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[alsa-devel] Jack insertion
by naveen krishna ch 18 Nov '08

18 Nov '08

Hello Alsa Devels I am working on a OMAP3 + TWL4030 combination for audio (alsa soc) I could develop the ASOC drivers with limited kcontorls and DAPM widgets for the setup. Now my issue is is there any work regarding the Headset JACK insertion/detection. Earlier there was a proposal for a jack insertion layer. Can anyone suggest me in this regard. Thank You, (: Naveen Krishna Ch :)

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[alsa-devel] [PATCH 001/002] snd-powermac: enable mic on iMac G4
by Risto Suominen 18 Nov '08

18 Nov '08

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[alsa-devel] [PATCH 0/3] HDMI ELD routines and proc interface
by Wu Fengguang 18 Nov '08

18 Nov '08

Hi Takashi, Here are the patches for separated HDMI ELD routines and proc interface. They have undergone basic tests on a G45 board and stress test with the patch below. [PATCH 1/3] introduce snd_print_pcm_rates() [PATCH 2/3] create hda_eld.c for ELD routines and proc interface [PATCH 3/3] ELD proc interface for HDMI sinks Here are two sample outputs from the stress testing: monitor name connection_type 2-reserved eld_version [0x0] reserved edid_version [0x0] no CEA EDID Timing Extension block present manufacture_id 0x3c93 product_id 0xc5ac port_id 0x486d85d5a3bfe049 support_hdcp 1 support_ai 0 audio_sync_delay 152 speakers [0x6f] FL/FR LFE FC RL/RR FLC/FRC RLC/RRC sad_count 8 sad0_coding_type [0x0] undefined sad0_channels 0 sad0_sampling_rates [0x0] sad1_coding_type [0x0] undefined sad1_channels 0 sad1_sampling_rates [0x0] sad2_coding_type [0x0] undefined sad2_channels 0 sad2_sampling_rates [0x0] sad3_coding_type [0x0] undefined sad3_channels 0 sad3_sampling_rates [0x0] sad4_coding_type [0x0] undefined sad4_channels 0 sad4_sampling_rates [0x0] sad5_coding_type [0x0] undefined sad5_channels 0 sad5_sampling_rates [0x0] sad6_coding_type [0x0] undefined sad6_channels 0 sad6_sampling_rates [0x0] sad7_coding_type [0x0] undefined sad7_channels 0 sad7_sampling_rates [0x0] monitor name ��p connection_type Display Port eld_version [0x0] reserved edid_version [0x7] reserved manufacture_id 0x59c7 product_id 0x3c72 port_id 0x323d7444a5fcbdba support_hdcp 0 support_ai 1 audio_sync_delay 268 speakers [0x33] FL/FR LFE RC FLC/FRC sad_count 14 sad0_coding_type [0x9] DSD (1-bit audio) sad0_channels 5 sad0_sampling_rates [0xe40] 48000 176400 192000 384000 sad1_coding_type [0x6] AAC-LC sad1_channels 2 sad1_sampling_rates [0x18a0] 44100 88200 384000 sad1_max_bitrate 1024000 sad2_coding_type [0xc] MLP (Dolby TrueHD) sad2_channels 8 sad2_sampling_rates [0x1ec0] 48000 88200 176400 192000 384000 sad3_coding_type [0x8] ATRAC sad3_channels 3 sad3_sampling_rates [0x1220] 44100 176400 sad3_max_bitrate 944000 sad4_coding_type [0xd] DST