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[alsa-devel] [PATCH] ASoC: Fix long name of control for dapm_switch
by Joonyoung Shim 06 Mar '09

06 Mar '09

The length of long control name doesn't include the length of widget name at present when widget->id is snd_soc_dapm_switch, so the long control name of dapm_switch is shortened. To avoid this problem, the long control name only for dapm_mixer_named_ctl must have simply the kcontrol name. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-dapm.c | 21 +++++++++------------ 1 files changed, 9 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4b8dbbf..7a1fed1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -331,26 +331,23 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, * for dapm_mixer_named_ctl this is simply the * kcontrol name. */ - name_len = strlen(w->kcontrols[i].name) + 1; - if (w->id == snd_soc_dapm_mixer) - name_len += 1 + strlen(w->name); + if (w->id == snd_soc_dapm_mixer_named_ctl) + name_len = strlen(w->kcontrols[i].name) + 1; + else + name_len = 2 + strlen(w->name) + + strlen(w->kcontrols[i].name); path->long_name = kmalloc(name_len, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; - switch (w->id) { - case snd_soc_dapm_mixer: - default: - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); - break; - case snd_soc_dapm_mixer_named_ctl: + if (w->id == snd_soc_dapm_mixer_named_ctl) snprintf(path->long_name, name_len, "%s", w->kcontrols[i].name); - break; - } + else + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); path->long_name[name_len - 1] = '\0'; -- 1.5.6.3 -- - Joonyoung Shim

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[alsa-devel] Fwd: [LAD] kernel 2.6.29 rc6 no working rme multiface 1
by Dmitry Baikov 05 Mar '09

05 Mar '09

---------- Forwarded message ---------- From: Dmitry Baikov <dsbaikov(a)gmail.com> Date: Mon, Mar 2, 2009 at 5:49 AM Subject: Re: [LAD] kernel 2.6.29 rc6 no working rme multiface 1 To: "m.wolkstein(a)gmx.de" <m.wolkstein(a)gmx.de> Cc: The Linux Audio Developers' Mailing List <linux-audio-dev(a)lists.linuxaudio.org> On Mon, Mar 2, 2009 at 4:59 AM, m.wolkstein(a)gmx.de <m.wolkstein(a)gmx.de> wrote: > hiho, > my name is michael wolkstein and i wrote the first time to this list. > sorry about my english. but i am no english native-speaker. > > after booting kernel 2.6.29 rc6 i get a problem with my rme multiface 1 connected true pcmcia cardbusinterface. > > i get this message after insert my card. > RME Hammerfall DSP 0000:16:00.0: enabling device (0000 -> 0002) > RME Hammerfall DSP 0000:16:00.0: PCI INT A -> GSI 16 (level, low) -> IRQ 16 > Hammerfall-DSP: no Digiface or Multiface connected! > RME Hammerfall DSP 0000:16:00.0: PCI INT A disabled > RME Hammerfall DSP: probe of 0000:16:00.0 failed with error -5 > > i have compile the kernel included alsa module. > also i have installed the newest alsa-firmware 1.0.19 and newest alsa-tools 1.0.19. > > my old kernel 2.6.24 load both firmwares correct. > > the old one 1.0.16 (debian unstable) > and the new 1.0.19 download from the alsa-project page > > the new kernel produce errors. kernel module snd_hdsp is already loaded with the new kernel. Have the same problem here. 2.6.24 works, 2.6.28 doesn't (tried 2.6.8-ARCH and 2.6.28.7-1-ARCH). I reported the problem to alsa-devel list, but they said I have hw problems (I think I do not). The latest arch kernel gives oops, if the card is inserted before booting. It crashes during udev event processing (i.e. something from userspace triggers the problem) If anybody is really interested, I can try to provide kernel backtrace (either type it in, or try to use serial console). Dmitry.

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[alsa-devel] [PATCH 1/4] ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite
by Philipp Zabel 05 Mar '09

05 Mar '09

For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result in exactly the same behaviour. Now it is possible to use 16-bit single slot transfers in pxa-ssp, which are needed for Magician to get two frame clock pulses per sample (one for each channel). Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 3 +-- sound/soc/pxa/zylonite.c | 7 +++++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4a973ab..c49bb12 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -644,8 +644,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - if (params_channels(params) > 1) - sscr0 |= SSCR0_EDSS; + /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 0140a25..9f6116e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -127,8 +127,11 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* We're not really in network mode but the emulation wants this. */ - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + /* Use network mode for stereo, one slot per channel. */ + if (params_channels(params) > 1) + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2); + else + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); if (ret < 0) return ret; -- 1.6.1.3

