March 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

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[alsa-devel] [PATCHv4 7/7] ASoC: TWL6030: Detect power-up sequence completion
by Olaya, Margarita 02 Mar '10

02 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. If NAUDINT irq is provided, CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. If AUDPWRON gpio line is provided but NAUDINT irq is not, then CODEC driver enables READYINT and polls on INTID register. If none of them are provided, then CODEC uses manual power sequences and disables all audio interrupts. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 61 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 1 + 2 files changed, 53 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index b8dd5ae..2847f1b 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -52,6 +52,7 @@ struct twl6030_data { int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; }; /* @@ -372,6 +373,7 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, static irqreturn_t twl6030_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6030_REG_INTID); @@ -391,7 +393,7 @@ static irqreturn_t twl6030_naudint_handler(int irq, void *data) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + complete(&priv->ready); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -626,11 +628,45 @@ static int twl6030_add_widgets(struct snd_soc_codec *codec) return 0; } +static int twl6030_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6030_data *priv = codec->private_data; + int time_left; + u8 intid; + + if (naudint) { + /* wait for ready interrupt with 48 ms timeout */ + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + } else { + /* retry 3 times only */ + for (time_left = 3; time_left > 0; time_left--) { + mdelay(16); + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6030_REG_INTID); + if (intid & TWL6030_READYINT) + break; + } + } + + if (!time_left) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + + priv->codec_powered = 1; + + return 0; +} + static int twl6030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct twl6030_data *priv = codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -643,8 +679,10 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for power-up completion */ + ret = twl6030_power_up_completion(codec, naudint); + if (ret) + return ret; /* sync registers updated during power-up sequence */ twl6030_read(codec, TWL6030_REG_NCPCTL); @@ -653,12 +691,11 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -1067,6 +1104,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1089,10 +1127,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) if (ret) goto gpio2_err; } else { - dev_warn(codec->dev, - "no naudint irq, audio interrupts disabled\n"); - twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, - TWL6030_ALLINT_MSK); + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + ~TWL6030_READYMSK & TWL6030_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 90b8a44..d591be1 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6030_READYMSK 0x40 #define TWL6030_ALLINT_MSK 0x7B /* NCPCTL (0x05) fields */ -- 1.6.1.3

2 8

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adap
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? Regards, Pedro

1 0

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adapter
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? (Please CC me directly as I'm not in both lists). Regards, Pedro

1 0

[alsa-devel] [PATCH 2.6.34] ASoC: Fix S3C64xx IIS driver for Samsung header reorg
by Mark Brown 01 Mar '10

01 Mar '10

The reorgs of the Samsung headers have moved the GPIO bank definitions from plat/ to mach/ - the IIS driver needs to be updated to take care of this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Ideally this could also go in via the same path as the ARM/Samsung changes which do the moves. If those get merged without this fix then it may as well go via ASoC. sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 93ed3aa..a72c251 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -22,8 +22,8 @@ #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> -#include <plat/gpio-bank-d.h> -#include <plat/gpio-bank-e.h> +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> #include <plat/gpio-cfg.h> #include <mach/map.h> -- 1.7.0

1 0

[alsa-devel] [Regression] With the new 2.6.33 when I plug headphones, the speaker doen't get off anymore
by giggzounet 01 Mar '10

01 Mar '10

Hi, I have installed debian stable lenny + backports on an eeepc 1201n. With the 2.6.30 or 2.6.32 from lenny-backport I don't have problem with sound : when I plug the headphones, speakers get off. I have tested with 2.6.33 and when I plug headphones, speakers don't get off anymore. so I have sound in headphones and speakers. 22:42 giggz@baal ~ % cat /proc/asound/card0/codec#* | grep Codec Codec: Realtek ALC269 Codec: Nvidia MCP7A HDMI I attach the output of alsa-info with the 2.6.32.9 and the 2.6.33 (I'm in a Uni Campus and it's quite difficult to have internet on a external laptop...so I attach them at this mail and I don't upload them on pastbin). I have opened a bug on the kernel bugzilla (bug 15399). Should I provide anything more ? Thx a lot, Best regards Guillaume

4 14

[alsa-devel] [PATCH 3/3] ALSA: usb-audio: support multiple formats with audio class v2 devices
by Clemens Ladisch 01 Mar '10

