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

[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixup for Acer AO725 laptop
by Takashi Iwai 21 Jan '13

21 Jan '13

Acer AO725 needs the same fixup as AO756. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6369b23..cf38861 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6266,6 +6266,7 @@ 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, 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), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), -- 1.8.1.1

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[alsa-devel] [PATCH 2/2] ALSA: hda - Add Conexant CX20755/20756/20757 codec IDs
by Takashi Iwai 21 Jan '13

21 Jan '13

These are just compatible with other CX2075x codecs. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_conexant.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd798c3..009b77a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4636,6 +4636,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15111, .name = "CX20753/4", .patch = patch_conexant_auto }, + { .id = 0x14f15113, .name = "CX20755", + .patch = patch_conexant_auto }, + { .id = 0x14f15114, .name = "CX20756", + .patch = patch_conexant_auto }, + { .id = 0x14f15115, .name = "CX20757", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -4659,6 +4665,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150b9"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); +MODULE_ALIAS("snd-hda-codec-id:14f15113"); +MODULE_ALIAS("snd-hda-codec-id:14f15114"); +MODULE_ALIAS("snd-hda-codec-id:14f15115"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.8.1.1

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[alsa-devel] Invitation to connect on LinkedIn
by Putinei Ionut via LinkedIn 21 Jan '13

21 Jan '13

LinkedIn ------------ Putinei Ionut requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, Find out why I use LinkedIn. Stay in touch and build your professional network. - Putinei Accept invitation from Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/rso/227244299/bCFj/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

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[alsa-devel] [PATCH] ASoC: arizona: Support clearing clocks
by Mark Brown 21 Jan '13

21 Jan '13

Some systems may wish to support switching between telephony and CD audio clock rates but this is restricted by enforcement of constraints on the current DAI clock. Support setting clocks to zero and don't enforce any constraints in that case in order to facilitate this use case. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4c2b626..8c56488 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -585,6 +585,10 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 147456000: val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; break; + case 0: + dev_dbg(arizona->dev, "%s cleared\n", name); + *clk = freq; + return 0; default: return -EINVAL; } @@ -803,6 +807,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, return 0; } + if (base_rate == 0) + return 0; + if (base_rate % 8000) constraint = &arizona_44k1_constraint; else -- 1.7.10.4

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[alsa-devel] question on snd_soc_dai_link_event
by Sebastien LEDUC 21 Jan '13

21 Jan '13

Hi, I have one question about current implementation of function snd_soc_dai_link_event Currently, on power up, it calls hw_params on the source and sink dai drivers, and digital_mute(0) on the sink dai driver I wonder why it does not call the full hw_params / prepare / trigger sequence on both dai drivers to start the 2 DAIs. Thanks in advance Regards, Sebastien

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[alsa-devel] Focusrite Scarlett 18i6 mixer
by Robin Gareus 20 Jan '13

20 Jan '13

Hi, I've started to reverse engineer the 18i6 mixer protocol and came up with a proof-of-concept test script which includes documentation: https://github.com/x42/scarlettmixer Except for a few fixed controls and switches, the 18i6's mixer and router is [likely] too complex to be exposed via ALSA's mixer interface. AFAICT it'll require a dedicated control application - like hdspmixer - in order to make full use of it. The current proof-of-concept uses pyUSB to send URB control messages and therefore requires root access. It will also not work while the device is in use (by snd-usb-audio). A dedicated ioctl interface - similar to SNDRV_HDSP_ - should be added to ALSA. However, it is yet unclear to me how much of the protocol should be moved to kernel-space. Anyway, the script can be used to configure the device, save settings to it and subsequently use those settings with snd-usb-audio and Martin's patch. Thoughts, comments? robin PS. the same protocol may apply to other Focusrite Scalett USB2 devices such as the 8i6 - some of the factory-reset traces hint at that, but I would not know.

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[alsa-devel] Must MIDI SysEx data contain the start byte 0xf0 and the end byte 0xf7?
by Joerg-Cyril.Hoehle＠t-systems.com 20 Jan '13

20 Jan '13

Hi, I've read about SysEx continuations aka. divided SysEx http://www.allegro.cc/forums/thread/607445 http://www.sonicspot.com/guide/midifiles.html What it means is that within a MIDI file, a long sysex could be split across several events with different delta times, giving the receiver some processing time in between (e.g. flash EEPROM etc.). The delta times tell when to send the parts. Now, is that truth? How to do that with ALSA? The documentation on snd_seq_ev_set_sysex insists: "the sysex data must contain the start byte 0xf0 and the end byte 0xf7." http://www.alsa-project.org/alsa-doc/alsa-lib/group___seq_middle.html#ga104… Would it be the sole business of the low-level ALSA driver to slowly send large (and complete) sysex data, out of application control (e.g. the delta times would get lost)? Should apps always coalesce partial sysex data and hand out the complete blob to ALSA's snd_seq_ev_set_sysex? Is divided SysEx too rare to worry about? I'm asking because we've seen a user of Wine & KORG stumble upon a non-F7-terminated SysEx. Regards, Jörg Höhle

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[alsa-devel] [PATCH] ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory
by Mark Brown 20 Jan '13

20 Jan '13

With some buses the transfers may DMAed, especially for larger blocks. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 625e973..3e4e40e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3131,9 +3131,12 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (!codec->using_regmap) return -EINVAL; - data = ucontrol->value.bytes.data; len = params->num_regs * codec->val_bytes; + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a @@ -3146,10 +3149,6 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, val &= params->mask; - data = kmemdup(data, len, GFP_KERNEL); - if (!data) - return -ENOMEM; - switch (codec->val_bytes) { case 1: ((u8 *)data)[0] &= ~params->mask; @@ -3171,8 +3170,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(codec->control_data, params->base, data, len); - if (params->mask) - kfree(data); + kfree(data); return ret; } -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: wm2200: Implement EQ and LHPF coefficient configuration
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 732b3dd..1be61e6 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1141,6 +1141,12 @@ SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SND_SOC_BYTES_MASK("EQL Coefficients", WM2200_EQL_1, 20, WM2200_EQL_ENA), +SND_SOC_BYTES_MASK("EQR Coefficients", WM2200_EQR_1, 20, WM2200_EQR_ENA), + +SND_SOC_BYTES("LHPF1 Coefficeints", WM2200_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficeints", WM2200_HPLPF2_2, 1), + SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, WM2200_OUT1_OSR_SHIFT, 1, 0), SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L, -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: wm_adsp: Make region identification errors more informative
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f956c35..267a6f1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -773,7 +773,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) break; default: - adsp_err(dsp, "Unknown region type %x\n", type); + adsp_err(dsp, "%s.%d: Unknown region type %x at %d\n", + file, blocks, type, pos); break; } -- 1.7.10.4

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