sad4_channels 3 sad4_sampling_rates [0xac0] 48000 88200 176400 384000 sad5_coding_type [0x2] AC-3 sad5_channels 3 sad5_sampling_rates [0x1a60] 44100 48000 176400 384000 sad5_max_bitrate 592000 sad6_coding_type [0xc] MLP (Dolby TrueHD) sad6_channels 3 sad6_sampling_rates [0xc40] 48000 192000 384000 sad7_coding_type [0x9] DSD (1-bit audio) sad7_channels 6 sad7_sampling_rates [0xc20] 44100 192000 384000 sad8_coding_type [0x5] MPEG2 sad8_channels 1 sad8_sampling_rates [0x1ac0] 48000 88200 176400 384000 sad8_max_bitrate 544000 sad9_coding_type [0xc] MLP (Dolby TrueHD) sad9_channels 7 sad9_sampling_rates [0x460] 44100 48000 192000 sad10_coding_type [0x1] LPCM sad10_channels 4 sad10_sampling_rates [0x1260] 44100 48000 176400 sad10_sample_bits 0x40000 sad11_coding_type [0x6] AAC-LC sad11_channels 7 sad11_sampling_rates [0x1240] 48000 176400 sad11_max_bitrate 408000 sad12_coding_type [0x0] undefined sad12_channels 0 sad12_sampling_rates [0x0] sad13_coding_type [0x0] undefined sad13_channels 0 sad13_sampling_rates [0x0] Thank you, Fengguang --- stress testing patch for ELD routines sound/pci/hda/hda_eld.c | 30 ++++++++++++++++++++---------- sound/pci/hda/patch_intelhdmi.c | 2 ++ 2 files changed, 22 insertions(+), 10 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_eld.c +++ sound-2.6/sound/pci/hda/hda_eld.c @@ -27,6 +27,15 @@ #include "hda_codec.h" #include "hda_local.h" +static unsigned long rand; + +static inline unsigned char random(void) +{ + /* See "Numerical Recipes in C", second edition, p. 284 */ + rand = rand * 1664525L + 1013904223L; + return (unsigned char) (rand >> 24); +} + enum eld_versions { ELD_VER_CEA_861D = 2, ELD_VER_PARTIAL = 31, @@ -167,6 +176,8 @@ static unsigned char hdmi_get_eld_byte(s val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); + val = AC_ELDD_ELD_VALID | random(); + #ifdef BE_PARANOID printk(KERN_INFO "ELD data byte %d: 0x%x\n", byte_index, val); #endif @@ -277,7 +288,6 @@ static int hdmi_update_sink_eld(struct s if (e->eld_ver != ELD_VER_CEA_861D && e->eld_ver != ELD_VER_PARTIAL) { snd_printd(KERN_INFO "Unknown ELD version %d\n", e->eld_ver); - goto out_fail; } e->eld_size = size; @@ -359,9 +369,6 @@ int snd_hdmi_get_eld(struct sink_eld *el int size; unsigned char *buf; - if (!hdmi_eld_valid(codec, nid)) - return -ENOENT; - size = snd_hdmi_get_eld_size(codec, nid); if (size == 0) { /* wfg: workaround for ASUS P5E-VM HDMI board */ @@ -516,6 +523,9 @@ int snd_hda_eld_proc_new(struct hda_code struct snd_info_entry *entry; int err; + /* Initialize random number generator */ + rand = jiffies; + snprintf(name, sizeof(name), "eld#%d", codec->addr); err = snd_card_proc_new(codec->bus->card, name, &entry); if (err < 0) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c @@ -635,6 +635,8 @@ static int patch_intel_hdmi(struct hda_c init_channel_allocations(); + hdmi_parse_eld(codec); + return 0; }