01 Mar '10

Change the parser to correctly handle v2 descriptors with multiple format bits set. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/format.c +++ linux/sound/usb/format.c @@ -37,19 +37,20 @@ * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, +static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, int format, void *_fmt, int protocol) { - int pcm_format, i; int sample_width, sample_bytes; + u64 pcm_formats; switch (protocol) { case UAC_VERSION_1: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; + format = 1 << format; break; } @@ -57,24 +58,7 @@ static int parse_audio_format_i_type(str struct uac_format_type_i_ext_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubslotSize; - - /* - * FIXME - * USB audio class v2 devices specify a bitmap of possible - * audio formats rather than one fix value. For now, we just - * pick one of them and report that as the only possible - * value for this setting. - * The bit allocation map is in fact compatible to the - * wFormatTag of the v1 AS streaming descriptors, which is why - * we can simply map the matrix. - */ - - for (i = 0; i < 5; i++) - if (format & (1UL << i)) { - format = i + 1; - break; - } - + format <<= 1; break; } @@ -82,15 +66,15 @@ static int parse_audio_format_i_type(str return -EINVAL; } - /* FIXME: correct endianess and sign? */ - pcm_format = -1; + pcm_formats = 0; - switch (format) { - case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ + if (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED)) { + /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); - /* fall-through */ - case UAC_FORMAT_TYPE_I_PCM: + format = 1 << UAC_FORMAT_TYPE_I_PCM; + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -99,22 +83,22 @@ static int parse_audio_format_i_type(str /* check the format byte size */ switch (sample_bytes) { case 1: - pcm_format = SNDRV_PCM_FORMAT_S8; + pcm_formats |= SNDRV_PCM_FMTBIT_S8; break; case 2: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S16_BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S16_LE; break; case 3: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S24_3LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3LE; break; case 4: - pcm_format = SNDRV_PCM_FORMAT_S32_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S32_LE; break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", @@ -122,30 +106,29 @@ static int parse_audio_format_i_type(str sample_width, sample_bytes); break; } - break; - case UAC_FORMAT_TYPE_I_PCM8: - pcm_format = SNDRV_PCM_FORMAT_U8; - + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM8)) { /* Dallas DS4201 workaround: it advertises U8 format, but really supports S8. */ if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - pcm_format = SNDRV_PCM_FORMAT_S8; - break; - case UAC_FORMAT_TYPE_I_IEEE_FLOAT: - pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; - break; - case UAC_FORMAT_TYPE_I_ALAW: - pcm_format = SNDRV_PCM_FORMAT_A_LAW; - break; - case UAC_FORMAT_TYPE_I_MULAW: - pcm_format = SNDRV_PCM_FORMAT_MU_LAW; - break; - default: - snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n", + pcm_formats |= SNDRV_PCM_FMTBIT_S8; + else + pcm_formats |= SNDRV_PCM_FMTBIT_U8; + } + if (format & (1 << UAC_FORMAT_TYPE_I_IEEE_FLOAT)) { + pcm_formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; + } + if (format & (1 << UAC_FORMAT_TYPE_I_ALAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_A_LAW; + } + if (format & (1 << UAC_FORMAT_TYPE_I_MULAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_MU_LAW; + } + if (format & ~0x3f) { + snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n", chip->dev->devnum, fp->iface, fp->altsetting, format); - break; } - return pcm_format; + return pcm_formats; } @@ -313,14 +296,14 @@ static int parse_audio_format_i(struct s default: pcm_format = SNDRV_PCM_FORMAT_S16_LE; } + fp->formats = 1uLL << pcm_format; } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); - if (pcm_format < 0) + fp->formats = parse_audio_format_i_type(chip, fp, format, + fmt, protocol); + if (!fp->formats) return -1; } - fp->formats = 1uLL << pcm_format; - /* gather possible sample rates */ /* audio class v1 reports possible sample rates as part of the * proprietary class specific descriptor.

2 2

[PATCH] ALSA: usb-audio: fix quirk formats
by Daniel Mack 01 Mar '10

01 Mar '10

'struct audioformat' takes a bit field of supported rates now rather than one fixed value. Adopt that change at two more places. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ff2df4e..2107d00 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2233,7 +2233,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3BE, + .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, .altsetting = 1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 00e8a97..0119fa0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -89,7 +89,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua1000_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_S32_LE, .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, -- 1.6.6.2

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[alsa-devel] [PATCH] opti92x: use PnP data to select Master Control port
by Krzysztof Helt 01 Mar '10

