December 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Device Tree simple-audio-card without codec HOWTO ?
by Aurelien Bouin 18 Dec '14

18 Dec '14

Hello, I use the iMX6 ESAI bus to send audio datas to other processors ... but actually I don't have any codec behind ... I am wondering if there is any solution with the devicetree to implement such configuration with "simple-audio-card" That will configure tdm slots and width. I tried without success that : sound { compatible = "simple-audio-card"; simple-audio-card,name = "Generic-ESAI-Sound-Card"; simple-audio-card,format = "left_j"; simple-audio-card,cpu { sound-dai = <&esai 0>; dai-tdm-slot-num = <32>; dai-tdm-slot-width = <8>; }; simple-audio-card,codec { sound-dai = <0 0>; }; }; Because for the moment I have not find another solution than doing a module : http://pastebin.com/cnGwnEjn Thank you for your help, Regards, Aurelien BOUIN

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[alsa-devel] [PATCH] ALSA: pcm: Don't leave PREPARED state after draining
by Takashi Iwai 18 Dec '14

18 Dec '14

When a PCM draining is performed to an empty stream that has been already in PREPARED state, the current code just ignores and leaves as it is, although the drain is supposed to set all such streams to SETUP state. This patch covers that overlooked case. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d9572ad2b..64d9863d6565 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1546,6 +1546,8 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) if (! snd_pcm_playback_empty(substream)) { snd_pcm_do_start(substream, SNDRV_PCM_STATE_DRAINING); snd_pcm_post_start(substream, SNDRV_PCM_STATE_DRAINING); + } else { + runtime->status->state = SNDRV_PCM_STATE_SETUP; } break; case SNDRV_PCM_STATE_RUNNING: -- 2.2.0

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[alsa-devel] [PATCH 1/2] ALSA: hda_controller: Separate stream_tag for input and output streams.
by libin.yang＠intel.com 18 Dec '14

18 Dec '14

From: rfredzim <rafal.f.redzimski(a)intel.com> Implemented separate stream_tag assignment for input and output streams. According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. This change is necessary to support HW which provides a total of more than 15 stream DMA engines which with legacy implementation causes an overflow on SDxCTL.STRM field (and the whole SDxCTL register) and as a result usage of Reserved value 0 in the SDxCTL.STRM field which confuses HDA controller. Signed-off-by: rfredzim <rafal.f.redzimski(a)intel.com> Signed-off-by: Jayachandran B <jayachandran.b(a)intel.com> Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 24 ++++++++++++++++++++++-- sound/pci/hda/hda_priv.h | 1 + 2 files changed, 23 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..dcca1ef 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1922,10 +1922,18 @@ int azx_mixer_create(struct azx *chip) EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return (index >= chip->capture_index_offset && + index < chip->capture_index_offset + chip->capture_streams); +} + /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1946,21 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout + * the stream direction group, + * valid values 1...15 + * use separate stream tag if the flag + * AZX_DCAPS_SEPARATE_STREAM_TAG is used + */ + if (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) + azx_dev->stream_tag = + is_input_stream(chip, i) ? + ++in_stream_tag : + ++out_stream_tag; + else + azx_dev->stream_tag = i + 1; } return 0; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index aa484fd..166e3e8 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -171,6 +171,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ enum { AZX_SNOOP_TYPE_NONE , -- 1.9.1

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[alsa-devel] [PATCH 1/3] ALSA: asihpi: relax firmware version check
by eliot＠blennerhassett.gen.nz 18 Dec '14

18 Dec '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Some products firmware is no longer being updated e.g. dsp5000, dsp8700 but it should continue to work with updated HPI versions. Avoid regression by allowing this firmware to be loaded as long as major version is the same. Warn about mismatching versions, as matching versions are preferred. Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> --- sound/pci/asihpi/hpidspcd.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c index ac91637..3603c24 100644 --- a/sound/pci/asihpi/hpidspcd.c +++ b/sound/pci/asihpi/hpidspcd.c @@ -1,8 +1,9 @@ -/***********************************************************************/ -/** +/*********************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support(a)audioscience.com> + Functions for reading DSP code using hotplug firmware loader + + Copyright (C) 1997-2014 AudioScience Inc. <support(a)audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -17,11 +18,7 @@ along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -\file -Functions for reading DSP code using -hotplug firmware loader from individual dsp code files -*/ -/***********************************************************************/ +***********************************************************************/ #define SOURCEFILE_NAME "hpidspcd.c" #include "hpidspcd.h" #include "hpidebug.h" @@ -68,17 +65,18 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, goto error2; } - if ((header.version >> 9) != (HPI_VER >> 9)) { - /* Consider even and subsequent odd minor versions to be compatible */ - dev_err(&dev->dev, "Incompatible firmware version DSP image %X != Driver %X\n", + if (HPI_VER_MAJOR(header.version) != HPI_VER_MAJOR(HPI_VER)) { + /* Major version change probably means Host-DSP protocol change */ + dev_err(&dev->dev, + "Incompatible firmware version DSP image %X != Driver %X\n", header.version, HPI_VER); goto error2; } if (header.version != HPI_VER) { - dev_info(&dev->dev, - "Firmware: release version mismatch DSP image %X != Driver %X\n", - header.version, HPI_VER); + dev_warn(&dev->dev, + "Firmware version mismatch: DSP image %X != Driver %X\n", + header.version, HPI_VER); } HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
by Chris J Arges 18 Dec '14

