt5640_snd_controls, > + ARRAY_SIZE(rt5640_snd_controls)); Set in the driver structure. > +static int rt5640_resume(struct snd_soc_codec *codec) > +{ > + int ret = 0 ; > + > + codec->cache_sync = 1; > + ret = snd_soc_cache_sync(codec); > + if (ret) { > + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); > + return ret; > + } > + rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); We also sync the cache in the bias management code. > +static int rt5640_i2c_probe(struct i2c_client *i2c, > + const struct i2c_device_id *id) > +{ > + struct rt5640_priv *rt5640; > + int ret; > + > + rt5640 = kzalloc(sizeof(struct rt5640_priv), GFP_KERNEL); > + if (NULL == rt5640) > + return -ENOMEM; devm_kzalloc() > +struct rt5640_pll_code { > + bool m_bp; /* Indicates bypass m code or not. */ > + int m_code; > + int n_code; > + int k_code; > +}; > + > +struct rt5640_priv { > + struct snd_soc_codec *codec; Move these into the body of the driver.

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 27 Mar '13

27 Mar '13

On 22.03.2013 17:05, Andreas Koch wrote: > Hi all, > > At 01:11 AM 3/22/2013, Daniel Mack wrote: >> Hi Jussi, >> Hi Andreas, >> >> On 21.03.2013 22:21, Jussi Laako wrote: >>> On 03/18/2013 03:49 PM, Daniel Mack wrote: >>>> Yes. If at all, we should add a SND_PCM_FORMAT_DSD, and a quirk for that >>>> device. But given that there is no application for DSD in userspace >>>> either, we probably don't need to care. >>> >>> Yes there is? At least my application. And I'd be happy to support any >>> custom format created for the purpose... :) >> >> Ah, interesting. So what *are* you using right now for your transport in >> userspace? Is your application abusing standard PCM interfaces that are >> 'known' to be in fact DSD? How do other DACs expose their interfaces in >> the Linux world, and which ones did you test your application with? >> >> If there's actual need for a new SND_PCM_FORMAT, it would be easy to >> patch it through all the layers down to userspace. >> >>> I would propose to create custom 8-bit sample formats for DSD purposes, >>> similar to the ASIO formats. Practically two main ones, byte with oldest >>> bit in MSB and byte with oldest bit in LSB. And maybe the "DSD-wide" one >>> which is 8-bit DSD sample. >> >> Andreas, which modes does the MPD-3 accept exactly on its two 'raw data' >> interfaces? The only hint I have is that one of it is 8-bit and the >> other is 16-bit, but given the lack of a proper test setup for this >> format, I can't currently test anything. > > The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for > 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz > on the DAC, but then sends either 8 bits or 16 bits depending on the > DSD rate. Since 11.2MHz DSD is already on the horizon we may have to > change this to a single 8-bit raw interface and then vary the > underlying USB transfer sample rate between 1x, 2x, 4x 352.8kHz - > that would be more logical and straight forward for the hardware > implementation. This is only supported through my own driver for the > Windows platform which uses the standard ASIO interface that supports > DSD at those rates. Since the Mac OS only supports PCM streams and > nothing else I brought together a bunch of DAC manufacturers and > playback software developers to standardize a method to "cannibalize" > the PCM stream for DSD data. The method is described in the attached > and is now supported by most manufacturers. > This also has been implemented on a few Linux platforms (Auraliti for > instance). Even though the DoP standard works really well so far, it > is a "bandaid" and should only be used if there is no other choice. > If you guys are discussing implementing a dedicated data stream in > your driver for DSD, then I think this would be better and safer in the future. Is there any official stable download location for this document which we could link to from our documentation? Thanks, Daniel

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 27 Mar '13

27 Mar '13

On 27.03.2013 16:08, Andreas Koch wrote: > At 02:48 AM 3/27/2013, Daniel Mack wrote: >> On 27.03.2013 06:50, Andreas Koch wrote: >>> Hi Daniel >>> >>>> Ok. So we could define 3 DSD formats, for 8-bit, 16-bit and 32-bit. On >>>> that three formats, arbitrary samples can be transported, and the >>>> mapping between the configured sample rate and the resulting DSD rates >>>> would be as follows (please tell me if I'm correct): >>>> >>>> 352.8kHz 705.6KHz 1411.2KHz >>>> >>>> 8-bit 2.8MHz 5.6MHz 11.2MHz >>>> 16-bit 5.6MHz 11.2MHz >>>> 32-bit 11.2MHz >>>> >>>> So driver would mark a stream as DSD capable (8, 16, 32bit, depending on >>>> the USB descriptors), and then leave it up to the application to do the >>>> sample frequency determination according to the above table, and also as >>>> stated in the DoP document. Other drivers for more sound cards >>>> (presumable on other bus types) might follow. >>>> >>>> I think it's ok to handle it like this, because after all, what an >>>> application sets as sample rate for the stream is the number of actual >>>> words that are transported, where as the DSD rate is referring to bits/s. >>>> >>>> Jussi, for your application, that means that you'd have to add these 3 >>>> new formats, and teach your application that they can be used for DSD >>>> right away. >>>> >>>> The question is whether all DSD-capable USB DACs behave correctly in >>>> order to make this logic work. Andreas? >>> >>> If I understand this correctly then your driver would send the DSD >>> data in whatever format the DAC requests via its descriptors. The DSD >>> rate would either control the word length (8,16,32) or the underlying >>> sample rate (1x, 2x, 4x 352.8kHz). That would be the most flexible >>> implementation and I don't see why any hardware implementation would >>> have a problem with that. >> >> The question is whether all DACs would assume the same DSD rate as in >> the table above (ie, 8-bit, 352.8kHz == 2.8MHz DSD). > > I don't see how they could assume any different rate, so yes, that > seems a good assumption. Ok, good. >> And do we actually need a 32-bit format at all or is 8 and 16 enough? > > Clearly, main stream is 2.8MHz DSD. But there are new A/D and D/A > chips becoming available that support up to 11.2MHz. So far I > haven't really seen any performance gain with that, but one never > knows what improvements will be made in the future. If that is a > problem on your side I wouldn't spend so much time on it, but if it > is a matter of just adding a few lines then why not. It a bit in a 64bit field, so its a limited resource. If there's no need for it, I'll leave it out for now. I'll send out some trivial patches for DSD stream marking soon. Thanks, Daniel

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Re: [alsa-devel] [PATCH 2/2] ARM: clk: imx35 sound admux_gate bugfix
by Mark Brown 27 Mar '13

27 Mar '13

On Wed, Mar 27, 2013 at 12:57:00PM +0100, Philipp Zabel wrote: > Am Mittwoch, den 27.03.2013, 12:22 +0100 schrieb Markus Pargmann: > > imx-ssi needs admux_gate clock to work on imx35. This patch registers > > this clock and enables it in the imx-ssi driver. > I have no idea what the admux_gate really is, but the ssi driver doesn't > seem like the right place. Isn't this more closely related to the > imx-audmux.c driver? This is a patch to the ASoC driver? I just ignored it since it said it was an ARM patch... Anyway, what I'd expect should be happening here is that the clock API walks up the tree enabling supply clocks - that would mean that either that code isn't there or there's a missing parent child relationship in the data.

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[alsa-devel] [PATCH v6] ASoC: soc-compress: Cancel delayed power down if needed
by Charles Keepax 27 Mar '13

27 Mar '13

When a new stream is being opened it is necessary to cancel any delayed power down of the audio. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Well hopefully sixth time lucky, generated against the asoc tree for-next branch. Charles sound/soc/soc-compress.c | 16 +++++++++++++--- 1 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index f9b2197..4b4b5fa 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -198,8 +198,10 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* first we call set_params for the platform driver @@ -211,19 +213,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm5102: Implement OSR support
by Mark Brown 27 Mar '13

27 Mar '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b7a3fdc..99972a5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,26 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const char *wm5102_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm5102_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm5102_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0xe00, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 0xe00, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0xe00, 3, + wm5102_osr_text, wm5102_osr_val), +}; + #define WM5102_NG_SRC(name, base) \ SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ @@ -761,6 +781,8 @@ ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, ARIZONA_OUT5_OSR_SHIFT, 1, 0), @@ -790,6 +812,10 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_VAL_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), +SOC_VAL_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), +SOC_VAL_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.7.10.4

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Re: [alsa-devel] [PATCH 2/2] ARM: clk: imx35 sound admux_gate bugfix
by Markus Pargmann 27 Mar '13