01 Mar '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Master Control port (MC) is available as the last PnP resource (OPT005). Use this value instead fo guessing. Also, add some comments to the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 120 +++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 43 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b865e45..5913717 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1558,7 +1558,7 @@ static int __devinit snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index a4af53b..becd90d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -144,12 +144,8 @@ struct snd_opti9xx { spinlock_t lock; + long wss_base; int irq; - -#ifdef CONFIG_PNP - struct pnp_dev *dev; - struct pnp_dev *devmpu; -#endif /* CONFIG_PNP */ }; static int snd_opti9xx_pnp_is_probed; @@ -159,12 +155,17 @@ static int snd_opti9xx_pnp_is_probed; static struct pnp_card_device_id snd_opti9xx_pnpids[] = { #ifndef OPTi93X /* OPTi 82C924 */ - { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 }, + { .id = "OPT0924", + .devs = { { "OPT0000" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0924 }, /* OPTi 82C925 */ - { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 }, + { .id = "OPT0925", + .devs = { { "OPT9250" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0925 }, #else /* OPTi 82C931/3 */ - { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 }, + { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, + .driver_data = 0x0931 }, #endif /* OPTi93X */ { .id = "" } }; @@ -207,24 +208,34 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, chip->hardware = hardware; strcpy(chip->name, snd_opti9xx_names[hardware]); - chip->mc_base_size = opti9xx_mc_size[hardware]; - spin_lock_init(&chip->lock); chip->irq = -1; +#ifndef OPTi93X +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; +#endif /* CONFIG_PNP */ + else { + chip->mc_base = 0xf8c; + chip->mc_base_size = opti9xx_mc_size[hardware]; + } +#else + chip->mc_base_size = opti9xx_mc_size[hardware]; +#endif + switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3; chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: case OPTi9XX_HW_82C925: - chip->mc_base = 0xf8c; chip->password = 0xe5; chip->pwd_reg = 3; break; @@ -292,7 +303,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, spin_unlock_irqrestore(&chip->lock, flags); return retval; } - + static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, unsigned char value) { @@ -341,7 +352,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, - long wss_base, + long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) { @@ -354,16 +365,23 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, switch (chip->hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C924: + /* opti 929 mode (?), OPL3 clock output, audio enable */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); + /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); case OPTi9XX_HW_82C925: + /* enable WSS mode */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + /* OPL3 FM synthesis */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20); + /* disable Sound Blaster IRQ and DMA */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); #ifdef CS4231 + /* cs4231/4248 fix enabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); #else + /* cs4231/4248 fix disabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02); #endif /* CS4231 */ break; @@ -411,21 +429,26 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, return -EINVAL; } - switch (wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (port & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -487,7 +510,7 @@ __skip_base: #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ @@ -729,15 +752,15 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, { struct pnp_dev *pdev; int err; + struct pnp_dev *devmpu; +#ifndef OPTi93X + struct pnp_dev *devmc; +#endif - chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - - pdev = chip->dev; - err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); @@ -750,9 +773,24 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else - if (pid->driver_data != 0x0924) - port = pnp_port_start(pdev, 1); + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + return err; + } + + port = pnp_port_start(pdev, 1); fm_port = pnp_port_start(pdev, 2) + 8; + /* + * The MC(0) is never accessed and card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); @@ -760,16 +798,16 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, dma2 = pnp_dma(pdev, 1); #endif /* CS4231 || OPTi93X */ - pdev = chip->devmpu; - if (pdev && mpu_port > 0) { - err = pnp_activate_dev(pdev); + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + + if (devmpu && mpu_port > 0) { + err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); mpu_port = -1; - chip->devmpu = NULL; } else { - mpu_port = pnp_port_start(pdev, 0); - mpu_irq = pnp_irq(pdev, 0); + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); } } return pid->driver_data; @@ -824,7 +862,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; - error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -865,10 +903,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, port + 4, irq, dma1, xdma2); + card->shortname, pcm->name, + chip->wss_base + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, port + 4, irq, dma1); + card->shortname, pcm->name, chip->wss_base + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) @@ -1062,9 +1101,6 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, snd_card_free(card); return error; } - if (hw <= OPTi9XX_HW_82C930) - chip->mc_base -= 0x80; - error = snd_opti9xx_read_check(chip); if (error) { snd_printk(KERN_ERR "OPTI chip not found\n"); -- 1.6.4 ---------------------------------------------------------------------- Codziennie 11 tys. ofert pracy Sprawdz >>> http://link.interia.pl/f25ac

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[alsa-devel] [PATCH 2/3] ALSA: usb-audio: use a format bitmask per alternate setting
by Clemens Ladisch 01 Mar '10