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[alsa-devel] [PATCH] hda: remove redundant get_amp_nid()
by Wu Fengguang 18 Nov '08

18 Nov '08

Remove get_amp_nid(): it duplicates the one defined in hda_local.h Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_via.c +++ sound-2.6/sound/pci/hda/patch_via.c @@ -142,8 +142,6 @@ enum { AUTO_SEQ_SIDE }; -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) - /* Some VT1708S based boards gets the micboost setting wrong, so we have * to apply some brute-force and re-write the TLV's by software. */ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,

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[alsa-devel] (no subject)
by Fernando Moro 18 Nov '08

18 Nov '08

Dear member of the alsa time, I am not sure if this is the email addres in which it is possible to talk about a problem with the Alsa plugins, but it is the unique address that I have found. I am writing to tell you about the compatibility of the alsa plugins. There are lots of users (like me) who have the Lexicon lambda soundcard and the lexicon alpha soundcard, but alsa plugins only support the Lexicon Omega!, and it is impossible to connect our soundcards to the computer. I think that it is a problem for the Ubuntu Studio, Musix or Studio64 users, who use this devices to record and play midi songs, and I think that linux can be better than windows making audio. I would be grateful if alsa team could help us with this problem somehow. I look forward to your response. Thank you for your attention. Your faithfully, Fernando. Sorry for my bad English, I am Spanish. _________________________________________________________________ Los más de lo más, Especial Rankings http://events.es.msn.com/dinero/listas/default.aspx

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[alsa-devel] [PATCH 002/002] snd-powermac: enable mic on iMac G4 (older kernels)
by Risto Suominen 17 Nov '08

17 Nov '08

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[alsa-devel] snd-usb-audio: minimum period time as module option?
by Klaus Schulz 17 Nov '08

17 Nov '08

Hi folks (probably Clemens). Would it be possible to to introduce a module-option variable to snd-usb-audio, so that it is possible to switch minimum period from outside usbaudio.c /* set the period time minimum 1ms */ /* FIXME: high-speed mode allows 125us minimum period, but many parts * in the current code assume the 1ms period. */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, 1000 * MIN_PACKS_URB, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); With 125us my EMU USB 0404 and also my PCM2707 devices do sound better. THX \Klaus

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Re: [alsa-devel] [PATCH] ALSA: hda - Add digital BEEP generator support for Realtek codecs.
by Takashi Iwai 17 Nov '08

17 Nov '08

At Sat, 15 Nov 2008 15:20:14 +0900, Kusanagi Kouichi wrote: > > Add digital BEEP generator support for Realtek codecs. > > Signed-off-by: Kusanagi Kouichi <slash(a)ma.neweb.ne.jp> Thanks for the patch. Do these beep widgets have the mute amp bits? If not, we'd likely need to add mute switches as well (otherwise it gets too annoying) controlling on software. Check patch_sigmatel.c in the latest master branch or topic/fix/hda branch of sound git tree: git://git.kernel.org/pub/scm/linux/kernel/tiwai/sound-2.6.git Another concern is that I don't want to add the digital beep control unconditionally. On many devices, the beep is implemented in the old good analog way. We'd need to add the check of analog/digital beep in a new spec field and in the preset table not to duplicate beep controls on these. thanks, Takashi > --- > sound/pci/hda/hda_beep.h | 2 +- > sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++ > 2 files changed, 63 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h > index de4036e..b483779 100644 > --- a/sound/pci/hda/hda_beep.h > +++ b/sound/pci/hda/hda_beep.h > @@ -38,7 +38,7 @@ struct hda_beep { > int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); > void snd_hda_detach_beep_device(struct hda_codec *codec); > #else > -#define snd_hda_attach_beep_device(...) > +#define snd_hda_attach_beep_device(...) 0 > #define snd_hda_detach_beep_device(...) > #endif > #endif > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index a378c01..d586600 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -31,6 +31,7 @@ > #include "hda_codec.h" > #include "hda_local.h" > #include "hda_patch.h" > +#include "hda_beep.h" > > #define ALC880_FRONT_EVENT 0x01 > #define ALC880_DCVOL_EVENT 0x02 > @@ -2788,6 +2789,7 @@ static void alc_free(struct hda_codec *codec) > kfree(spec->kctl_alloc); > } > kfree(spec); > + snd_hda_detach_beep_device(codec); > codec->spec = NULL; /* to be sure */ > } > > @@ -3960,6 +3962,12 @@ static int patch_alc880(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC880_AUTO) > setup_preset(spec, &alc880_presets[board_config]); > > @@ -5558,6 +5566,12 @@ static int patch_alc260(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC260_AUTO) > setup_preset(spec, &alc260_presets[board_config]); > > @@ -6811,6 +6825,12 @@ static int patch_alc882(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC882_AUTO) > setup_preset(spec, &alc882_presets[board_config]); > > @@ -8916,6 +8936,12 @@ static int patch_alc883(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC883_AUTO) > setup_preset(spec, &alc883_presets[board_config]); > > @@ -10739,6 +10765,12 @@ static int patch_alc262(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC262_AUTO) > setup_preset(spec, &alc262_presets[board_config]); > > @@ -11752,6 +11784,12 @@ static int patch_alc268(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC268_AUTO) > setup_preset(spec, &alc268_presets[board_config]); > > @@ -12458,6 +12496,12 @@ static int patch_alc269(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC269_AUTO) > setup_preset(spec, &alc269_presets[board_config]); > > @@ -13624,6 +13668,12 @@ static int patch_alc861(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x23); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC861_AUTO) > setup_preset(spec, &alc861_presets[board_config]); > > @@ -14578,6 +14628,12 @@ static int patch_alc861vd(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x23); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC861VD_AUTO) > setup_preset(spec, &alc861vd_presets[board_config]); > > @@ -16412,6 +16468,12 @@ static int patch_alc662(struct hda_codec *codec) > } > } > > + err = snd_hda_attach_beep_device(codec, 0x1); > + if (err < 0) { > + alc_free(codec); > + return err; > + } > + > if (board_config != ALC662_AUTO) > setup_preset(spec, &alc662_presets[board_config]); > > -- > 1.5.6.5 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