27 Mar '13

Hi Philipp, On Wed, Mar 27, 2013 at 12:57:00PM +0100, Philipp Zabel wrote: > Hi Markus, > > Am Mittwoch, den 27.03.2013, 12:22 +0100 schrieb Markus Pargmann: > > imx-ssi needs admux_gate clock to work on imx35. This patch registers > > this clock and enables it in the imx-ssi driver. > > I have no idea what the admux_gate really is, but the ssi driver doesn't > seem like the right place. Isn't this more closely related to the > imx-audmux.c driver? Actually admux clock is not audmux clock, that one is seperated. So I am not sure where this belongs. As imx-ssi was not working without it, I added it here. But you are right, I also spoke with Sascha and I will simply move the clock enable to imx35-dt.c. > > > A reboot without this patch results in a failing AC97 for example on > > pcm043. > > > > Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> > > Cc: stable(a)vger.kernel.org > > --- > > arch/arm/mach-imx/clk-imx25.c | 6 ++++-- > > arch/arm/mach-imx/clk-imx27.c | 6 ++++-- > > arch/arm/mach-imx/clk-imx31.c | 6 ++++-- > > arch/arm/mach-imx/clk-imx35.c | 6 ++++-- > > arch/arm/mach-imx/clk-imx51-imx53.c | 9 ++++++--- > > sound/soc/fsl/imx-ssi.c | 16 +++++++++++++++- > > sound/soc/fsl/imx-ssi.h | 2 ++ > > 7 files changed, 39 insertions(+), 12 deletions(-) > > > > diff --git a/arch/arm/mach-imx/clk-imx25.c b/arch/arm/mach-imx/clk-imx25.c > > index 69858c7..174ff8b 100644 > > --- a/arch/arm/mach-imx/clk-imx25.c > > +++ b/arch/arm/mach-imx/clk-imx25.c > > @@ -285,8 +285,10 @@ int __init mx25_clocks_init(void) > > clk_register_clkdev(clk[lcdc_ipg], "ipg", "imx21-fb.0"); > > clk_register_clkdev(clk[lcdc_ahb], "ahb", "imx21-fb.0"); > > clk_register_clkdev(clk[wdt_ipg], NULL, "imx2-wdt.0"); > > - clk_register_clkdev(clk[ssi1_ipg], NULL, "imx-ssi.0"); > > - clk_register_clkdev(clk[ssi2_ipg], NULL, "imx-ssi.1"); > > + clk_register_clkdev(clk[ssi1_ipg], "ipg", "imx-ssi.0"); > > + clk_register_clkdev(clk[ssi2_ipg], "ipg", "imx-ssi.1"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.0"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.1"); > > clk_register_clkdev(clk[esdhc1_ipg_per], "per", "sdhci-esdhc-imx25.0"); > > clk_register_clkdev(clk[esdhc1_ipg], "ipg", "sdhci-esdhc-imx25.0"); > > clk_register_clkdev(clk[esdhc1_ahb], "ahb", "sdhci-esdhc-imx25.0"); > > diff --git a/arch/arm/mach-imx/clk-imx27.c b/arch/arm/mach-imx/clk-imx27.c > > index 30b3242..439d5bd 100644 > > --- a/arch/arm/mach-imx/clk-imx27.c > > +++ b/arch/arm/mach-imx/clk-imx27.c > > @@ -252,8 +252,10 @@ int __init mx27_clocks_init(unsigned long fref) > > clk_register_clkdev(clk[usb_div], "per", "mxc-ehci.2"); > > clk_register_clkdev(clk[usb_ipg_gate], "ipg", "mxc-ehci.2"); > > clk_register_clkdev(clk[usb_ahb_gate], "ahb", "mxc-ehci.2"); > > - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); > > - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); > > + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); > > + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.0"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.1"); > > clk_register_clkdev(clk[nfc_baud_gate], NULL, "imx27-nand.0"); > > clk_register_clkdev(clk[vpu_baud_gate], "per", "coda-imx27.0"); > > clk_register_clkdev(clk[vpu_ahb_gate], "ahb", "coda-imx27.0"); > > diff --git a/arch/arm/mach-imx/clk-imx31.c b/arch/arm/mach-imx/clk-imx31.c > > index b5b65f3..478b33e 100644 > > --- a/arch/arm/mach-imx/clk-imx31.c > > +++ b/arch/arm/mach-imx/clk-imx31.c > > @@ -171,8 +171,10 @@ int __init mx31_clocks_init(unsigned long fref) > > clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1.0"); > > clk_register_clkdev(clk[sdhc1_gate], NULL, "imx31-mmc.0"); > > clk_register_clkdev(clk[sdhc2_gate], NULL, "imx31-mmc.1"); > > - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); > > - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); > > + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); > > + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.1"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.0"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.1"); > > clk_register_clkdev(clk[firi_gate], "firi", NULL); > > clk_register_clkdev(clk[ata_gate], NULL, "pata_imx"); > > clk_register_clkdev(clk[rtic_gate], "rtic", NULL); > > diff --git a/arch/arm/mach-imx/clk-imx35.c b/arch/arm/mach-imx/clk-imx35.c > > index b95898a..4ab18ac 100644 > > --- a/arch/arm/mach-imx/clk-imx35.c > > +++ b/arch/arm/mach-imx/clk-imx35.c > > @@ -233,8 +233,10 @@ int __init mx35_clocks_init(void) > > clk_register_clkdev(clk[kpp_gate], NULL, "imx-keypad"); > > clk_register_clkdev(clk[owire_gate], NULL, "mxc_w1"); > > clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); > > - clk_register_clkdev(clk[ssi1_gate], NULL, "imx-ssi.0"); > > - clk_register_clkdev(clk[ssi2_gate], NULL, "imx-ssi.1"); > > + clk_register_clkdev(clk[ssi1_gate], "ipg", "imx-ssi.0"); > > + clk_register_clkdev(clk[ssi2_gate], "ipg", "imx-ssi.1"); > > + clk_register_clkdev(clk[admux_gate], "admux", "imx-ssi.0"); > > + clk_register_clkdev(clk[admux_gate], "admux", "imx-ssi.1"); > > /* i.mx35 has the i.mx21 type uart */ > > clk_register_clkdev(clk[uart1_gate], "per", "imx21-uart.0"); > > clk_register_clkdev(clk[ipg], "ipg", "imx21-uart.0"); > > diff --git a/arch/arm/mach-imx/clk-imx51-imx53.c b/arch/arm/mach-imx/clk-imx51-imx53.c > > index 0f39f8c..b413a1e 100644 > > --- a/arch/arm/mach-imx/clk-imx51-imx53.c > > +++ b/arch/arm/mach-imx/clk-imx51-imx53.c > > @@ -275,9 +275,12 @@ static void __init mx5_clocks_common_init(unsigned long rate_ckil, > > clk_register_clkdev(clk[usboh3_gate], "ipg", "imx-udc-mx51"); > > clk_register_clkdev(clk[usboh3_gate], "ahb", "imx-udc-mx51"); > > clk_register_clkdev(clk[nfc_gate], NULL, "imx51-nand"); > > - clk_register_clkdev(clk[ssi1_ipg_gate], NULL, "imx-ssi.0"); > > - clk_register_clkdev(clk[ssi2_ipg_gate], NULL, "imx-ssi.1"); > > - clk_register_clkdev(clk[ssi3_ipg_gate], NULL, "imx-ssi.2"); > > + clk_register_clkdev(clk[ssi1_ipg_gate], "ipg", "imx-ssi.0"); > > + clk_register_clkdev(clk[ssi2_ipg_gate], "ipg", "imx-ssi.1"); > > + clk_register_clkdev(clk[ssi3_ipg_gate], "ipg", "imx-ssi.2"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.0"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.1"); > > + clk_register_clkdev(clk[dummy], "admux", "imx-ssi.2"); > > I think those should be removed completely. If really needed, it should > rather be added to the device tree for i.MX5: Oh yes, I grepped for imx-ssi.. However next patch version will not touch any of the other imxs code. > > --- a/arch/arm/boot/dts/imx53.dtsi > +++ b/arch/arm/boot/dts/imx53.dtsi > @@ -162,7 +162,8 @@ > compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; > reg = <0x50014000 0x4000>; > interrupts = <30>; > - clocks = <&clks 49>; > + clocks = <&clks 49>, <&clks 0>; > + clock-names = "ipg", "admux"; > fsl,fifo-depth = <15>; > fsl,ssi-dma-events = <25 24 23 22>; /* TX0 RX0 TX1 RX1 */ > status = "disabled"; > @@ -791,7 +792,8 @@ > compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; > reg = <0x63fcc000 0x4000>; > interrupts = <29>; > - clocks = <&clks 48>; > + clocks = <&clks 48>, <&clks 0>; > + clock-names = "ipg", "admux"; > fsl,fifo-depth = <15>; > fsl,ssi-dma-events = <29 28 27 26>; /* TX0 RX0 TX1 RX1 */ > status = "disabled"; > @@ -815,7 +817,8 @@ > compatible = "fsl,imx53-ssi", "fsl,imx21-ssi"; > reg = <0x63fe8000 0x4000>; > interrupts = <96>; > - clocks = <&clks 50>; > + clocks = <&clks 50>, <&clks 0>; > + clock-names = "ipg", "admux"; > fsl,fifo-depth = <15>; > fsl,ssi-dma-events = <47 46 45 44>; /* TX0 RX0 TX1 RX1 */ > status = "disabled"; > > But again, I'm not convinced the ssi driver is the right place to enable > the "admux" gate. > > > clk_register_clkdev(clk[ssi_ext1_gate], "ssi_ext1", NULL); > > clk_register_clkdev(clk[ssi_ext2_gate], "ssi_ext2", NULL); > > clk_register_clkdev(clk[sdma_gate], NULL, "imx35-sdma"); > > diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c > > index 55464a5..9f96785 100644 > > --- a/sound/soc/fsl/imx-ssi.c > > +++ b/sound/soc/fsl/imx-ssi.c > > @@ -535,7 +535,7 @@ static int imx_ssi_probe(struct platform_device *pdev) > > > > ssi->irq = platform_get_irq(pdev, 0); > > > > - ssi->clk = devm_clk_get(&pdev->dev, NULL); > > + ssi->clk = devm_clk_get(&pdev->dev, "ipg"); > > if (IS_ERR(ssi->clk)) { > > ret = PTR_ERR(ssi->clk); > > dev_err(&pdev->dev, "Cannot get the clock: %d\n", > > @@ -544,6 +544,15 @@ static int imx_ssi_probe(struct platform_device *pdev) > > } > > clk_prepare_enable(ssi->clk); > > > > + ssi->admux_clk = devm_clk_get(&pdev->dev, "admux"); > > + if (IS_ERR(ssi->admux_clk)) { > > + if (PTR_ERR(ssi->admux_clk) == -ENOMEM) > > + goto failed_admux_clk_mem; > > + ssi->admux_clk = NULL; > > + } else { > > + clk_prepare_enable(ssi->admux_clk); > > + } > > + > > res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > if (!res) { > > ret = -ENODEV; > > @@ -631,6 +640,9 @@ failed_pdev_fiq_alloc: > > failed_register: > > release_mem_region(res->start, resource_size(res)); > > failed_get_resource: > > + if (ssi->admux_clk) > > + clk_disable_unprepare(ssi->admux_clk); > > +failed_admux_clk_mem: > > clk_disable_unprepare(ssi->clk); > > failed_clk: > > > > @@ -652,6 +664,8 @@ static int imx_ssi_remove(struct platform_device *pdev) > > > > release_mem_region(res->start, resource_size(res)); > > clk_disable_unprepare(ssi->clk); > > + if (ssi->admux_clk) > > + clk_disable_unprepare(ssi->admux_clk); > > > > return 0; > > } > > diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h > > index dc114bd..3dfbd92 100644 > > --- a/sound/soc/fsl/imx-ssi.h > > +++ b/sound/soc/fsl/imx-ssi.h > > @@ -194,6 +194,8 @@ struct imx_ssi { > > > > struct snd_soc_dai *imx_ac97; > > struct clk *clk; > > + /* imx35 has a admux clock that has to be active for sound to work */ > > + struct clk *admux_clk; > > void __iomem *base; > > int irq; > > int fiq_enable; > > regards > Philipp > > Thanks, Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

[alsa-devel] need help for PCM constraint/rules
by Takashi Sakamoto 27 Mar '13

27 Mar '13

Dear developers, I'm a newbile for this project and working for kernel land driver for Fireworks with ALSA framework. I need your help for constraint/rule of PCM component. Related source is: https://github.com/takaswie/snd-firewire-improve/blob/master/fireworks/fire… My device (AudioFirePre8) changes its channels according to sampling rate. For example: 32.0 - 48.0kHz : 16ch: efw->channels_sets[0] 88.2 - 96.0kHz : 12ch: efw->channels_sets[1] 176.4 - 192.0kHz : 10ch: efw->channels_sets[2] (but not available) The other devices such as AudioFire12 has the same number of channels (12ch)in each sampling rate. I managed to write this rule refering to rme9652.c but I cannot achieve my purpose. It's my glad to get some help about this. Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp ======== Fireworks is Echo Audio's series of Firewire Audio interface. http://echoaudio.com/products/audiofire-pre8 It utilizes IEC 61883-1 and -6 in its implementation and familiar with current snd-firewire-lib.ko. FFADO also produce driver for it. It's a user-land implementation. http://www.ffado.org/ My aim is writing kernel-land driver for it with ALSA framework. Takashi Sakamoto o-takashi(a)sakamocchi.jp

2 4

[alsa-devel] [PATCH] ASoC: soc-compress: Cancel delayed power down if needed
by Charles Keepax 27 Mar '13

27 Mar '13

When a new stream is being opened it is necessary to cancel any delayed power down of the audio. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index f9fb112..eb8191b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -194,6 +194,7 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -216,6 +217,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto out; } + /* cancel any delayed stream shutdown that is pending */ + if (codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&rtd->delayed_work); + } + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -- 1.7.2.5

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[alsa-devel] [PATCH 0/2] ARM: imx35 clk bugfixes
by Markus Pargmann 27 Mar '13

27 Mar '13

Hi, Two bugfixes which enable needed clocks. In both cases the first boot worked because the clocks were enabled by default after powerup. After disabling the unused clocks in the boot process, some parts were not working anymore and device init for example after a reboot did not work. Regards, Markus

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Re: [alsa-devel] Support for DSD streams
by Daniel Mack 27 Mar '13

