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Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo
by S. Aguinaga 23 Jan '10

23 Jan '10

Hello Giuliano, No luck, same behavior. I edited the first file listed listed below and recompiled it and installed it. Then I restarted the system with the indigo card in place, and the VU meters show levels in the -120 to -100 dB? I do have alsa-driver-1.0.22 my kernel is 2.6.31 // Salvador ________________________________ From: Giuliano Pochini <pochini(a)shiny.it> To: S. Aguinaga <sa_jr(a)yahoo.com> Sent: Sun, January 17, 2010 4:04:38 PM Subject: Re: [alsa-devel] Echo Indigo card on Sabayon Gentoo On Thu, 14 Jan 2010 12:13:55 -0800 (PST) "S. Aguinaga" <sa_jr(a)yahoo.com> wrote: > Hello Fellows, > Just downloaded the latest (released) alsa-lib, alsa-driver, and alsa-utils, but my Indigo card does not work. It has been tested on the same system using the same drivers and on the same system using Fedora 10, that works fine. First install a recent version of echomixer. If you are using alsa-driver 1.0.22 or kernel 2.6.32, try to add this line: #define ECHOCARD_HAS_LINE_OUT_GAIN in alsa-driver-1.0.22/alsa-kernel/pci/echoaudio/indigoio.c or in linux-2.6.32/sound/pci/echoaudio/indigoio.c and recompile. It re-enables the line-out control that I removed last year because it was not supposed to exist in that card. -- Giuliano.

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[alsa-devel] [PATCH] modem.conf Off-hook improve behavior
by David Fries 23 Jan '10

23 Jan '10

From: David Fries <david(a)fries.net> Only restore the old value if it differs from the requested value, because if it has changed restoring the old value overrides the change. Take for example, a voice modem with a .conf that sets preserve off-hook. Start playback (on-hook to off-hook), start record (off-hook to off-hook), stop playback (off-hook to restore on-hook), stop record (on-hook to restore off-hook), Clearly you don't want to leave the modem "on the phone" now that there isn't any playback or recording active. Signed-off-by: David Fries <david(a)fries.net> --- Comments welcome. The default modem.conf has, hooks.0 { type ctl_elems hook_args [ { name "Off-hook Switch" preserve true value "on" lock false optional true } ] } I wrote the patch to be, if you didn't modify it, don't restore the value. I also thought that it might work to check to see if the current mixer value matches the requested value (which is the value the program set on open), and only set the old value if it hasn't been externally modified. The theory being if something else changed the mixer value, we don't want to by default overwrite that change. Some additional flags could also be added to leave the current behavior alone and add the if you didn't modify it, don't restore it, or if it changed after you set it, don't restore it. Here are some flag names if that would be preferred, preserve_if_different restore_if_unmodified no_squash restore_overrides preserve_if_changed. Setting the lock isn't really an option, because then you can't manually "hang up" the modem if you needed to. On a side note, do any of the alsa voice/software modems support notifying userspace when the line is ringing? I'm using intel8x0m, not that I think it matters here. diff --git a/include/control.h b/include/control.h index 2361dc3..ded884a 100644 --- a/include/control.h +++ b/include/control.h @@ -419,6 +419,7 @@ int snd_ctl_elem_value_malloc(snd_ctl_elem_value_t **ptr); void snd_ctl_elem_value_free(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_clear(snd_ctl_elem_value_t *obj); void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value_t *src); +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right); void snd_ctl_elem_value_get_id(const snd_ctl_elem_value_t *obj, snd_ctl_elem_id_t *ptr); unsigned int snd_ctl_elem_value_get_numid(const snd_ctl_elem_value_t *obj); snd_ctl_elem_iface_t snd_ctl_elem_value_get_interface(const snd_ctl_elem_value_t *obj); diff --git a/src/control/control.c b/src/control/control.c index df249dd..159b09a 100644 --- a/src/control/control.c +++ b/src/control/control.c @@ -2256,6 +2256,18 @@ void snd_ctl_elem_value_copy(snd_ctl_elem_value_t *dst, const snd_ctl_elem_value } /** + * \brief compare one #snd_ctl_elem_value_t to another + * \param dst pointer to destination + * \param src pointer to source + * \return 0 on match, less than or greater than otherwise, see memcmp + */ +int snd_ctl_elem_value_compare(snd_ctl_elem_value_t *left, const snd_ctl_elem_value_t *right) +{ + assert(left && right); + return memcmp(left, right, sizeof(*left)); +} + +/** * \brief Get CTL element identifier of a CTL element id/value * \param obj CTL element id/value * \param ptr Pointer to returned CTL element identifier diff --git a/src/control/setup.c b/src/control/setup.c index 408244e..3606daf 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -192,7 +192,17 @@ int snd_sctl_remove(snd_sctl_t *h) return err; } } - if (elem->preserve) { + /* Only restore the old value if it differs from the requested + * value, because if it has changed restoring the old value + * overrides the change. Take for example, a voice modem with + * a .conf that sets preserve off-hook. Start playback (on-hook + * to off-hook), start record (off-hook to off-hook), stop + * playback (off-hook to restore on-hook), stop record (on-hook + * to restore off-hook), Clearly you don't want to leave the + * modem "on the phone" now that there isn't any playback or + * recording active. + */ + if (elem->preserve && snd_ctl_elem_value_compare(elem->val, elem->old)) { err = snd_ctl_elem_write(h->ctl, elem->old); if (err < 0) { SNDERR("Cannot restore ctl elem");