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[alsa-devel] [PATCH] hda: remove redundant get_amp_nid()
by Wu Fengguang 17 Nov '08

17 Nov '08

Remove get_amp_nid(): it duplicates the one defined in hda_local.h Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 -- 1 file changed, 2 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_via.c +++ sound-2.6/sound/pci/hda/patch_via.c @@ -142,8 +142,6 @@ enum { AUTO_SEQ_SIDE }; -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) - /* Some VT1708S based boards gets the micboost setting wrong, so we have * to apply some brute-force and re-write the TLV's by software. */ static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag,

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[alsa-devel] [PATCH RESEND 3/3] ALSA SOC driver for s3c24xx with uda134x
by Christian Pellegrin 17 Nov '08

17 Nov '08

Adds audio support for SMDK2410 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- arch/arm/mach-s3c2410/mach-smdk2410.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) -- Generated on 20081115 against f7e1798a78a30bae1cf3ebc3e1b6f7c1bac57756..5b7f05e9b57e7e442b1054cbb4423cf9909fb204 -- diff --git a/arch/arm/mach-s3c2410/mach-smdk2410.c b/arch/arm/mach-s3c2410/mach-smdk2410.c index 152527b..0001d52 100644 --- a/arch/arm/mach-s3c2410/mach-smdk2410.c +++ b/arch/arm/mach-s3c2410/mach-smdk2410.c @@ -43,6 +43,7 @@ #include <asm/mach/irq.h> #include <mach/hardware.h> +#include <mach/regs-gpio.h> #include <asm/irq.h> #include <asm/mach-types.h> @@ -53,6 +54,8 @@ #include <plat/common-smdk.h> +#include <sound/s3c24xx_uda134x.h> + static struct map_desc smdk2410_iodesc[] __initdata = { /* nothing here yet */ }; @@ -85,12 +88,27 @@ static struct s3c2410_uartcfg smdk2410_uartcfgs[] __initdata = { } }; +static struct s3c24xx_uda134x_platform_data s3c24xx_uda134x_data = { + .l3_clk = S3C2410_GPB4, + .l3_data = S3C2410_GPB3, + .l3_mode = S3C2410_GPB2, + .model = UDA134X_UDA1341, +}; + +static struct platform_device s3c24xx_uda134x = { + .name = "s3c24xx_uda134x", + .dev = { + .platform_data = &s3c24xx_uda134x_data, + } +}; + static struct platform_device *smdk2410_devices[] __initdata = { &s3c_device_usb, &s3c_device_lcd, &s3c_device_wdt, &s3c_device_i2c, &s3c_device_iis, + &s3c24xx_uda134x, }; static void __init smdk2410_map_io(void) -- 1.4.4.4

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