27 Mar '13

On 24.03.2013 02:33, Andreas Koch wrote: > Hi Daniel, > > At 11:43 AM 3/23/2013, Daniel Mack wrote: >> Hi Andreas, >> >> On 23.03.2013 18:35, Andreas Koch wrote: >>>> Yes, understood. The thing is just that we shouldn't break exising >>>> solutions that already rely on this "bandaid". >>> >>> You wouldn't break any existing solution based on DoP, because that >>> solution is dealt with in the application on one side and the DAC on >>> the other side. The driver and USB transfer both think it is PCM data >>> that is being sent. >> >> Well, that was exactly the problem with your device. With the driver as >> it was, it exposed a PCM interface which is 16-bit wide, and that was >> picked by ALSA when playing 16-bit material (because ALSA takes whatever >> is offered by the hardware natively). > > Not quite - the 16-bit wide interface offered by my DAC is not PCM, > but raw data. For the correct quirks handling, I'd need to know whether all products with your vendor ID are DSD capable on alt interface settings > 1. Is that assumption correct or should I narrow it down to the product ID of the MPD-3? Thanks, Daniel

1 0

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 27 Mar '13

27 Mar '13

On 27.03.2013 06:50, Andreas Koch wrote: > Hi Daniel > >> Ok. So we could define 3 DSD formats, for 8-bit, 16-bit and 32-bit. On >> that three formats, arbitrary samples can be transported, and the >> mapping between the configured sample rate and the resulting DSD rates >> would be as follows (please tell me if I'm correct): >> >> 352.8kHz 705.6KHz 1411.2KHz >> >> 8-bit 2.8MHz 5.6MHz 11.2MHz >> 16-bit 5.6MHz 11.2MHz >> 32-bit 11.2MHz >> >> So driver would mark a stream as DSD capable (8, 16, 32bit, depending on >> the USB descriptors), and then leave it up to the application to do the >> sample frequency determination according to the above table, and also as >> stated in the DoP document. Other drivers for more sound cards >> (presumable on other bus types) might follow. >> >> I think it's ok to handle it like this, because after all, what an >> application sets as sample rate for the stream is the number of actual >> words that are transported, where as the DSD rate is referring to bits/s. >> >> Jussi, for your application, that means that you'd have to add these 3 >> new formats, and teach your application that they can be used for DSD >> right away. >> >> The question is whether all DSD-capable USB DACs behave correctly in >> order to make this logic work. Andreas? > > If I understand this correctly then your driver would send the DSD > data in whatever format the DAC requests via its descriptors. The DSD > rate would either control the word length (8,16,32) or the underlying > sample rate (1x, 2x, 4x 352.8kHz). That would be the most flexible > implementation and I don't see why any hardware implementation would > have a problem with that. The question is whether all DACs would assume the same DSD rate as in the table above (ie, 8-bit, 352.8kHz == 2.8MHz DSD). And do we actually need a 32-bit format at all or is 8 and 16 enough? > The only potential fly in the ointment is > that we assume that the raw data format means DSD. In other words, if > a DAC uses raw data for any other purpose with similar wordlength > and/or sample rate spec. then we are in trouble. The chance for that > is small though. As there's nothing the UAC2 spec helps us with, we have to live with quirks for that. Daniel

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Clemens Ladisch 27 Mar '13

27 Mar '13

Andreas Koch wrote: > ... your driver would send the DSD data in whatever format the DAC > requests via its descriptors. [...] The only potential fly in the > ointment is that we assume that the raw data format means DSD. Add some vendor-specific descriptor to specify that raw means DSD. Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: export snd_soc_register_component
by Stephen Warren 27 Mar '13

27 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> Without this, modules will fail to link against those symbols. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ae68b0..561d589 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4195,6 +4195,7 @@ error_component_name: return ret; } +EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core @@ -4220,6 +4221,7 @@ found: dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); kfree(cmpnt->name); } +EXPORT_SYMBOL_GPL(snd_soc_unregister_component); /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, -- 1.7.10.4

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[alsa-devel] [PATCH 0/53] ASoC: switch over to use snd_soc_register_component()
by Kuninori Morimoto 26 Mar '13

26 Mar '13

Hi Mark, Liam, Lars, Stephen These patches switch over to use snd_soc_register_component() instead of snd_soc_register_dai[s]() on all drivers Last patch moves snd_soc_[un]register_dai[s]() to non global function. I could compile test on some of them, but couldn't for all drivers These patches are based on asoc/topic/component branch Kuninori Morimoto (53): ASoC: switch over to use snd_soc_register_component() on atmel ssc ASoC: switch over to use snd_soc_register_component() on au1x i2sc ASoC: switch over to use snd_soc_register_component() on au1x psc-ac97 ASoC: switch over to use snd_soc_register_component() on au1x psc-i2s ASoC: switch over to use snd_soc_register_component() on au1x ac97c ASoC: switch over to use snd_soc_register_component() on bf6xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx ac97 ASoC: switch over to use snd_soc_register_component() on bf5xx i2s ASoC: switch over to use snd_soc_register_component() on bf5xx tdm ASoC: switch over to use snd_soc_register_component() on ep93xx ac97 ASoC: switch over to use snd_soc_register_component() on ep93xx i2s ASoC: switch over to use snd_soc_register_component() on davinci i2s ASoC: switch over to use snd_soc_register_component() on davinci vcif ASoC: switch over to use snd_soc_register_component() on davinci mcasp ASoC: switch over to use snd_soc_register_component() on dw i2s ASoC: switch over to use snd_soc_register_component() on mpc5200 ac97 ASoC: switch over to use snd_soc_register_component() on mpc5200 i2s ASoC: switch over to use snd_soc_register_component() on fsl ssi ASoC: switch over to use snd_soc_register_component() on imx ssi ASoC: switch over to use snd_soc_register_component() on jz4740 i2s ASoC: switch over to use snd_soc_register_component() on kirkwood i2s ASoC: switch over to use snd_soc_register_component() on sst ASoC: switch over to use snd_soc_register_component() on mxs saif ASoC: switch over to use snd_soc_register_component() on nuc900 ac97 ASoC: switch over to use snd_soc_register_component() on omap mcbsp ASoC: switch over to use snd_soc_register_component() on omap mcpdm ASoC: switch over to use snd_soc_register_component() on omap dmic ASoC: switch over to use snd_soc_register_component() on omap hdmi ASoC: switch over to use snd_soc_register_component() on pxa ssp ASoC: pxa2xx-ac97: move EXPORT_SYMBOL_GPL() next to definition ASoC: switch over to use snd_soc_register_component() on pxa2xx ac97 ASoC: switch over to use snd_soc_register_component() on pxa2xx i2s ASoC: switch over to use snd_soc_register_component() on mmp sspa ASoC: switch over to use snd_soc_register_component() on s6000 i2s ASoC: switch over to use snd_soc_register_component() on s3c24xx i2s ASoC: switch over to use snd_soc_register_component() on s3c i2s ASoC: switch over to use snd_soc_register_component() on s3c ac97 ASoC: switch over to use snd_soc_register_component() on samsung spdif ASoC: switch over to use snd_soc_register_component() on samsung i2s ASoC: switch over to use snd_soc_register_component() on samsung pcm ASoC: switch over to use snd_soc_register_component() on goni_wm8994 ASoC: switch over to use snd_soc_register_component() on sh4 ssi ASoC: switch over to use snd_soc_register_component() on sh4 siu ASoC: switch over to use snd_soc_register_component() on sh4 hac ASoC: switch over to use snd_soc_register_component() on spear spdif out ASoC: switch over to use snd_soc_register_component() on spear spdif in ASoC: switch over to use snd_soc_register_component() on tegra30 i2s ASoC: switch over to use snd_soc_register_component() on tegra20 spdif ASoC: switch over to use snd_soc_register_component() on tegra20 ac97 ASoC: switch over to use snd_soc_register_component() on tegra20 i2s ASoC: switch over to use snd_soc_register_component() on txx9aclc ac97 ASoC: switch over to use snd_soc_register_component() on ux500 msp ASoC: snd_soc_[un]register_dai[s]() become non global function include/sound/soc-dai.h | 8 -------- sound/soc/atmel/atmel_ssc_dai.c | 11 ++++++++--- sound/soc/au1x/ac97c.c | 9 +++++++-- sound/soc/au1x/i2sc.c | 9 +++++++-- sound/soc/au1x/psc-ac97.c | 9 +++++++-- sound/soc/au1x/psc-i2s.c | 9 +++++++-- sound/soc/blackfin/bf5xx-ac97.c | 9 +++++++-- sound/soc/blackfin/bf5xx-i2s.c | 9 +++++++-- sound/soc/blackfin/bf5xx-tdm.c | 9 +++++++-- sound/soc/blackfin/bf6xx-i2s.c | 9 +++++++-- sound/soc/cirrus/ep93xx-ac97.c | 9 +++++++-- sound/soc/cirrus/ep93xx-i2s.c | 9 +++++++-- sound/soc/davinci/davinci-i2s.c | 15 ++++++++++----- sound/soc/davinci/davinci-mcasp.c | 15 ++++++++++----- sound/soc/davinci/davinci-vcif.c | 11 ++++++++--- sound/soc/dwc/designware_i2s.c | 9 +++++++-- sound/soc/fsl/fsl_ssi.c | 11 ++++++++--- sound/soc/fsl/imx-ssi.c | 11 ++++++++--- sound/soc/fsl/mpc5200_psc_ac97.c | 8 ++++++-- sound/soc/fsl/mpc5200_psc_i2s.c | 9 +++++++-- sound/soc/jz4740/jz4740-i2s.c | 9 +++++++-- sound/soc/kirkwood/kirkwood-i2s.c | 11 ++++++++--- sound/soc/mid-x86/sst_platform.c | 8 ++++++-- sound/soc/mxs/mxs-saif.c | 11 ++++++++--- sound/soc/nuc900/nuc900-ac97.c | 9 +++++++-- sound/soc/omap/omap-dmic.c | 9 +++++++-- sound/soc/omap/omap-hdmi.c | 9 +++++++-- sound/soc/omap/omap-mcbsp.c | 9 +++++++-- sound/soc/omap/omap-mcpdm.c | 9 +++++++-- sound/soc/pxa/mmp-sspa.c | 9 +++++++-- sound/soc/pxa/pxa-ssp.c | 9 +++++++-- sound/soc/pxa/pxa2xx-ac97.c | 11 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 9 +++++++-- sound/soc/s6000/s6000-i2s.c | 9 +++++++-- sound/soc/samsung/ac97.c | 12 ++++++++---- sound/soc/samsung/goni_wm8994.c | 11 ++++++++--- sound/soc/samsung/i2s.c | 14 ++++++++++---- sound/soc/samsung/pcm.c | 11 ++++++++--- sound/soc/samsung/s3c-i2s-v2.c | 9 +++++---- sound/soc/samsung/s3c-i2s-v2.h | 7 ++++--- sound/soc/samsung/s3c2412-i2s.c | 12 +++++++++--- sound/soc/samsung/s3c24xx-i2s.c | 11 ++++++++--- sound/soc/samsung/spdif.c | 11 ++++++++--- sound/soc/sh/hac.c | 10 +++++++--- sound/soc/sh/siu_dai.c | 11 ++++++++--- sound/soc/sh/ssi.c | 10 +++++++--- sound/soc/soc-core.c | 12 ++++-------- sound/soc/spear/spdif_in.c | 9 +++++++-- sound/soc/spear/spdif_out.c | 9 +++++++-- sound/soc/tegra/tegra20_ac97.c | 15 ++++++++++----- sound/soc/tegra/tegra20_i2s.c | 15 ++++++++++----- sound/soc/tegra/tegra20_spdif.c | 15 ++++++++++----- sound/soc/tegra/tegra30_i2s.c | 15 ++++++++++----- sound/soc/txx9/txx9aclc-ac97.c | 9 +++++++-- sound/soc/ux500/ux500_msp_dai.c | 13 +++++++++---- 55 files changed, 405 insertions(+), 165 deletions(-)