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[alsa-devel] default alsa pcm improvement
by Aleksandar Kostadinov 23 Jan '10

23 Jan '10

Hello, _From some time now ALSA is a real pleasure to use. Namely it works by OOB and has lots of predefined useful PCMs so one doesn't need to be a guru to do advanced operations. I think that a small improvement in how default PCM is defined can help users with multiple sound cards that like to use pure ALSA and not some of the many available sound servers. I want to be able to switch the default sound device easily in asoundrc but the way the default pcm is defined makes this not very straightforward. I like to use the definitions of default:<CARD> but if I define "pcm.!default" in .asoundrc I lose the "default:<CARD>" definition. I now can rename the "default" definition in default.conf to someting like "defaultorg" and then reference it asoundrc but that requires root as well change can possibly disappear after an alsa-lib package upgrade. Or I can use the dmix and dsnoop predefined PCMs to put together PCMs that I can later reference in my "pcm.!default" definition but in this case one needs to know if the card needs dsnoop/dmix and probably other tweaks that default.conf contains. Additionally this needs to be done for every sound card one owns. So I would suggest the "default" PCM as defined now in the default.conf file to be renamed to something like "alsadefault". Then "default" PCM should be an alias to the alsadefault pcm. This will be backwards compatible to what users have now and owners of multiple sound devices can define their "default" pcm in asoundrc referencing the alsadefault pcm benefiting from the good work already done and not have to mess with a complex asoundrc file or make changes to the system configurations files. Please CC me in replies because I'm not subscribed. Thanks again for the cool work!

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[alsa-devel] [PATCHv2 4/6] OMAP4: MCPDM: Register McPDM platform device
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Add platform device for McPDM interface in OMAP4 and register its memory and irq resources. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/devices.c | 37 +++++++++++++++++++++++++++++++++++++ 1 files changed, 37 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/devices.c b/arch/arm/plat-omap/devices.c index 30b5db7..572265c 100644 --- a/arch/arm/plat-omap/devices.c +++ b/arch/arm/plat-omap/devices.c @@ -28,6 +28,7 @@ #include <plat/menelaus.h> #include <plat/mcbsp.h> #include <plat/dsp_common.h> +#include <plat/omap44xx.h> #if defined(CONFIG_OMAP_DSP) || defined(CONFIG_OMAP_DSP_MODULE) @@ -192,6 +193,41 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, /*-------------------------------------------------------------------------*/ +#if defined(CONFIG_SND_OMAP_SOC_MCPDM) || \ + defined(CONFIG_SND_OMAP_SOC_MCPDM_MODULE) + +static struct resource mcpdm_resources[] = { + { + .name = "mcpdm_mem", + .start = OMAP44XX_MCPDM_BASE, + .end = OMAP44XX_MCPDM_BASE + SZ_4K, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcpdm_irq", + .start = INT_44XX_MCPDM_IRQ, + .end = INT_44XX_MCPDM_IRQ, + .flags = IORESOURCE_IRQ, + }, +}; + +static struct platform_device omap_mcpdm_device = { + .name = "omap-mcpdm", + .id = -1, + .num_resources = ARRAY_SIZE(mcpdm_resources), + .resource = mcpdm_resources, +}; + +static void omap_init_mcpdm(void) +{ + (void) platform_device_register(&omap_mcpdm_device); +} +#else +static inline void omap_init_mcpdm(void) {} +#endif + +/*-------------------------------------------------------------------------*/ + #if defined(CONFIG_MMC_OMAP) || defined(CONFIG_MMC_OMAP_MODULE) || \ defined(CONFIG_MMC_OMAP_HS) || defined(CONFIG_MMC_OMAP_HS_MODULE) @@ -385,6 +421,7 @@ static int __init omap_init_devices(void) omap_init_dsp(); omap_init_kp(); omap_init_rng(); + omap_init_mcpdm(); omap_init_uwire(); omap_init_wdt(); return 0; -- 1.6.0.4