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[alsa-devel] Keyboard volume buttons
by Harsha, Priya 05 Mar '09

05 Mar '09

Hi, Can anyone tell me how keyboard's volume buttons are converted into real volume change? Will there be an application that monitors the keyboard volume button events and call ALSA API to set the volume? Thanks Harsha

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[PATCH] pxa-ssp: add two more bitfields
by Daniel Mack 05 Mar '09

05 Mar '09

Add two more bitfields for the PSP register. As they seem to exist for PXA3xx only, define them conditionally. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index a9ee355..018f6d6 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -109,6 +109,11 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ +#if defined(CONFIG_PXA3xx) +#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ +#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ +#endif + #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ #define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ -- 1.6.1.3

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Add mixer control for the HS ramp delay
by Peter Ujfalusi 05 Mar '09

05 Mar '09

Hello, The following patch adds user control for the desired ramp delay used on the Headset output - it has been hardwired to use 874/645/437 ms. The actuall delay depends on the MCLK frequency, for example: value 0x0 in RAMP_DELAY means 2^19/f(MCLK), which is: 27 ms when the f(MCLK) = 19.2 MHz 20 ms when the f(MCLK) = 26 MHz 14 ms when the f(MCLK) = 38.4 MHz value 0x5 in RAMP_DELAY means 2^24/f(MCLK), which is: 874 ms when the f(MCLK) = 19.2 MHz 645 ms when the f(MCLK) = 26 MHz 437 ms when the f(MCLK) = 38.4 MHz I think having the actual ms values in the enums are makes more sense than to have the calculation ("2^24/F(MCLK)" for example). If one can suggest more descriptive name, I would love to change the texts. --- Peter Ujfalusi (1): ASoC: TWL4030: Make the HS ramp delay configurable sound/soc/codecs/twl4030.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-)

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[alsa-devel] pcm_oss: Can period_bytes - buffer_used be less than 0?
by Roel Kluin 05 Mar '09

05 Mar '09

vi sound/core/oss/pcm_oss.c +1572 (in linus' git tree) and note: static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) { ... size_t size; ... :1608 size = runtime->oss.period_bytes - runtime->oss.period_ptr; can this period_bytes - buffer_used become less than 0? Roel

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Re: [alsa-devel] [PATCH] ASoC: make ops a pointer in 'struct snd_soc_dai'
by Timur Tabi 05 Mar '09

05 Mar '09

Eric Miao wrote: > Updated as below, (gzip version attached due to mailing list > mail size limitation). > > From 74a2b6dfff2948d9d04b33d0a02f645f8c12caf7 Mon Sep 17 00:00:00 2001 > From: Eric Miao <eric.miao(a)marvell.com> > Date: Mon, 2 Mar 2009 22:38:31 +0800 > Subject: [PATCH] ASoC: make ops a pointer of 'struct snd_soc_dai' > > Considering the fact that most cpu_dai or codec_dai are using a same > 'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better > made a pointer instead, to make sharing easier and code a bit cleaner. > > The patch below is rather preliminary since the asoc tree is being > actively developed, and this touches almost every piece of code, > (and possibly many others in development need to be changed as > well). Building of all codecs are OK, yet to every SoC, I didn't test > that. > > Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Tested on an MPC8610 HPCD, and it works. That's all I can vouch for, though. Acked-by: Timur Tabi <timur(a)freescale.com> -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 2/2] ASoC: Add headset jack detection for SDP3430 machine driver
by Lopez Cruz, Misael 04 Mar '09

04 Mar '09

Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/sdp3430.c | 49 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4eab4b4..b595aa4 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -122,7 +123,7 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); @@ -144,6 +145,29 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Headset jack */ +struct snd_soc_jack *hs_jack; + +/* Headset jack DAPM pins */ +struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + .invert = 0, + }, +}; + +/* Headset jack gpios */ +struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .invert = 0, + .debounce_time = 200, + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -194,8 +218,25 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Headset jack detection */ + hs_jack = kzalloc(sizeof(struct snd_soc_jack), GFP_KERNEL); + if (!hs_jack) + return -ENOMEM; + + ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", + SND_JACK_HEADSET, hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); platform_device_put(sdp3430_snd_device); @@ -206,6 +247,10 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + kfree(hs_jack); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); -- 1.5.4.3

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[alsa-devel] [PATCH 1/3] ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888)
by Herton Ronaldo Krzesinski 04 Mar '09

04 Mar '09

Add additional 4 channel mode for 3stack-hp models. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 1 files changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c02f78..3696ff3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8068,24 +8068,45 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; -- 1.6.1.3

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