6 137

[alsa-devel] [PATCH 1/2] ASoC: compress - add support for metadata apis
by Vinod Koul 26 Mar '13

26 Mar '13

From: Jeeja KP <jeeja.kp(a)intel.com> Compress core added metadata apis in 9727b4, so add same in ASoC Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Rebased on asoc-next sound/soc/soc-compress.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b5b3db7..f9b2197 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -322,11 +322,38 @@ static int soc_compr_copy(struct snd_compr_stream *cstream, return ret; } +static int sst_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->set_metadata) + ret = platform->driver->compr_ops->set_metadata(cstream, metadata); + + return ret; +} + +static int sst_compr_get_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->get_metadata) + ret = platform->driver->compr_ops->get_metadata(cstream, metadata); + + return ret; +} /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_metadata = sst_compr_set_metadata, + .get_metadata = sst_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

2 2

[alsa-devel] [PATCH 00/14] ASoC: Cleanup dmaengine PCM users
by Lars-Peter Clausen 26 Mar '13

26 Mar '13

Hi, This series cleans up a few of the current users of the dmaengine pcm lib in order to make the introduction of a new more generic PCM dmaengine driver more straight forward. The series consists mainly of two types of changes. One is removing all unnecessary wrappers of snd_soc_dmaengine_close(). For some reason quite a few drivers started providing their own pcm_close callback, which does nothing but calls snd_soc_dmaengine_close, although snd_soc_dmaengine_close can be used directly as the pcm_close callback if nothing else needs to be done. The other is getting rid of snd_dmaengine_pcm_{get,set}_data(). These two functions were initially introduced to deal with a few oddball dmaengine drivers which require that data is passed to them via the private_data field of the channel and we had to keep the data allocated until the channel had been released. A few other driver started using these functions although there is no need to do so. E.g. some drivers just call snd_dmaengine_pcm_set_data(), but never attempt to read the data again using snd_dmaengine_pcm_get_data(). For the drivers, for which snd_dmaengine_pcm_{set,get}_data() was originally introduced, the responsibility of keeping the dma config allocated is shifted from the pcm driver to the i2s drivers. Which in turn makes the code much more straight forward since for most i2s drivers the dma config is either static or setup once at driver probe time. The series has only been compile tested, so please test these patches, especially the non trivial ones for ep93xx, mmp, imx and mxs. - Lars Lars-Peter Clausen (14): ASoC: ux500_pcm: Remove duplicated SNDRV_PCM_HW_PARAM_PERIODS constraint ASoC: spear_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: omap-pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: tegra_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to wrap snd_dmaengine_pcm_close() ASoC: atmel-pcm-dma: No need to wrap snd_dmaengine_pcm_close() ASoC: ux500_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: speaer_pcm: No need to use snd_dmaengine_pcm_set_data() ASoC: atmel-pcm-dma: Do not use snd_dmaengine_pcm_{set,get}_data() ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params ASoC: mmp-pcm: Allocate dma filter parameters on the stack ASoC: imx-pcm: Embed the imx_dma_data struct in the dma_params struct ASoC: mxs: Embed the mxs_dma_data struct in the mxs_pcm_dma_params struct ASoC: dmaengine-pcm: Remove snd_dmaengine_pcm_{set,get}_data include/sound/dmaengine_pcm.h | 3 --- sound/soc/atmel/atmel-pcm-dma.c | 23 +++++++--------------- sound/soc/cirrus/edb93xx.c | 1 - sound/soc/cirrus/ep93xx-ac97.c | 9 +++++---- sound/soc/cirrus/ep93xx-i2s.c | 12 ++++++------ sound/soc/cirrus/ep93xx-pcm.c | 37 +++-------------------------------- sound/soc/cirrus/ep93xx-pcm.h | 20 ------------------- sound/soc/cirrus/simone.c | 2 -- sound/soc/cirrus/snappercl15.c | 1 - sound/soc/fsl/fsl_ssi.c | 17 ++++++++-------- sound/soc/fsl/imx-pcm-dma.c | 35 ++------------------------------- sound/soc/fsl/imx-pcm.h | 17 ++++++++++++++-- sound/soc/fsl/imx-ssi.c | 12 ++++++++---- sound/soc/mxs/mxs-pcm.c | 43 +++++------------------------------------ sound/soc/mxs/mxs-pcm.h | 4 +++- sound/soc/mxs/mxs-saif.c | 6 +++--- sound/soc/omap/omap-pcm.c | 8 +------- sound/soc/pxa/mmp-pcm.c | 33 +++++-------------------------- sound/soc/soc-dmaengine-pcm.c | 29 --------------------------- sound/soc/spear/spear_pcm.c | 18 ++--------------- sound/soc/tegra/tegra_pcm.c | 8 +------- sound/soc/ux500/ux500_pcm.c | 25 +----------------------- 22 files changed, 76 insertions(+), 287 deletions(-) delete mode 100644 sound/soc/cirrus/ep93xx-pcm.h -- 1.8.0

9 30

[alsa-devel] [PATCH] ASoC: omap: Call omap_mcbsp_set_threshold() from mcbsp hw_params
by Lars-Peter Clausen 26 Mar '13

26 Mar '13

The omap PCM driver provides a set_threshold callback which gets called by the PCM driver when either playback or capture is started. The only DAI driver which sets this callback is the mcbsp driver. This patch removes the callback from the PCM driver and moves the invocation of the omap_mcbsp_set_threshold() function to the mcbsp hw_params callback since this is the only place where the threshold size can change. Doing so allows us to use the default dmaengine PCM trigger callback in the omap PCM driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Only compile tested, but I don't see why it should not work Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-mcbsp.c | 12 +++++------- sound/soc/omap/omap-pcm.c | 33 +-------------------------------- sound/soc/omap/omap-pcm.h | 1 - 3 files changed, 6 insertions(+), 40 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8d2defd..406fc87 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -62,24 +62,22 @@ enum { * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) +static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, + unsigned int packet_size) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_pcm_dma_data *dma_data; int words; - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or based on the * period size in THRESHOLD mode, otherwise use McBSP threshold = 1 * for mono streams. */ - if (dma_data->packet_size) - words = dma_data->packet_size; + if (packet_size) + words = packet_size; else words = 1; @@ -245,7 +243,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } if (mcbsp->pdata->buffer_size) { - dma_data->set_threshold = omap_mcbsp_set_threshold; if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; int divider = 0; @@ -276,6 +273,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Use packet mode for non mono streams */ pkt_size = channels; } + omap_mcbsp_set_threshold(substream, pkt_size); } dma_data->packet_size = pkt_size; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index c722c2e..1626fb7 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -129,37 +129,6 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct omap_pcm_dma_data *dma_data; - int ret = 0; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* Configure McBSP internal buffer usage */ - if (dma_data->set_threshold) - dma_data->set_threshold(substream); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - default: - ret = -EINVAL; - } - - if (ret == 0) - ret = snd_dmaengine_pcm_trigger(substream, cmd); - - return ret; -} - static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) { snd_pcm_uframes_t offset; @@ -208,7 +177,7 @@ static struct snd_pcm_ops omap_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = omap_pcm_hw_params, .hw_free = omap_pcm_hw_free, - .trigger = omap_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = omap_pcm_pointer, .mmap = omap_pcm_mmap, }; diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index cabe74c..39e6e45 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -31,7 +31,6 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - void (*set_threshold)(struct snd_pcm_substream *substream); int data_type; /* 8, 16, 32 (bits) or 0 to let omap-pcm * to decide the sDMA data type */ int packet_size; /* packet size only in PACKET mode */ -- 1.8.0

3 2

Re: [alsa-devel] [PATCH] ASoC: core: Fix to check return value of snd_soc_update_bits_locked()
by Mark Brown 26 Mar '13

26 Mar '13

On Tue, Mar 26, 2013 at 02:41:05PM +0900, Joonyoung Shim wrote: > It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is > success. So just check return value is negative. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH -next] ASoC: arizona: remove duplicated include from arizona.c
by Mark Brown 26 Mar '13

26 Mar '13

On Tue, Mar 26, 2013 at 02:47:08PM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Remove duplicated include. Applied, thanks.

1 0

[alsa-devel] ALSA regression Macbook Air 5, 2: No sound from the loudspeaker
by Daniel Wagner 26 Mar '13

26 Mar '13

Hi, I experience a regression on my Macbook Air 2012 model with the 3.9-rc kernels (v3.9-rc2-284-gaea8b5d) The symptoms are that there is just no sound from the loudspeakers. The headphone output works fine. I was able to pin point it down to one commit 1077a024 "ALSA: hda - Use generic parser for Cirrus codec" If I revert this one and all came back to work. By comparing the alsa-info.txt output I figured that there is something wrong with the GPIO settings on node 0x01: This is the excerpt with the previous patch mentioned reverted: Codec: Cirrus Logic CS4206 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10134206 Subsystem Id: 0x106b5600 Revision Id: 0x100302 No Modem Function Group found Default PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=0, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 and here the output vanilla kernel: Codec: Cirrus Logic CS4206 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10134206 Subsystem Id: 0x106b5600 Revision Id: 0x100302 No Modem Function Group found Default PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D3 CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=0, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 The diff is "IO[3] data=1 vs data=0". I used then the patch function from the awesome hda-analyzer tool to patch this on the vanilla kernel and there suddenly I have output from the speakers. I don't know if someone else has reported this already or if there is already a fix. I was without internet for few days and didn't report this in time. If that is the case, sorry for the noise. The complete logs from hda-analyzer are here: http://www.monom.org/misc/alsa/alsa-info.txt-v3.9-rc2-284-gaea8b5d http://www.monom.org/misc/alsa/alsa-info.txt-revert-1077a024 cheers, daniel

1 1

[alsa-devel] channel swapping issue on OMAP3/TWL4030 is back
by Peter Meerwald 26 Mar '13

26 Mar '13

Hello, I'm running Linux 3.7 on a beagle-xm and observe very reproducible stereo channel swapping issue on playback of a stereo stream (one channel has a sine, the other is zero); the channel swap occurs on starting the playback with a scope I have observed that the audio data indeed swaps relative to the FSX; so the problem is on the I2S bus I see the issue with the TWL4030 audio codec; McBSP2 is slave, TWL4030 is master on a custom board I see the same issue with a TLV320AIC3104 on McBSP3 (have tried to configured the TLV320AIC3104 as slave and master, same channel swapping issue for both configs) the channel swapping issue has been discussed here, http://comments.gmane.org/gmane.linux.alsa.devel/82178, and patches have been pointed out for 2.6.32 it appears that the swapping issue is back, probably due to the DMA rework in recent kernel? can someone confirm the issue? any suggestions? thanks, regards, p. -- Peter Meerwald +43-664-2444418 (mobile)