01 Mar '10

In preparation for USB audio 2.0 support, change the audioformat structure so that it uses a bitmask to specify possible formats. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/card.h +++ linux/sound/usb/card.h @@ -9,7 +9,7 @@ struct audioformat { struct list_head list; - snd_pcm_format_t format; /* format type */ + u64 formats; /* ALSA format bits */ unsigned int channels; /* # channels */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int frame_size; /* samples per frame for non-audio */ --- linux/sound/usb/endpoint.c +++ linux/sound/usb/endpoint.c @@ -94,7 +94,7 @@ int snd_usb_add_audio_endpoint(struct sn if (subs->endpoint == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; - subs->formats |= 1ULL << fp->format; + subs->formats |= fp->formats; return 0; } } @@ -268,7 +268,7 @@ int snd_usb_parse_audio_endpoints(struct */ if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && - fp->format == SNDRV_PCM_FORMAT_S16_LE && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) --- linux/sound/usb/format.c +++ linux/sound/usb/format.c @@ -319,7 +319,7 @@ static int parse_audio_format_i(struct s return -1; } - fp->format = pcm_format; + fp->formats = 1uLL << pcm_format; /* gather possible sample rates */ /* audio class v1 reports possible sample rates as part of the @@ -361,16 +361,16 @@ static int parse_audio_format_ii(struct switch (format) { case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ - // fp->format = SNDRV_PCM_FORMAT_AC3; - fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ + // fp->formats = SNDRV_PCM_FMTBIT_AC3; + fp->formats = SNDRV_PCM_FMTBIT_U8; /* temporary hack to receive byte streams */ break; case UAC_FORMAT_TYPE_II_MPEG: - fp->format = SNDRV_PCM_FORMAT_MPEG; + fp->formats = SNDRV_PCM_FMTBIT_MPEG; break; default: snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); - fp->format = SNDRV_PCM_FORMAT_MPEG; + fp->formats = SNDRV_PCM_FMTBIT_MPEG; break; } --- linux/sound/usb/pcm.c +++ linux/sound/usb/pcm.c @@ -58,7 +58,9 @@ static struct audioformat *find_format(s list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (fp->format != format || fp->channels != channels) + if (!(fp->formats & (1uLL << format))) + continue; + if (fp->channels != channels) continue; if (rate < fp->rate_min || rate > fp->rate_max) continue; @@ -428,10 +430,15 @@ static int hw_check_valid_format(struct struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + struct snd_mask check_fmts; unsigned int ptime; /* check the format */ - if (!snd_mask_test(fmts, fp->format)) { + snd_mask_none(&check_fmts); + check_fmts.bits[0] = (u32)fp->formats; + check_fmts.bits[1] = (u32)(fp->formats >> 32); + snd_mask_intersect(&check_fmts, fmts); + if (snd_mask_empty(&check_fmts)) { hwc_debug(" > check: no supported format %d\n", fp->format); return 0; } @@ -584,7 +591,7 @@ static int hw_rule_format(struct snd_pcm fp = list_entry(p, struct audioformat, list); if (!hw_check_valid_format(subs, params, fp)) continue; - fbits |= (1ULL << fp->format); + fbits |= fp->formats; } oldbits[0] = fmt->bits[0]; --- linux/sound/usb/proc.c +++ linux/sound/usb/proc.c @@ -79,11 +79,16 @@ static void proc_dump_substream_formats( list_for_each(p, &subs->fmt_list) { struct audioformat *fp; + snd_pcm_format_t fmt; fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %s\n", - snd_pcm_format_name(fp->format)); + snd_iprintf(buffer, " Format:"); + for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + if (fp->formats & (1uLL << fmt)) + snd_iprintf(buffer, " %s", + snd_pcm_format_name(fmt)); + snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, --- linux/sound/usb/quirks-table.h +++ linux/sound/usb/quirks-table.h @@ -279,7 +279,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 4, .iface = 0, .altsetting = 1, @@ -296,7 +296,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -580,7 +580,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, @@ -597,7 +597,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -793,7 +793,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -810,7 +810,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 2, .altsetting = 1, --- linux/sound/usb/quirks.c +++ linux/sound/usb/quirks.c @@ -174,7 +174,7 @@ static int create_uaxx_quirk(struct snd_ const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, --- linux/sound/usb/urb.c +++ linux/sound/usb/urb.c @@ -662,7 +662,7 @@ static int prepare_nodata_playback_urb(s urb->number_of_packets = ctx->packets; urb->transfer_buffer_length = offs * stride; memset(urb->transfer_buffer, - subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, + runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, offs * stride); return 0; } @@ -924,7 +924,7 @@ void snd_usb_init_substream(struct snd_u snd_usb_set_pcm_ops(as->pcm, stream); list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= 1ULL << fp->format; + subs->formats |= fp->formats; subs->endpoint = fp->endpoint; subs->num_formats++; subs->fmt_type = fp->fmt_type;

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