18 Dec '14

The num_controls field was incorrectly set to 0 causing 6i6 to not be initialized. Set this to 9. Reported-and-tested-by: Mark Roberts <sunifiram(a)gmail.com> Signed-off-by: Chris J Arges <chris.j.arges(a)canonical.com> --- sound/usb/mixer_scarlett.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 9109652..7438e7c 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -655,7 +655,7 @@ static struct scarlett_device_info s6i6_info = { .names = NULL }, - .num_controls = 0, + .num_controls = 9, .controls = { { .num = 0, .type = SCARLETT_OUTPUTS, .name = "Monitor" }, { .num = 1, .type = SCARLETT_OUTPUTS, .name = "Headphone" }, -- 1.9.1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Create capture source ctls when stereo mix input is added
by Takashi Iwai 18 Dec '14

18 Dec '14

When the stereo mix input is explicitly enabled via a user hint, the driver should create always a capture source enum ctl and disable the auto-mic switch. Otherwise the behavior gets confused. For doing it, this patch just sets spec->suppress_auto_mic flag appropriately. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 63b69f750d8e..293af6b5ba3d 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3224,6 +3224,8 @@ static int create_input_ctls(struct hda_codec *codec) "Stereo Mix", 0); if (err < 0) return err; + else + spec->suppress_auto_mic = 1; } return 0; -- 2.2.0

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[alsa-devel] [PATCH] ALSA: hda - Add quirk for Packard Bell EasyNote MX65
by Takashi Iwai 17 Dec '14

17 Dec '14

Packard Bell EasyNote MX65 with AD1986A codec needs a few fixups, namely, the pin config overrides to set only the known I/O pins and the EAPD has to be turned on. In addition, add stereo mix input forcibly for avoiding the weird KDE behavior by this update. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88251 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index eb54da757407..a9d78e275138 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -257,6 +257,18 @@ static void ad1986a_fixup_eapd(struct hda_codec *codec, } } +/* enable stereo-mix input for avoiding regression on KDE (bko#88251) */ +static void ad1986a_fixup_eapd_mix_in(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + ad1986a_fixup_eapd(codec, fix, action); + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_ENABLE; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -265,6 +277,8 @@ enum { AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, AD1986A_FIXUP_EAPD, + AD1986A_FIXUP_EAPD_MIX_IN, + AD1986A_FIXUP_EASYNOTE, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -329,6 +343,30 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad1986a_fixup_eapd, }, + [AD1986A_FIXUP_EAPD_MIX_IN] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd_mix_in, + }, + [AD1986A_FIXUP_EASYNOTE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0421402f }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x90a70130 }, /* int mic */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x04a19040 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x21, 0x411111f0 }, /* N/A */ + { 0x22, 0x411111f0 }, /* N/A */ + { 0x23, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + { 0x25, 0x411111f0 }, /* N/A */ + {} + }, + .chained = true, + .chain_id = AD1986A_FIXUP_EAPD_MIX_IN, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -342,6 +380,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), + SND_PCI_QUIRK(0x1631, 0xc022, "PackardBell EasyNote MX65", AD1986A_FIXUP_EASYNOTE), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), -- 2.2.0

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[alsa-devel] [PATCH] usb: add native DSD support for Matrix Audio DACs
by Jurgen Kramer 17 Dec '14

17 Dec '14

This patch adds native DSD support for two XMOS based DACs from Matrix Audio: - X-Sabre - Mini-i Pro Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4dbfb3d..a739841 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1245,8 +1245,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - /* iFi Audio micro/nano iDSD */ - case USB_ID(0x20b1, 0x3008): + case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ + case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ + case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 1.9.3

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[alsa-devel] Again: Alsa Driver Scarlett
by Orm Finnendahl 17 Dec '14

17 Dec '14

Hi, this is relating to my post from September: There doesn't seem to be any progress concerning the integration of the Scarlett alsa-driver patch into Alsa. Can someone on the list tell me how to invoke a process to get it integrated? The patch sources are here: https://github.com/smilingthax/alsa-driver_scarlett At the moment most of our students regularly have to reapply the patch as soon as they upgrade their systems and it's pretty time consuming and painful for the studio technicians and me to guide them through the process of patching kernel modules, so it would be great to simplify this process by integrating it into the official alsa distribution. I can confirm that the patch still works well on our systems. -- Orm ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- Forwarded message from Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> ----- Subject: Re: [alsa-devel] Alsa Driver Scarlett From: Orm Finnendahl <orm.finnendahl(a)selma.hfmdk-frankfurt.de> To: alsa-devel(a)alsa-project.org Date: Tue, 16 Sep 2014 15:49:25 +0200 Am Dienstag, den 16. September 2014 um 15:22:27 Uhr (+0200) schrieb Johannes Kroll: > > I guess I'll need to find the time to port the patch to a recent kernel > when I upgrade, unless someone else does it first. The patch still works on our recent kernels (we are on debian testing with a 3.14-2 kernel), atm there is no porting necessary. I can also confirm that the driver has no known issues on our systems (we use a couple of Scarlett 18i20 in different configurations). It would be great, though if the patch could get integrated into alsa. At the moment it is quite painful for our students with linux laptops as they have to reapply the patch on every upgrade. -- Orm Finnendahl ------------------------------------------------------------------ Prof. Orm Finnendahl Komposition Hochschule für Musik und Darstellende Kunst Eschersheimer Landstr. 29-39 60322 Frankfurt am Main ----- End forwarded message -----

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Re: [alsa-devel] New codec support for ALC256
by Takashi Iwai 17 Dec '14

17 Dec '14

At Wed, 17 Dec 2014 07:56:56 +0000, Kailang wrote: > > Attach the patch. Don't we need the addition to alc_fill_eapd_coef()? thanks, Takashi

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