4 11

[alsa-devel] [bug] sbawe: the PnP won't stop
by David Flater 26 Mar '13

26 Mar '13

I've got an ISA PnP SB AWE type card in a system with a non-working DMA 5. Without intervention, the Sound Blaster always gets the broken DMA. I have no BIOS options to control the allocation of DMAs, and the PnP configuration done within ALSA ignores /etc/isapnp.conf [*]. DMA 6 works fine if I can get sbawe to stop re-initializing the PnP and just use it. # Use setup from isapnp with dma16 moved to 6. modprobe snd-sbawe isapnp=0 port=0x220 mpu_port=0x330 fm_port=0x388 awe_port=0x620 irq=5 dma8=1 dma16=6 modprobe snd-emu8000-synth Problem is, regardless whether isapnp is 0 or 1, sbawe always PnPs the card and puts it back on broken DMA 5. If I have already got the card correctly configured with /sbin/isapnp beforehand, sbawe leaves the original resources tied up as a ghost card and allocates again (/proc/dma): 1: SoundBlaster - 8bit 2: floppy 3: SoundBlaster - 8bit 4: cascade 5: SoundBlaster - 16bit 6: SoundBlaster - 16bit To get it working I ripped the following lines of code out of kernel 3.8.4, blocking PnP completely. As for a proper nondestructive fix, I'm not sure whether the intent was (A) you were only supposed to get one of the two registrations, or (B) the PnP registration was not supposed to reinit everything if it was already set up. --- sound/isa/sb/sb16.c~ 2013-03-14 14:27:14.000000000 -0400 +++ sound/isa/sb/sb16.c 2013-03-23 20:18:50.137987332 -0400 @@ -674,13 +674,6 @@ #ifdef CONFIG_PNP if (!err) isa_registered = 1; - - err = pnp_register_card_driver(&sb16_pnpc_driver); - if (!err) - pnp_registered = 1; - - if (isa_registered) - err = 0; #endif return err; } [*] So is there a different config file for PnP within ALSA? -- DWF

1 1

Re: [alsa-devel] BUG: soft lockup when recording audio on MX28EVK with ASoC sgtl5000
by Fabio Estevam 26 Mar '13

26 Mar '13

Hi Hector, On Mon, Mar 25, 2013 at 7:52 AM, Hector Palacios <hector.palacios(a)digi.com> wrote: > Hello, > > I just tried recording audio on Freescale's MX28EVK that uses ASoC sgtl5000 > (kernel v3.8) with: > > arecord -M -f cd sound.wav --duration 10 > > and got a scheduler message: > > [ 789.041847] [sched_delayed] sched: RT throttling activated I have just tested arecord on 3.8,4 and I haven't seen the 'soft lockup' bug. However, I see that mxs saif gets busy after the record and any attempt to do an 'aplay' after the first arecord fails. With the attached patch applied I am able to do arecord/aplay sequence several times. I don't have a cable handy here to actually test if the sound is recorded properly or not. Could you please test it against 3.8.4? Thanks, Fabio Estevam

1 1

[alsa-devel] [PATCH 1/2] ASoC: tegra: add Tegra114 support to the AHUB driver
by Stephen Warren 25 Mar '13

25 Mar '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra114's AHUB shares a design with Tegra30, with the followin changes: * Supports more (10 vs. 4) bi-directional FIFO channels into RAM. * Requires a separate block of registers to support the above. * Supports more attached clients, i.e. new audio multiplexing and de-multiplexing modules. * Is affected by more clocks due to the above. This change fully defines the device tree binding changes required to represent these changes, and minimally extends the driver to support the new hardware, without exposing any of the new FIFO channels. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- .../bindings/sound/nvidia,tegra30-ahub.txt | 26 ++++++-- sound/soc/tegra/tegra30_ahub.c | 64 ++++++++++++++------ sound/soc/tegra/tegra30_ahub.h | 16 +++++ 3 files changed, 84 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt index 1ac7b16..0e5c12c 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-ahub.txt @@ -1,12 +1,22 @@ NVIDIA Tegra30 AHUB (Audio Hub) Required properties: -- compatible : "nvidia,tegra30-ahub" +- compatible : "nvidia,tegra30-ahub", "nvidia,tegra114-ahub", etc. - reg : Should contain the register physical address and length for each of - the AHUB's APBIF registers and the AHUB's own registers. + the AHUB's register blocks. + - Tegra30 requires 2 entries, for the APBIF and AHUB/AUDIO register blocks. + - Tegra114 requires an additional entry, for the APBIF2 register block. - interrupts : Should contain AHUB interrupt -- nvidia,dma-request-selector : The Tegra DMA controller's phandle and - request selector for the first APBIF channel. +- nvidia,dma-request-selector : A list of the DMA channel specifiers. Each + entry contains the Tegra DMA controller's phandle and request selector. + If a single entry is present, the request selectors for the channels are + assumed to be contiguous, and increment from this value. + If multiple values are given, one value must be given per channel. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: + - Tegra30: Requires d_audio, apbif, i2s0, i2s1, i2s2, i2s3, i2s4, dam0, + dam1, dam2, spdif_in. + - Tegra114: Additionally requires amx, adx. - ranges : The bus address mapping for the configlink register bus. Can be empty since the mapping is 1:1. - #address-cells : For the configlink bus. Should be <1>; @@ -25,7 +35,13 @@ ahub@70080000 { reg = <0x70080000 0x200 0x70080200 0x100>; interrupts = < 0 103 0x04 >; nvidia,dma-request-selector = <&apbdma 1>; - + clocks = <&tegra_car 106>, <&tegra_car 107>, <&tegra_car 30>, + <&tegra_car 11>, <&tegra_car 18>, <&tegra_car 101>, + <&tegra_car 102>, <&tegra_car 108>, <&tegra_car 109>, + <&tegra_car 110>, <&tegra_car 162>; + clock-names = "d_audio", "apbif", "i2s0", "i2s1", "i2s2", + "i2s3", "i2s4", "dam0", "dam1", "dam2", + "spdif_in"; ranges; #address-cells = <1>; #size-cells = <1>; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index e5cfb4a..4405c3a 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -287,16 +287,27 @@ int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif) } EXPORT_SYMBOL_GPL(tegra30_ahub_unset_rx_cif_source); -static const char * const configlink_clocks[] = { - "i2s0", - "i2s1", - "i2s2", - "i2s3", - "i2s4", - "dam0", - "dam1", - "dam2", - "spdif_in", +#define CLK_LIST_MASK_TEGRA30 BIT(0) +#define CLK_LIST_MASK_TEGRA114 BIT(1) + +#define CLK_LIST_MASK_TEGRA30_OR_LATER \ + (CLK_LIST_MASK_TEGRA30 | CLK_LIST_MASK_TEGRA114) + +static const struct { + const char *clk_name; + u32 clk_list_mask; +} configlink_clocks[] = { + { "i2s0", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s1", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s2", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s3", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "i2s4", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam0", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam1", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "dam2", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "spdif_in", CLK_LIST_MASK_TEGRA30_OR_LATER }, + { "amx", CLK_LIST_MASK_TEGRA114 }, + { "adx", CLK_LIST_MASK_TEGRA114 }, }; #define LAST_REG(name) \ @@ -424,8 +435,24 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .cache_type = REGCACHE_RBTREE, }; +static struct tegra30_ahub_soc_data soc_data_tegra30 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA30, +}; + +static struct tegra30_ahub_soc_data soc_data_tegra114 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA114, +}; + +static const struct of_device_id tegra30_ahub_of_match[] = { + { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 }, + { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 }, + {}, +}; + static int tegra30_ahub_probe(struct platform_device *pdev) { + const struct of_device_id *match; + const struct tegra30_ahub_soc_data *soc_data; struct clk *clk; int i; struct resource *res0, *res1, *region; @@ -436,16 +463,24 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (ahub) return -ENODEV; + match = of_match_device(tegra30_ahub_of_match, &pdev->dev); + if (!match) + return -EINVAL; + soc_data = match->data; + /* * The AHUB hosts a register bus: the "configlink". For this to * operate correctly, all devices on this bus must be out of reset. * Ensure that here. */ for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { - clk = clk_get(&pdev->dev, configlink_clocks[i]); + if (!(configlink_clocks[i].clk_list_mask & + soc_data->clk_list_mask)) + continue; + clk = clk_get(&pdev->dev, configlink_clocks[i].clk_name); if (IS_ERR(clk)) { dev_err(&pdev->dev, "Can't get clock %s\n", - configlink_clocks[i]); + configlink_clocks[i].clk_name); ret = PTR_ERR(clk); goto err; } @@ -592,11 +627,6 @@ static int tegra30_ahub_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id tegra30_ahub_of_match[] = { - { .compatible = "nvidia,tegra30-ahub", }, - {}, -}; - static const struct dev_pm_ops tegra30_ahub_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_ahub_runtime_suspend, tegra30_ahub_runtime_resume, NULL) diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index e690e2e..7189be9 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -468,7 +468,23 @@ extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); +struct tegra30_ahub_soc_data { + u32 clk_list_mask; + /* + * FIXME: There are many more differences in HW, such as: + * - More APBIF channels. + * - Extra separate chunks of register address space to represent + * the extra APBIF channels. + * - More units connected to the AHUB, so that tegra30_ahub_[rt]xcif + * need expansion, coupled with there being more defined bits in + * the AHUB routing registers. + * However, the driver doesn't support those new features yet, so we + * don't represent them here yet. + */ +}; + struct tegra30_ahub { + const struct tegra30_ahub_soc_data *soc_data; struct device *dev; struct clk *clk_d_audio; struct clk *clk_apbif; -- 1.7.10.4

2 4

[alsa-devel] [PATCH] ASoC: imx-sgtl5000: use of_node to match cpu dai
by Shawn Guo 25 Mar '13

25 Mar '13

Since imx-sgtl5000 is only used on DT platform, it makes more sense to use cpu_of_node rather than cpu_dai_name to match cpu dai. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 424347e..9584e78 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -148,7 +148,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->dai.stream_name = "HiFi"; data->dai.codec_dai_name = "sgtl5000"; data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.cpu_of_node = ssi_np; data->dai.platform_name = "imx-pcm-audio"; data->dai.init = &imx_sgtl5000_dai_init; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 1.7.9.5

2 1

Re: [alsa-devel] [Alsa-user] Audio-gd NFB-17.32 (VIA VT1731)
by Torstein Hegge 25 Mar '13

25 Mar '13

On Mon, Mar 25, 2013 at 08:38:16 +0100, Clemens Ladisch wrote: > Torstein Hegge wrote: > > I think we can conclude that the VIA VT1731 has the same issue as the > > C-Media CM6631. > > > > I'll add the VT1731 as one of the devices that needs a reset after > > sample rate change in the next iteration of the sample rate workaround > > patch. > > Why not just do it for all UAC 2 devices? If new devices that needs this just keeps popping up, that might indeed be the better solution. Torstein

1 0

[alsa-devel] No dmix/dsnoop on Intel ICH4/5 by default?
by Rene Herman 24 Mar '13

24 Mar '13

Good day. I'm out of touch of with alsa -- but should dmix/dsnoop still be enabled by default on hardware that doesn't do hardware mixing? It seems it isn't on my Intel ICH4: $ cat /proc/asound/cards 0 [ICH5 ]: ICH4 - Intel ICH5 Intel ICH5 with AD1980 at irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7134[0] at 0xe8200000 irq 22 I believe that it should be from /usr/share/alsa/cards/ICH4.conf, but I don't get mixing on this setup when using the "default" pcm... (which I may not mind at that much, but I was looking at some other sound-related problem and this has me rather confused) This is arch linux, with alsa-lib, -plugins and -utils 1.0.26 installed, using kernel 3.6.11-1-ARCH (ie, standard). Regards, Rene