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[alsa-devel] [PATCHv2 3/6] ARM: OMAP4: Add McPDM base address
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Define McPDM physical and L3 base address for OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/omap44xx.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/omap44xx.h b/arch/arm/plat-omap/include/plat/omap44xx.h index 8e4c581..4c64f20 100644 --- a/arch/arm/plat-omap/include/plat/omap44xx.h +++ b/arch/arm/plat-omap/include/plat/omap44xx.h @@ -43,6 +43,8 @@ #define OMAP44XX_SCU_BASE 0x48240000 #define OMAP44XX_LOCAL_TWD_BASE 0x48240600 #define OMAP44XX_WKUPGEN_BASE 0x48281000 +#define OMAP44XX_MCPDM_BASE 0x40132000 +#define OMAP44XX_MCPDM_L3_BASE 0x49032000 #define OMAP44XX_MAILBOX_BASE (L4_44XX_BASE + 0xF4000) -- 1.6.0.4

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[alsa-devel] [PATCHv2 2/6] OMAP4: IRQ: Add McPDM IRQ definition.
by Candelaria Villareal, Jorge 23 Jan '10

23 Jan '10

Support for McPDM IRQs in OMAP4 Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- arch/arm/plat-omap/include/plat/irqs.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/plat-omap/include/plat/irqs.h b/arch/arm/plat-omap/include/plat/irqs.h index c0ab7c8..bbb8eb6 100644 --- a/arch/arm/plat-omap/include/plat/irqs.h +++ b/arch/arm/plat-omap/include/plat/irqs.h @@ -432,7 +432,7 @@ #define INT_44XX_PARTHASH_IRQ (79 + IRQ_GIC_START) #define INT_44XX_MMC3_IRQ (94 + IRQ_GIC_START) #define INT_44XX_MMC4_IRQ (96 + IRQ_GIC_START) - +#define INT_44XX_MCPDM_IRQ (112 + IRQ_GIC_START) /* Max. 128 level 2 IRQs (OMAP1610), 192 GPIOs (OMAP730/850) and * 16 MPUIO lines */ -- 1.6.0.4

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[alsa-devel] Conexant 5051 subsystem 0x103c30d6
by Jacopo De Simoi 22 Jan '10

22 Jan '10

Dear all, this is a follow up to my previous posting to the ml about having CX 5051 working on my laptop; I found out that the patch provided in http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024284.ht… fits perfectly my model as well. Hence I suggest to add a quirk matching my subsystem id as follows: SND_PCI_QUIRK(0x103c, 0x30d6, "HP DV2700", CXT5051_F700) Thanks __J

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[alsa-devel] Reg: Audio recording with twl4030 codec
by shariff 22 Jan '10

22 Jan '10

Hi all, I am working with OMAP3 based custom board that has twl4030 codec & using linux-2.6.29 kernel from opensource. I have some queries regarding Audio recording. Audio codec is connected to twl4030 as bellow TWL4030 |--------------------| | |------ MAINMIC McBSP2 ------ |i2s/tdm | | ___________| --> One Analog MIC is connected to Main Mic and I am using Audio path not the Voice path. After selecting Main Mic through amixer control. and recording audio using "arecord -c 1 -r 8000 -f S16_LE -t wav -d 20 record-file.wav" --> Here record is successful but with lot of noise. 1) Can we record Mono(1 channel) through I2S from TWL4030 codec? 2) To record stereo(2 channels), do we need two Mic's ? (i.e do I need Sub Mic also). Regards, Shariff

