April 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Bard 22 Apr '13

22 Apr '13

Dear Mark, Please see my reply below. Thanks. -----Original Message----- From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] Sent: Tuesday, April 16, 2013 10:38 PM To: Bard Cc: Flove; Oder Chiou; swarren(a)nvidia.com; bard Subject: Re: [PATCH] ASoC: add RT5640 CODEC driver On Tue, Apr 16, 2013 at 10:10:37PM +0800, bardliao(a)realtek.com wrote: Please remember to post things to the list... just a quick scan through here: > +static int rt5640_reg_init(struct snd_soc_codec *codec) { > + int i; > + for (i = 0; i < RT5640_INIT_REG_LEN; i++) > + snd_soc_write(codec, init_list[i].reg, init_list[i].val); > + return 0; > +} This should be converted over to regmap. There's support for paging in the core, look at regmap_range. Bard: I will use regmap_register_patch to do that. > +/* IN1/IN2 Input Type */ > +static const char * const rt5640_input_mode[] = { > + "Single ended", "Differential"}; > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in1_mode_enum, RT5640_IN1_IN2, > + RT5640_IN_SFT1, rt5640_input_mode); > + > +static const SOC_ENUM_SINGLE_DECL( > + rt5640_in2_mode_enum, RT5640_IN3_IN4, > + RT5640_IN_SFT2, rt5640_input_mode); Platform data... Bard: Can I export a function that machine can configure it by this function? > +static int rt5640_vol_rescale_put(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, val2; > + > + val = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[0]; > + val2 = VOL_RESCALE_MAX_VOL - ucontrol->value.integer.value[1]; > + return snd_soc_update_bits_locked(codec, mc->reg, RT5640_L_VOL_MASK | > + RT5640_R_VOL_MASK, val << mc->shift | val2); } This looks like a variant on the _RANGE controls? Bard: Yes, we want to limit the max value of volume. If it is not good, I will change it. > +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; > +} Remove this, it doesn't actually do anything any more. Bard: I will remove it. ------Please consider the environment before printing this e-mail.

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[alsa-devel] I connected my digital piano and it just worked :-)
by David Henningsson 22 Apr '13

22 Apr '13

On a slightly different note; I needed to connect my digital piano to my laptop the other day, but the laptop had no MIDI interface. So I tried connecting it through USB, because the piano had a USB port too. I knew it required a 3rd party driver on Windows, so I didn't expect much. It just worked. :-) From someone who knows that fixing hardware often takes a lot of effort and hard work, I just wanted to shout a "thank you" and a "keep up the good work" to all contributors! :-) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] ALSA: hda - Limit internal mic boost for a few Asus machines
by David Henningsson 22 Apr '13

22 Apr '13

These are being reported as being so noisy at high mic boost levels, so they are unusable in practice. Therefore artificially limit the boosts. BugLink: https://bugs.launchpad.net/bugs/1089795 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 28b3f5a..6e5d383 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3156,6 +3156,38 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + int i; + + /* The mic boosts on level 2 and 3 are too noisy + on the internal mic input. + Therefore limit the boost to 0 or 1. */ + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + unsigned int defcfg; + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) != INPUT_PIN_ATTR_INT) + continue; + + snd_hda_override_amp_caps(codec, nid, HDA_INPUT, + (0x00 << AC_AMPCAP_OFFSET_SHIFT) | + (0x01 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x2f << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3190,6 +3222,7 @@ enum { ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, ALC269_FIXUP_ACER_AC700, + ALC269_FIXUP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -3428,6 +3461,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_DMIC, }, + [ALC269_FIXUP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3458,11 +3495,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), -- 1.7.9.5

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[alsa-devel] [PATCH 0/7] Add capture support to compress API
by Richard Fitzgerald 22 Apr '13

22 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 Thess patches apply to: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git branch "for-next" based on top of commit 126825e ALSA: snd-usb: add quirks handler for DSD streams - RichF 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

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[alsa-devel] [PATCH v3] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 22 Apr '13

22 Apr '13

This is an other version of the patch send via git send-email. I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Commit message: Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- sound/usb/helper.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..6209024 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -86,14 +86,22 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + + if (requesttype & USB_DIR_IN) + timeout = USB_CTRL_GET_TIMEOUT; + else + timeout = USB_CTRL_SET_TIMEOUT; + err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); + if (size > 0) { memcpy(data, buf, size); kfree(buf); -- 1.7.9.5