4 11

[alsa-devel] [PATCH v4] ALSA: usb: Work around CM6631 sample rate change bug
by Torstein Hegge 24 Mar '13

24 Mar '13

The C-Media CM6631 USB-to-S/PDIF receiver doesn't respond to changes in sample rate while the interface is active. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all current C-Media UAC V2.0 devices, including CM6610, CM6620 and CM6631, but has only been tested with CM6631. Signed-off-by: Torstein Hegge <hegge(a)resisty.net> --- Changes since v3: - Use given target rate instead of the new rate read from device when checking if a reset is needed. Some devices (at least Schiit Bifrost) doesn't report a reliable sample rate back right after setting a new rate. This appears to fix the issues with v3 of this patch with the Schiit, as reported by Chris Hermansen in http://thread.gmane.org/gmane.linux.alsa.user/36935/focus=37284 Changes since v2: - More USB ids covered. - int crate -> int cur_rate - unsigned int previous_rate -> int prev_rate sound/usb/clock.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 46 insertions(+), 4 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2..60c2775 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,18 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data))) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -290,9 +302,39 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + switch (chip->usb_id) { + /* C-Media CM6610/CM6620/CM6631 */ + case USB_ID(0x054c, 0x06cf): /* Sony */ + case USB_ID(0x0b05, 0x17a8): /* Asus Xonar Essence One */ + case USB_ID(0x0d8c, 0x0301): + case USB_ID(0x0d8c, 0x0302): + case USB_ID(0x0d8c, 0x0304): /* CM6631 (Schiit) */ + case USB_ID(0x0d8c, 0x0305): + case USB_ID(0x0d8c, 0x0306): + case USB_ID(0x0d8c, 0x0309): /* CM6631 */ + case USB_ID(0x0d8c, 0x0310): + case USB_ID(0x0d8c, 0x0311): /* CM6610A */ + case USB_ID(0x0d8c, 0x0312): /* CM6620A */ + case USB_ID(0x0d8c, 0x0313): /* CM6630A */ + case USB_ID(0x0d8c, 0x0314): /* CM6631A */ + case USB_ID(0x0d8c, 0x0315): /* CM6632A */ + case USB_ID(0x0d8c, 0x0319): /* CM6631A */ + case USB_ID(0x200c, 0x1030): /* Reloop */ + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + break; + } + } return 0; } -- 1.7.10.4

2 2

Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 23 Mar '13

23 Mar '13

On 03/22/2013 06:05 PM, Andreas Koch wrote: > The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for > 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz on > the DAC, I see, this is different from how I've been dealing with it. Since I always use uint8 format, but instead vary the rate to 352.8/705.6/etc... Just because it's much more straightforward form the application's perspective -> output rate is always bit rate divided by eight (=packet rate). And switch between uint8 / uint32 sample format equals to switch between DSD <-> PCM. - Jussi

1 0

[alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Daniel Mack 23 Mar '13

23 Mar '13

Thanks to Andreas, I got my hands on a Playback Design MPD-3 and hacked up some small hanges necessary to make it fully work with snd-usb. The device reports 0x80000000 in the bmFormats field of two of three of its alternate settings, which wrongly made the driver believe it's a usual PCM endpoint (actually due to a fix-up fallback). However, bit 31 of this mask in fact denotes 'raw data'. The effect of this issue is that in addition to the first altsetting with a bit depth of 24, the driver exposed 8-bit and 16-bit native audio formats on the raw data endpoints, which do not work as expected. Also, those devices need a 50ms delay after switching the USB interface. Thanks, Daniel Daniel Mack (3): ALSA: snd-usb: handle the bmFormats field as unsigned int ALSA: snd-usb: handle raw data format of UAC2 devices ALSA: snd-usb: add delay quirk for "Playback Design" products include/linux/usb/audio-v2.h | 2 ++ sound/usb/format.c | 20 ++++++++++++-------- sound/usb/format.h | 2 +- sound/usb/pcm.c | 7 +++++++ sound/usb/stream.c | 2 +- 5 files changed, 23 insertions(+), 10 deletions(-) -- 1.7.10.4

4 16

Re: [alsa-devel] Support for DSD streams
by Daniel Mack 23 Mar '13

23 Mar '13

Hi Andreas, On 23.03.2013 18:35, Andreas Koch wrote: >> Yes, understood. The thing is just that we shouldn't break exising >> solutions that already rely on this "bandaid". > > You wouldn't break any existing solution based on DoP, because that > solution is dealt with in the application on one side and the DAC on > the other side. The driver and USB transfer both think it is PCM data > that is being sent. Well, that was exactly the problem with your device. With the driver as it was, it exposed a PCM interface which is 16-bit wide, and that was picked by ALSA when playing 16-bit material (because ALSA takes whatever is offered by the hardware natively). > The applications have a preference setting to > select DoP and likewise the DAC's constantly monitor the PCM > formatted data stream and automatically switch to DSD if they find > the pattern. Hmm, but your device behaves differently here, right? At least, when normal PCM material is sent to the DSD interface, it doesn't play it or switch back or something. And I think other DACs will have similar behaviour. I need to think about all that more deeply next week :) Daniel

1 0

Re: [alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

On 03/23/2013 02:28 PM, Laxman Dewangan wrote: > On Saturday 23 March 2013 06:59 PM, Lars-Peter Clausen wrote: >> On 03/23/2013 02:16 PM, Laxman Dewangan wrote: >>> On Saturday 23 March 2013 05:05 AM, Stephen Warren wrote: >>>> On 03/22/2013 10:26 AM, Lars-Peter Clausen wrote: >>>>> Hi, >>>>> >>>>> The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and >>>>> SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and >>>>> SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really >>>>> work? The old non-dmaengine based driver implemented this properly. I'd >>>>> reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it >>>>> where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with >>>>> the >>>>> dmaengine based driver the pointer always gets reset to 0 for both >>>>> SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. >>>> It's quite possible this hasn't been tested, or tested recently. I don't >>>> recall how to test PAUSE/RESUME. Can you enlighten me? >>>> >>>> Laxman, can you comment here? >>> The tegra20/Tegra30 does not support the pause of the channel. It was >>> through global pause. hence the pause and resume is implemented like stop >>> transfer and again restart transfer. hence you are seeing this as the reset >>> pointer as 0. >>> >> Yea, but in that case you shouldn't implement pause/resume at all. You've >> changed the semantics on pause/resume on which applications rely. If >> pause/resume are not implemented applications usually workaround it. >> >> - Lars >> > > So should we return -EINVAL in case of following? > > case SNDRV_PCM_TRIGGER_RESUME: > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > Don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME bits in the pcm_hardware struct and use snd_dmaengine_pcm_trigger as your trigger callback. The ALSA core will take care of everything else. - Lars

1 0

Re: [alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

On 03/23/2013 02:16 PM, Laxman Dewangan wrote: > On Saturday 23 March 2013 05:05 AM, Stephen Warren wrote: >> On 03/22/2013 10:26 AM, Lars-Peter Clausen wrote: >>> Hi, >>> >>> The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and >>> SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and >>> SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really >>> work? The old non-dmaengine based driver implemented this properly. I'd >>> reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it >>> where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with the >>> dmaengine based driver the pointer always gets reset to 0 for both >>> SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. >> It's quite possible this hasn't been tested, or tested recently. I don't >> recall how to test PAUSE/RESUME. Can you enlighten me? >> >> Laxman, can you comment here? > The tegra20/Tegra30 does not support the pause of the channel. It was > through global pause. hence the pause and resume is implemented like stop > transfer and again restart transfer. hence you are seeing this as the reset > pointer as 0. > Yea, but in that case you shouldn't implement pause/resume at all. You've changed the semantics on pause/resume on which applications rely. If pause/resume are not implemented applications usually workaround it. - Lars > So pause is stop and resume is the start again. > > If it is require to implement the dmaengine_pause/resume() then it can not > be done truely, it can be done atelast in resolution of one buffer transfer. > We can not stop dma in mid of buffer transfer and start from there. We can > do cancel of current buffer transfer and then start from begining of current > buffer, not from the point it was paused. > > However, tegra114 support this pause and resume truely.

1 0

[alsa-devel] Support for DSD streams (was: Re: [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work)
by Daniel Mack 23 Mar '13

23 Mar '13

Hi Andreas, On 22.03.2013 17:05, Andreas Koch wrote: > At 01:11 AM 3/22/2013, Daniel Mack wrote: >> Hi Jussi, >> Hi Andreas, >> >> On 21.03.2013 22:21, Jussi Laako wrote: >>> On 03/18/2013 03:49 PM, Daniel Mack wrote: >>>> Yes. If at all, we should add a SND_PCM_FORMAT_DSD, and a quirk for that >>>> device. But given that there is no application for DSD in userspace >>>> either, we probably don't need to care. >>> >>> Yes there is? At least my application. And I'd be happy to support any >>> custom format created for the purpose... :) >> >> Ah, interesting. So what *are* you using right now for your transport in >> userspace? Is your application abusing standard PCM interfaces that are >> 'known' to be in fact DSD? How do other DACs expose their interfaces in >> the Linux world, and which ones did you test your application with? >> >> If there's actual need for a new SND_PCM_FORMAT, it would be easy to >> patch it through all the layers down to userspace. >> >>> I would propose to create custom 8-bit sample formats for DSD purposes, >>> similar to the ASIO formats. Practically two main ones, byte with oldest >>> bit in MSB and byte with oldest bit in LSB. And maybe the "DSD-wide" one >>> which is 8-bit DSD sample. >> >> Andreas, which modes does the MPD-3 accept exactly on its two 'raw data' >> interfaces? The only hint I have is that one of it is 8-bit and the >> other is 16-bit, but given the lack of a proper test setup for this >> format, I can't currently test anything. > > The 8-bit raw format is for 2.8MHz DSD and the 16-bit format is for > 5.6MHz DSD. In both cases the driver sets the sample rate to 352.8kHz > on the DAC, but then sends either 8 bits or 16 bits depending on the > DSD rate. And prior to sending either 8 bits or 16 bits, currently the correct USB alternate interface has to be set, right? > Since 11.2MHz DSD is already on the horizon we may have to > change this to a single 8-bit raw interface and then vary the > underlying USB transfer sample rate between 1x, 2x, 4x 352.8kHz - Yes, especially given that if there's no common standard across different devices, either the drivers or the applications have to implement special handling code for this, and know which interface supports what. So for the ALSA layer, we should probably have a generic SND_PCM_FORMAT_DSD and then rely on the sample rate setting mechanisms to select the right DSD mode. Other solutions, like having a different SND_PCM_FORMAT_DSD_{2_8,5_6,11_2}_MHZ would be hairy to implement, as some devices need to get a sample rate update request, while others don't etc ... > that would be more logical and straight forward for the hardware > implementation. This is only supported through my own driver for the > Windows platform which uses the standard ASIO interface that supports > DSD at those rates. Since the Mac OS only supports PCM streams and > nothing else I brought together a bunch of DAC manufacturers and > playback software developers to standardize a method to "cannibalize" > the PCM stream for DSD data. The method is described in the attached > and is now supported by most manufacturers. Very interesting read. For ALSA though, it's no problem to support that in all layers and make the drivers tell applications about it. We just have to decide for a sane way :) > This also has been implemented on a few Linux platforms (Auraliti for > instance). Even though the DoP standard works really well so far, it > is a "bandaid" and should only be used if there is no other choice. Yes, understood. The thing is just that we shouldn't break exising solutions that already rely on this "bandaid". Daniel