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Re: [alsa-devel] [PATCH 1/4] ASoC: add a WM8978 codec driver
by Guennadi Liakhovetski 22 Jan '10

22 Jan '10

On Tue, 19 Jan 2010, Mark Brown wrote: > On Tue, Jan 19, 2010 at 09:08:57AM +0100, Guennadi Liakhovetski wrote: > > This all looks pretty good - there's a few issues below but they're > largely stylistic rather than anything fundamental. > > > + 0x0000, 0x0000, 0x0000, 0x01ff, /* 0x08...0x0b */ /* 0x0b contains the VU bit */ > > + 0x01ff, 0x0000, 0x0100, 0x01ff, /* 0x0c...0x0f */ /* 0x0c and 0x0f contain the VU bit */ > > + 0x01ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ /* 0x10 contains the VU bit */ > > What do these comments refer to - do you mean to say that these are not > the actual chip register defaults? The normal way to deal with those is > to have a write (or other update of the cache defaults). This avoids > potential confusion later on if the chip updates the register defaults > or when reviewing against the datasheet. No, I meant, that there's a "Value Update" bit in that register, usually bit 8. So, that bit, of course, is not in default, but it was suggested to me on IRC to set it in cache to force automatic value update. > > +#define ARRAY_SINGLE(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ > > + ARRAY_SIZE(xtexts), xtexts) > > This looks generic - please namespace it and add it to the header file, > other drivers can benefit from it. SOC_ARRAY_SINGLE()? > > +static const struct soc_enum wm8978_enum[] = { > > + /* adc */ > > + ARRAY_SINGLE(WM8978_COMPANDING_CONTROL, 1, wm8978_companding), > > Please define individual variables for this - indexing into the array > gets hard to follow, it's far too easy to have an off by one errors > trying to match things up. > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > > > +/* Mixer #3: Boost (Input) mixer */ > > +static const struct snd_kcontrol_new wm8978_left_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Left PGA Mute", WM8978_LEFT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > +static const struct snd_kcontrol_new wm8978_right_boost_mixer[] = { > > +SOC_DAPM_SINGLE("Right PGA Mute", WM8978_RIGHT_INP_PGA_CONTROL, 6, 1, 0), > > +}; > > These should have "Switch" at the end of the name. Also, the names are > going be concatenated with the mixer names so you'll end up with things > "Left Boost Mixer Left PGA Switch" - the individual controls should > probably be just "Switch" so you end up with "Left Boost Mixer Switch". > > > +/* Mic Input boost vol */ > > +static const struct snd_kcontrol_new wm8978_mic_boost_controls[] = { > > +SOC_DAPM_SINGLE("Left Mic Volume", WM8978_LEFT_ADC_BOOST_CONTROL, 4, 7, 0), > > +SOC_DAPM_SINGLE("Right Mic Volume", WM8978_RIGHT_ADC_BOOST_CONTROL, 4, 7, 0), > > +}; > > This doesn't seem to be referenced anywhere? > > > +#define MIXER_ARRAY(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ > > + m, ARRAY_SIZE(m)) > > Again, this should be done somewhere generic. Probably by going through > and changing the SND_SOC_DAPM_MIXER definition. Well, having discussed this privately and having looked at the heades, I'm not sure, this is a good idea. First, there are SND_SOC_DAPM_PGA() SND_SOC_DAPM_MIXER() SND_SOC_DAPM_MIXER_NAMED_CTL() SND_SOC_DAPM_PGA_E() SND_SOC_DAPM_MIXER_E() SND_SOC_DAPM_MIXER_NAMED_CTL_E() - 6 macros that follow the pattern .kcontrols = wcontrols, .num_kcontrols = wncontrols, and lots of macros like SOC_ENUM_DOUBLE() SOC_ENUM_SINGLE() SOC_ENUM_SINGLE_EXT() ... doing .max = xmax. BTW, many of the users even open-code array sizes instead of using ARRAY_SIZE() like static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", "Off"}; static const char *spk_function[] = {"On", "Off"}; static const struct soc_enum tosa_enum[] = { SOC_ENUM_SINGLE_EXT(5, jack_function), SOC_ENUM_SINGLE_EXT(2, spk_function), }; Changing only one of them would make the