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[alsa-devel] [patch] ALSA: compress: info leak in snd_compr_get_caps()
by Dan Carpenter 22 Apr '13

22 Apr '13

If the ->get_caps() function doesn't clear the buffer then there would stack information leaked to userspace. For example, soc_compr_get_caps() can return success without clearing the buffer. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Perhaps the soc_compr_get_caps() function should return an error code if the platform->driver->compr_ops is NULL. I'm not sure about that, and it's a separate issue anyway. diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..8d3190a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -375,6 +375,7 @@ snd_compr_get_caps(struct snd_compr_stream *stream, unsigned long arg) if (!stream->ops->get_caps) return -ENXIO; + memset(&caps, 0, sizeof(caps)); retval = stream->ops->get_caps(stream, &caps); if (retval) goto out;

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Re: [alsa-devel] [PATCH v2] Add M2Tech hiFace USB-SPDIF driver
by Pavel Hofman 22 Apr '13

22 Apr '13

On 22.4.2013 10:31, Pietro Cipriano wrote: > Il 22/04/2013 09:40, Pavel Hofman ha scritto: >> >> >> On 20.4.2013 22:15, Daniel Mack wrote: >>> On 13.02.2013 18:11, Antonio Ospite wrote: >>>> Add driver for M2Tech hiFace USB-SPDIF interface and compatible >>>> devices. >>>> >>>> M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output >>>> Interface, see http://www.m2tech.biz/hiface.html >>> >>> Just curious: is there a new version of this patch set coming? >> >> I have read users report large latencies (unable to watch movies) of >> hiFace1 in OSX while reportedly they have not identified any major >> latency in windows. Perhaps the fixed PCM_PACKET_SIZE > The issue is related to a no controllable buffer dimension, which > depends by Audio Core thus present only on OSx. A easy workaround is of > to set up a higher output sampling frequency, for instance at 192KHz the > delay is no more perceptible, obviously the DAC must be compliant with > this setting. > > Pietro Thanks for very fast reponse. So is it possible to reduce the period size in the linux driver? Right now if I calculate correctly it is 4096 bytes / 32bits / 2 channels / 48000 Hz = 10 ms. Thanks, Pavel.

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Re: [alsa-devel] [PATCH 26/28] proc: Supply a function to remove a proc entry by PDE [RFC]
by Rafael J. Wysocki 22 Apr '13

22 Apr '13

On Tuesday, April 16, 2013 07:27:34 PM David Howells 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 The ACPI changes look good to me. Acked-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> > --- > > 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 > } > -- I speak only for myself. Rafael J. Wysocki, Intel Open Source Technology Center.

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[alsa-devel] [PATCH v2] ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
by Daniel Schürmann 20 Apr '13

20 Apr '13

Thank you Daniel for review. Here is an updated version of the patch against https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/ 581cbef46ae60073252208fc1f26dd1044f6e215 Set the timeout for USB control set messages according to the USB 2 spec §9.2.6.4 to 5000 ms. To avoid new issues, the get timeout is unchanged at 1000 ms, though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout > 1240 ms Signed-off-by: Daniel Schürmann <daschuer(a)mixxx.org> --- diff --git a/sound/usb/helper.c b/sound/usb/helper.c index c1db28f..c1caae4 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -23,6 +23,11 @@ #include "helper.h" #include "quirks.h" +/* Value from 9.2.6.4 USB 2 spec */ +#define USB_MSG_SET_TIMEOUT 5000 +/* Value from spec is 500 but we pick 1000 for legacy reasons */ +#define USB_MSG_GET_TIMEOUT 1000 + /* * combine bytes and get an integer value */ @@ -86,14 +91,19 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, { int err; void *buf = NULL; + int timeout; if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) return -ENOMEM; } + if (requesttype & USB_DIR_IN) + timeout = USB_MSG_GET_TIMEOUT; + else + timeout = USB_MSG_SET_TIMEOUT; err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, 1000); + value, index, buf, size, timeout); if (size > 0) { memcpy(data, buf, size); kfree(buf);

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[alsa-devel] DAPM widget power off
by yuvaraj r 20 Apr '13

20 Apr '13

Hi All, I have two different dais 1. Hifi Multimedia 2. Voice call dai Before opening voice call dai. I closed Hifi stream. After 5 sec Voice call stream dapm widget got powered off automatically. I am assuming that stream close event of Multimedia causing problem. What will issue in my cause. Thanks Yuvaraj

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