1 0

[alsa-devel] ASoC: tegra: Does the PCM_TRIGGER_{PAUSE, RESUME} emulation really work?
by Lars-Peter Clausen 23 Mar '13

23 Mar '13

Hi, The tegra pcm driver tries to emulate SNDRV_PCM_TRIGGER_PAUSE and SNDRV_PCM_TRIGGER_RESUME. By passing SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_STOP to snd_dmaengine_pcm_trigger. But does this really work? The old non-dmaengine based driver implemented this properly. I'd reset the dma position pointer for SNDRV_PCM_TRIGGER_START, but keep it where it was for SNDRV_PCM_TRIGGER_RESUME. But as far as I can see with the dmaengine based driver the pointer always gets reset to 0 for both SNDRV_PCM_TRIGGER_START and SNDRV_PCM_TRIGGER_RESUME. - Lars

2 2

Re: [alsa-devel] [PATCH V2] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 22 Mar '13

22 Mar '13

On Fri, Mar 22, 2013 at 4:33 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 424347e..71ce1f6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -128,18 +128,30 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > goto fail; > } > > + ret = of_property_read_u32(codec_np, "clock-frequency", > + &data->clk_frequency); I just noticed that you assign 'ret' here, but you never check the value. So I think you should either: 1) Check the value of 'ret' 2) Remove the "ret = "

1 0

[alsa-devel] [PATCH 0/2] ASoC: dmaengine_pcm: support generic DMA binding users
by Shawn Guo 22 Mar '13

22 Mar '13

The series adds a snd_dmaengine_generic_pcm_open() for users that adopt generic DMA binding and helplers. Vinod, AS per the request from Mark, I add the first patch to mark name parameter of dmaengine helpers as const, so that clients can mark const on their side. We need to have these two patches on a topic branch, as I've heard a few people converting their ASoC driver needing the patches. Can you ack the first patch, so that Mark can maintain them in a branch for anyone who needs it to pull? Shawn Guo (2): dmaengine: add const for name parameter ASoC: dmaengine_pcm: add snd_dmaengine_generic_pcm_open() drivers/dma/dmaengine.c | 2 +- drivers/dma/of-dma.c | 6 +++--- include/linux/dmaengine.h | 7 ++++--- include/linux/of_dma.h | 2 +- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-dmaengine-pcm.c | 39 +++++++++++++++++++++++++++++++++++++++ 6 files changed, 50 insertions(+), 8 deletions(-) -- 1.7.9.5

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Re: [alsa-devel] [PATCH 0/3] ALSA: snd-usb: Some small fixes to make Playback Design products work
by Jussi Laako 22 Mar '13

22 Mar '13

On 03/22/2013 03:27 AM, Andreas Koch wrote: > I wouldn't worry about DSD wide, because that is a dead format that I > was involved in launching for professional recording applications more > than 10 years ago. It never took off. I just need it for my own purposes, but I can make custom stuff as necessary if there's no interest from others. Right now I'm just hoping to see a well defined common sample formats defined for ALSA that can be utilized by drivers. DoP could be implemented inside the driver or at ALSA layer instead of applications doing it. Not that I would mind, but I would find it cleaner than the current approach... :) Currently, ASIO is the only API to support DSD sample formats. I understand that Microsoft or Apple is a slow to turn giant, but I'd hope ALSA wouldn't be such. ;) There are already those IEC958 and DD/dts things being used for movie playback, so this wouldn't be such a big difference afterall. - Jussi

2 2

[alsa-devel] [PATCH] SoC: dma-sh7760: Fix compile error
by Lars-Peter Clausen 22 Mar '13

22 Mar '13

The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Given that nobody noticed this over the course of the last three years the code is probably pretty dead by now anyway. --- sound/soc/sh/dma-sh7760.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8f..1a8b03e 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; -- 1.8.0

2 2

[alsa-devel] [PATCH 1/2] ASoC: compress - add support for metadata apis
by Vinod Koul 22 Mar '13

22 Mar '13

From: Jeeja KP <jeeja.kp(a)intel.com> Compress core added metadata apis in 9727b4, so add same in ASoC Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Mark, Takashi: How do you guys wnat to take this one, Ideally this should go to mark's tree but the dependent apis are in takashi's tree, either way is fine with me :) --- sound/soc/soc-compress.c | 27 +++++++++++++++++++++++++++ 1 files changed, 27 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3ea7956..4730789 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -302,11 +302,38 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, return 0; } +static int sst_compr_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->set_metadata(cstream, metadata); + + return ret; +} + +static int sst_compr_get_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + int ret = 0; + + if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) + ret = platform->driver->compr_ops->get_metadata(cstream, metadata); + + return ret; +} /* ASoC Compress operations */ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_metadata = sst_compr_set_metadata, + .get_metadata = sst_compr_get_metadata, .get_params = soc_compr_get_params, .trigger = soc_compr_trigger, .pointer = soc_compr_pointer, -- 1.7.0.4

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Re: [alsa-devel] 答复: 答复: how to combine mutil-channel in playback or capture?
by Daniel Mack 22 Mar '13

22 Mar '13

Hi, please don't top-post. On 22.03.2013 11:11, Zhang wei wrote: > But what is the mechanism to synchronize the data from different > channel,and maybe different channel pairs use different DMA buffer. You can try and see if syncing the individual, non-combinded PCM streams helps: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html#pcm_sync > So, I guess ,about the combine ,according to the alsa.conf,the alsa-lib > opened the 4 devices(in my case) and got every channel paris’s data in > turns in a monent. And then combine them together in order as one frame? It will feed the input to all pcm streams. I don't really know if it uses the sync mechanism on that, but I'd guess so. > Because all the CODEC have the same clock timing,so the different DMA > buffer stored the different channel data for the same moment. Alsa-lib > obtains the same size data from each buffer in turns in once.repeat it > again and again. > > Right? Yes, and so the streams won't drift. Why don't you just go ahead and try it? You should play back some wav file with a test pattern which you can recognize on a DSO or IIS analyzer on the bus. Then see if they are in sync. Daniel

1 0

Re: [alsa-devel] 答复: how to combine mutil-channel in playback or capture?
by Daniel Mack 22 Mar '13

22 Mar '13

On 22.03.2013 10:23, Zhang wei wrote: > Thank you. Yes,you're right. I need 4 different stereo Codecs at > least more than one on my board and i think that will be more funy > than 1 Codec.In this case ,the struct snd_soc_card will be > initialized as following: Struct snd_soc_card test_card= { > .name=”Test Card”, .dai_link=&test_dai_codecs, .num_links=4; } As a > result ,there would be 4 devices named pcmC0Dnp(n=0,1,2,3) under the > directory /dev/snd. I wounder how and who combines the multi-channel > audio data togother? Is it the plugin ? Yes, you can use the alsa userspace for that. See http://www.alsa-project.org/main/index.php/Asoundrc Daniel

1 0

[alsa-devel] how to combine mutil-channel in playback or capture?
by Zhang wei 22 Mar '13

22 Mar '13

I have already achieved the alsa-driver which can playback and capture function with two channel. My board based on ARM9 which have 3 ADC & 1 DAC in IIS bus and all of them shared the same BCLK and LRCK. The pins figure Like this: _________________ | BCLK |___________ | | | LRCK |___________ | | | ADCDAT0 |___________ | | | ADCDAT1 |___________ | | | ADCDAT2 |___________ | | | DACDATA |___________ | | |_______________| Every ADCDATn(n=0,1,2) has two channel so the question is how can I get multi-channel audio data such as 4 or 6 in capture funciton? BR

2 1

[alsa-devel] [PATCH] sound: max98090: Remove executable bit
by Joe Perches 21 Mar '13

21 Mar '13

Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches <joe(a)perches.com> --- include/sound/max98090.h | 0 sound/soc/codecs/max98090.c | 0 sound/soc/codecs/max98090.h | 0 3 files changed, 0 insertions(+), 0 deletions(-) diff --git a/include/sound/max98090.h b/include/sound/max98090.h old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h old mode 100755 new mode 100644

2 13

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

On 03/21/2013 12:46 PM, Martin Fuzzey wrote: > On 21/03/13 17:11, Timur Tabi wrote: >> On Thu, Mar 21, 2013 at 3:39 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: >> >>> With this patch and that DT example the frequency of clock 162 will be _set_ >>> to 20MHz >> Does that mean that it ignores the ' clocks' parameter? > No, in this case the clocks parameter must contain the phandle of a clock > whose rate is configurable. > >> If clock-frequency is omitted the binding is still correct (hence the >> optional) but the frequency of clock 162 would not be modified. >> >> In the documentation I wrote "If both a clock and clock-frequency are >> provided the clock's rate will be set. " maybe this is not clear enough? >> It's not clear what it will be set *to*. > It will be set to the frequency specified by the clock-frquency attribute. > > Would > "If both a clock and clock-frequency are provided clock must support > the set_rate operation and its frequency will be set to the value specified > by clock-frequency" be better? Yes, that's much better. > > > The possible configurations and their use cases are: > > 1) only 'clocks' specified > clock points to a clock specified in the DT which already has an > appropriate frequency and is > configured by some other means external to the sgtl5000 driver > (bootloader, board setup code, or just a fixed rate clock) > > 2) only 'clock-frequency' specified > The chip is assumed to be clocked by a signal having the given > frequency, which may even be generated by a clock unknown to linux. > This could actually be represented as a special case of 1) by defining a > fixed-rate clock in the DT. > > 3) Both 'clocks' and 'clock-frequency' specified > The chip is assumed to be clocked by a rate programmable clock defined > in the clock tree. > clk_set_rate() will be called for this clock to set its rate to that > specified by clock-frequency This should all be in the binding document. > Cases 1 and 2 exist in the current code, this patch adds support for case 3. > > Prior to commit 81e8e4926167ab32593bbb915b45a42024ca1020 "ASoC: fsl: add > sgtl5000 clock support for imx-sgtl5000" > only case 2 was supported > This explains why 2 is not implemented as a special case of 1) > plain 'clock-frequency' was supported before the driver learnt how to > get a clock from the DT And this should be in the patch description. -- Timur Tabi

1 0

Re: [alsa-devel] [PATCH 50/54 v2] ASoC: switch over to use snd_soc_register_component() on tegra20 ac97
by Stephen Warren 21 Mar '13

21 Mar '13

On 03/21/2013 11:19 AM, Lucas Stach wrote: > Am Donnerstag, den 21.03.2013, 11:03 -0600 schrieb Stephen Warren: >> On 03/21/2013 04:37 AM, Kuninori Morimoto wrote: >> >> Lucas, can you please test this to make sure that the change from >> snd_soc_register_dais() to snd_soc_register_dai() (the latter inside >> snd_soc_register_component()) coupled with the change to the >> tegra_wm9712.c machine driver yields the correct results. Thanks. > > Can someone please FWD me the whole patch? I'm not subscribed to > ALSA-devel and wasn't on CC for this patch. I'll do that off-list.