headers and the API look inconsistent. Changing all of them... I really don't think we want to endeavor this now, at least not me;) Besides, doing this locally is very convenient, and is available to everyone - there's no space science involved with this. One can be even as evil as doing +#define _A(n, r, s, i, m) SND_SOC_DAPM_MIXER(n, r, s, i, \ + m, ARRAY_SIZE(m)) ... +#undef _A +#define _A(xreg, xshift, xtexts) SOC_ENUM_SINGLE(xreg, xshift, \ + ARRAY_SIZE(xtexts), xtexts) ... +#undef _A I think, it's best to leave these things as they are. Otherwise we can give this to kernel janitors as a small exercise;) > > + /* Output PLL to GPIO1 */ > > + snd_soc_write(codec, WM8978_GPIO_CONTROL, > > + snd_soc_read(codec, WM8978_GPIO_CONTROL) | 0x4); > > This should be being done unconditionally. Put a switch in via the > set_dai_clkdiv() call. Ok, I moved it to wm8978_set_dai_clkdiv() under "case WM8978_OPCLKDIV:" > > + /* Mic bias */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + (snd_soc_read(codec, 1) & ~4) | 0x10); > > + > > + /* Out-1 enabled, left/right input channel enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_2, 0x1bf); > > + > > + /* Out-2 disabled, right/left output channel enabled, dac enabled */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_3, 0x10f); > > These should all be being managed via DAPM. Yes, I'm having quite a bit of trouble with DAPM, but I'm getting through slowly. Will need some more time for a next version. > > > +static int wm8978_set_bias_level(struct snd_soc_codec *codec, > > + enum snd_soc_bias_level level) > > +{ > > + u16 power1 = snd_soc_read(codec, WM8978_POWER_MANAGEMENT_1) & ~3; > > This bitmask maintains everything except the two LSB... > > > + switch (level) { > > + case SND_SOC_BIAS_ON: > > + case SND_SOC_BIAS_PREPARE: > > + power1 |= 1; /* VMID 75k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, power1); > > + break; > > + case SND_SOC_BIAS_STANDBY: > > + power1 |= 0xC; > > ...but this is also managing other bits. Right, bits 7-6 are either kept or set, nothing wrong with that. > > + if (codec->bias_level == SND_SOC_BIAS_OFF) { > > + /* Initial cap charge at VMID 5k */ > > + snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, > > + power1 | 0x3); > > + mdelay(100); > > + } > > > +/* Also supports 12kHz */ > > +#define WM8978_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ > > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | \ > > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) > > SNDRV_PCM_RATE_8000_48000 > > > +static int wm8978_suspend(struct platform_device *pdev, pm_message_t state) > > +{ > > + struct snd_soc_device *socdev = platform_get_drvdata(pdev); > > + struct snd_soc_codec *codec = socdev->card->codec; > > + > > + /* we only need to suspend if we are a valid card */ > > + if (!codec->card) > > + return 0; > > Don't need to check for the card any more, the core will stop you > getting called without a card. > > > + /* Sync reg_cache with the hardware */ > > + for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { > > + if (i == WM8978_RESET) > > + continue; > > You can also skip the write if the register has the default value (this > will speed up resume slightly). > > > + data = cpu_to_be16((i << 9) | (cache[i] & 0x1ff)); > > + codec->hw_write(codec->control_data, (char *)&data, 2); > > + } > > Just use snd_soc_write()? I'll take care about the rest. Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Add PCI quirk for 17" MacBook Pro 5, 2 model (July 2009) to use existing mb5 model
by murray.alex＠gmail.com 22 Jan '10

22 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 7e0ea37..d044db9 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -9223,6 +9223,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x10ec, 0x0885, "MacbookPro 5,2", ALC885_MB5), {} /* terminator */ }; -- 1.6.3.3

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