1 0

[alsa-devel] [PATCH] ALSA: hda - Introduce "Headset Mic" name
by David Henningsson 21 Mar '13

21 Mar '13

Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_auto_parser.h | 2 ++ sound/pci/hda/patch_sigmatel.c | 6 +++++- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a3ea76a..6b173b3 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -260,6 +260,22 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } } + /* Take first mic to be a headset mic pin */ + if (cond_flags & HDA_PINCFG_HEADSET_MIC) { + for (i = 0; i < cfg->num_inputs; i++) { + int attr; + unsigned int def_conf; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr <= INPUT_PIN_ATTR_DOCK) + continue; + cfg->inputs[i].is_headset_mic = 1; + break; + } + } + /* FIX-UP: * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. @@ -388,6 +404,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); */ static const char *hda_get_input_pin_label(struct hda_codec *codec, + const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) { unsigned int def_conf; @@ -400,6 +417,8 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec, switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: + if (item && item->is_headset_mic) + return "Headset Mic"; if (!check_location) return "Mic"; attr = snd_hda_get_input_pin_attr(def_conf); @@ -480,7 +499,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, has_multiple_pins = 1; if (has_multiple_pins && type == AUTO_PIN_MIC) has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + return hda_get_input_pin_label(codec, &cfg->inputs[input], + cfg->inputs[input].pin, has_multiple_pins); } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); @@ -649,7 +669,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } } if (!name) - name = hda_get_input_pin_label(codec, nid, true); + name = hda_get_input_pin_label(codec, NULL, nid, true); break; } if (!name) diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index f748071..98a72d0 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -36,6 +36,7 @@ enum { struct auto_pin_cfg_item { hda_nid_t pin; int type; + int is_headset_mic:1; }; struct auto_pin_cfg; @@ -80,6 +81,7 @@ struct auto_pin_cfg { /* bit-flags for snd_hda_parse_pin_def_config() behavior */ #define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ #define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Take first mic as headset mic */ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, struct auto_pin_cfg *cfg, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3be877b..1d9d642 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3528,8 +3528,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; int err; + int flags = 0; - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + if (spec->headset_jack) + flags |= HDA_PINCFG_HEADSET_MIC; + + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, flags); if (err < 0) return err; -- 1.7.9.5

2 7

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

On Thu, Mar 21, 2013 at 3:39 AM, Martin Fuzzey <mfuzzey(a)parkeon.com> wrote: > With this patch and that DT example the frequency of clock 162 will be _set_ > to 20MHz Does that mean that it ignores the ' clocks' parameter? > If clock-frequency is omitted the binding is still correct (hence the > optional) but the frequency of clock 162 would not be modified. > > In the documentation I wrote "If both a clock and clock-frequency are > provided the clock's rate will be set. " maybe this is not clear enough? It's not clear what it will be set *to*.

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: hda - Introduce "Headset Mic" name
by David Henningsson 21 Mar '13

21 Mar '13

Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_auto_parser.h | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index a3ea76a..6b173b3 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -260,6 +260,22 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, } } + /* Take first mic to be a headset mic pin */ + if (cond_flags & HDA_PINCFG_HEADSET_MIC) { + for (i = 0; i < cfg->num_inputs; i++) { + int attr; + unsigned int def_conf; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr <= INPUT_PIN_ATTR_DOCK) + continue; + cfg->inputs[i].is_headset_mic = 1; + break; + } + } + /* FIX-UP: * If no line-out is defined but multiple HPs are found, * some of them might be the real line-outs. @@ -388,6 +404,7 @@ EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); */ static const char *hda_get_input_pin_label(struct hda_codec *codec, + const struct auto_pin_cfg_item *item, hda_nid_t pin, bool check_location) { unsigned int def_conf; @@ -400,6 +417,8 @@ static const char *hda_get_input_pin_label(struct hda_codec *codec, switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: + if (item && item->is_headset_mic) + return "Headset Mic"; if (!check_location) return "Mic"; attr = snd_hda_get_input_pin_attr(def_conf); @@ -480,7 +499,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, has_multiple_pins = 1; if (has_multiple_pins && type == AUTO_PIN_MIC) has_multiple_pins &= check_mic_location_need(codec, cfg, input); - return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + return hda_get_input_pin_label(codec, &cfg->inputs[input], + cfg->inputs[input].pin, has_multiple_pins); } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); @@ -649,7 +669,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, } } if (!name) - name = hda_get_input_pin_label(codec, nid, true); + name = hda_get_input_pin_label(codec, NULL, nid, true); break; } if (!name) diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h index f748071..c7826ce 100644 --- a/sound/pci/hda/hda_auto_parser.h +++ b/sound/pci/hda/hda_auto_parser.h @@ -36,6 +36,7 @@ enum { struct auto_pin_cfg_item { hda_nid_t pin; int type; + unsigned int is_headset_mic:1; }; struct auto_pin_cfg; @@ -80,6 +81,7 @@ struct auto_pin_cfg { /* bit-flags for snd_hda_parse_pin_def_config() behavior */ #define HDA_PINCFG_NO_HP_FIXUP (1 << 0) /* no HP-split */ #define HDA_PINCFG_NO_LO_FIXUP (1 << 1) /* don't take other outs as LO */ +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Take first mic as headset mic */ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, struct auto_pin_cfg *cfg, -- 1.7.9.5

1 1

Re: [alsa-devel] [PATCH] Sound: sgtl5000 Allow codec clock frequency to be set.
by Timur Tabi 21 Mar '13

21 Mar '13

Martin Fuzzey wrote: > +If a clock is provided, clock-frequency is optional > + > +If no clock is provided clock-frequency is required (this represents the codec > +being clocked by an external signal not present in the clock tree) > + > +If both a clock and clock-frequency are provided the clock's rate will be set. > + > + > Example: > > codec: sgtl5000@0a { > compatible = "fsl,sgtl5000"; > reg = <0x0a>; > + clock-frequency = <20000000>; > + clocks = <&clks 162>; /* cko1 */ The code looks okay, but I'm not sure this example is right. The binding says that clock-frequency is optional if 'clocks' is provided, but here you list both. You should probably have two examples. -- Timur Tabi

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: wm8903: Add the DAC boost control
by Mark Brown 20 Mar '13

20 Mar '13

On Wed, Mar 20, 2013 at 05:37:32PM +0100, Alban Bedel wrote: > Signed-off-by: Alban Bedel <alban.bedel(a)avionic-design.de> Applied, thanks.

1 0

[alsa-devel] [PATCH] salsa-lib: Rename _sndrv_pcm_format to _snd_pcm_format
by Mihail Zenkov 20 Mar '13

20 Mar '13

Rename _sndrv_pcm_format to _snd_pcm_format for compatible with alsa-lib. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/asound.h +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/asound.h @@ -133,7 +133,7 @@ SND_PCM_ACCESS_LAST = SND_PCM_ACCESS_RW_NONINTERLEAVED, } snd_pcm_access_t; -typedef enum _sndrv_pcm_format { +typedef enum _snd_pcm_format { SND_PCM_FORMAT_UNKNOWN = -1, SND_PCM_FORMAT_S8 = 0, /* signed */ SND_PCM_FORMAT_U8, /* unsigned */

2 1

[alsa-devel] [PATCH] salsa-lib: Set SNDRV_PCM_HW_PARAMS_NORESAMPLE by default
by Mihail Zenkov 20 Mar '13

20 Mar '13

As we don't allow resampling set SNDRV_PCM_HW_PARAMS_NORESAMPLE by default to respect snd_pcm_hw_rule_noresample in drivers. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/pcm_params.c +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/pcm_params.c @@ -436,6 +436,7 @@ params->rmask = ~0U; params->cmask = 0; params->info = ~0U; + params->flags = SNDRV_PCM_HW_PARAMS_NORESAMPLE; } int snd_pcm_hw_params_any(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)

2 1

[alsa-devel] [PATCH] salsa-lib: Fix mistakes after refactoring pcm_params.c
by Mihail Zenkov 20 Mar '13

20 Mar '13

Fix some regressions after refactoring pcm_params.c These mistakes don't allow snd_pcm_hw_param_set_near to establish correct value. --- salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884/src/pcm_params.c +++ salsa-lib-a55ce0c348cd16e7853dae11e8e0e1b5a101d884-1/src/pcm_params.c @@ -590,11 +590,9 @@ save = *params; err = hw_param_set_max(params, var, *val, dir ? *dir : 0); - if (!err) { - err = hw_param_update_var(pcm, params, var, err); - if (!err) - return _snd_pcm_hw_param_get_max(params, var, val, dir); - } + err = hw_param_update_var(pcm, params, var, err); + if (!err) + return _snd_pcm_hw_param_get_max(params, var, val, dir); *params = save; return err; } @@ -683,7 +681,7 @@ snd_pcm_hw_params_t save = *params; int err = hw_param_set(params, var, val, dir); err = hw_param_update_var(pcm, params, var, err); - if (err) + if (err < 0) *params = save; return err; } @@ -712,7 +710,7 @@ save = *params; i->integer = 1; err = hw_param_update_var(pcm, params, var, 1); - if (err) + if (err < 0) *params = save; return err; } @@ -827,7 +825,7 @@ else if (valdir == 0) maxdir = -1; else { - valdir = 1; + maxdir = 1; max--; } save = *params;

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Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. ...but I applied this anyway since it's clearly an improvement and a small leak is better than a crash.

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Re: [alsa-devel] [PATCH] sound: soc: fix invalid free of devm_ allocated data
by Mark Brown 20 Mar '13

20 Mar '13

On Sat, Mar 16, 2013 at 01:45:34PM +0200, Silviu-Mihai Popescu wrote: > The objects allocated by devm_* APIs are managed by devres and are freed when > the device is detached. Hence there is no need to use kfree() explicitly. In this case since we're in error handling paths which might not be fatal (though I suspect they probably should be normally) it looks like the better fix is to use devm_kfree() to free immediately. Otherwise the unused memory will end up hanging around for as long as the device is bound.

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[alsa-devel] [PATCH 1/2] ASoC: spear_pcm: Update to new pcm_new() API
by Lars-Peter Clausen 20 Mar '13

20 Mar '13

Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Btw. where are the Kconfig and Makefile files for spear? --- sound/soc/spear/spear_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5..5e7aebe 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) -- 1.8.0

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[alsa-devel] [PATCH 0/3] ASoC: omap-mcpdm: Fix for full duplex audio
by Peter Ujfalusi 20 Mar '13

20 Mar '13

Hi, We had this issue for a long time but I had no time to think and fix it. With this series the full duplex audio will work correctly with McPDM. Under the hood we have HW limitation which prevents us to start up/downlink separately. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcpdm: Collect link direction configuration under a struct ASoC: omap-mcpdm: Fix for full duplex audio use case ASoC: omap-mcpdm: Remove leftower define for IO address sound/soc/omap/omap-mcpdm.c | 66 ++++++++++++++++++++++++++++++++------------- 1 file changed, 48 insertions(+), 18 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: fsl: imx-pcm-fiq: Use 'unsigned int' for period
by Fabio Estevam 20 Mar '13

20 Mar '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Fix the following warning when building with W=1 option: sound/soc/fsl/imx-pcm-fiq.c: In function 'snd_hrtimer_callback': sound/soc/fsl/imx-pcm-fiq.c:76:12: warning: comparison between signed and unsigned integer expressions [-Wsign-compare] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 920f945..47228c0 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -34,7 +34,7 @@ #include "imx-ssi.h" struct imx_pcm_runtime_data { - int period; + unsigned int period; int periods; unsigned long offset; unsigned long last_offset; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: snd-usb: mixer: propagate errors up the call chain
by Daniel Mack 20 Mar '13

20 Mar '13

In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f7..db30d02 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -1356,8 +1357,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ -- 1.8.1.4

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