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[alsa-devel] [PATCH] ALSA SOC driver for s3c24xx: 8 bit sound fix
by Christian Pellegrin 10 Nov '08

10 Nov '08

fixes playing/recording of 8 bit audio files. Generated on 20081108 against v2.6.27 Signed-off-by: Christian Pellegrin <chripell(a)fsfe.org> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; + default: + return -EINVAL; } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); -- 1.4.4.4

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[alsa-devel] [PATCH] gusextreme: Fix build errors
by Ville Syrjala 10 Nov '08

10 Nov '08

gusextreme depends on opl3 support. Add the approriate select to Kconfig. Also remove the unnecessary hwdep select. Relevant build errors: ERROR: "snd_opl3_hwdep_new" [sound/isa/gus/snd-gusextreme.ko] undefined! ERROR: "snd_opl3_create" [sound/isa/gus/snd-gusextreme.ko] undefined! Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- sound/isa/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb4..ce0aa04 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -211,7 +211,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_HWDEP + select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help -- 1.5.6.4

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[alsa-devel] [PATCH] Renamed the struct soc_enum field .mask to .count. The field refers to the number of enumerable entries instead of being a bit mask. This change makes the interface a more clear.
by Marc Singer 10 Nov '08

10 Nov '08

Signed-off-by: Marc Singer <elf(a)buici.com> --- include/sound/soc-codec.h | 14 +++++++------- sound/soc/soc-core.c | 20 ++++++++++---------- sound/soc/soc-dapm.c | 14 +++++++------- 3 files changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index d8595ef..d59015a 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -79,13 +79,13 @@ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ .private_value = (reg_left) | ((shift) << 8) | \ ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) } -#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ +#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xcount, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .texts = xtexts } -#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \ - SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts) -#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \ -{ .mask = xmask, .texts = xtexts } + .count = xcount, .texts = xtexts } +#define SOC_ENUM_SINGLE(xreg, xshift, xcount, xtexts) \ + SOC_ENUM_DOUBLE(xreg, xshift, xshift, xcount, xtexts) +#define SOC_ENUM_SINGLE_EXT(xcount, xtexts) \ +{ .count = xcount, .texts = xtexts } #define SOC_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ .info = snd_soc_info_enum_double, \ @@ -180,7 +180,7 @@ struct soc_enum { unsigned short reg2; unsigned char shift_l; unsigned char shift_r; - unsigned int mask; + unsigned int count; /* Count of options */ const char **texts; void *dapm; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 706f98d..b8efb4b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1262,10 +1262,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->count; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->count - 1) + uinfo->value.enumerated.item = e->count - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; @@ -1288,7 +1288,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); @@ -1317,14 +1317,14 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short val; unsigned short mask, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->count - 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->count - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r; @@ -1351,10 +1351,10 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = e->mask; + uinfo->value.enumerated.items = e->count; - if (uinfo->value.enumerated.item > e->mask - 1) - uinfo->value.enumerated.item = e->mask - 1; + if (uinfo->value.enumerated.item > e->count - 1) + uinfo->value.enumerated.item = e->count - 1; strcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item]); return 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f203693..5b83329 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -119,13 +119,13 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; int val, item, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(w->codec, e->reg); item = (val >> e->shift_l) & (bitmask - 1); p->connect = 0; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->count; i++) { if (!(strcmp(p->name, e->texts[i])) && item == i) p->connect = 1; } @@ -162,7 +162,7 @@ static int dapm_connect_mux(struct snd_soc_card *soc_card, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int i; - for (i = 0; i < e->mask; i++) { + for (i = 0; i < e->count; i++) { if (!(strcmp(control_name, e->texts[i]))) { list_add(&path->list, &soc_card->dapm_paths); list_add(&path->list_sink, &dest->sources); @@ -1197,7 +1197,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, bitmask; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; val = snd_soc_read(widget->codec, e->reg); ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); @@ -1227,15 +1227,15 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; int ret = 0; - for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) + for (bitmask = 1; bitmask < e->count; bitmask <<= 1) ; - if (ucontrol->value.enumerated.item[0] > e->mask - 1) + if (ucontrol->value.enumerated.item[0] > e->count - 1) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; mask = (bitmask - 1) << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->mask - 1) + if (ucontrol->value.enumerated.item[1] > e->count - 1) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= (bitmask - 1) << e->shift_r; -- 1.5.6.5

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[alsa-devel] Non-functional Altec Lansing FX5051 USB-soundcard + speakers
by Jeremy Nickurak 10 Nov '08

10 Nov '08

About 14 months ago I purchased a nice new ~$300 set of altec lansing 5.1 speakers, which include a USB-sound card. Since that time they've largely gone unused, or at least not properly used. I brought this up on the mailing list at the time, but didn't get much response, so I thought I'd poke my head up again and see if anybody has any insight. First off, the set works correctly, out-of-the-box under windows, without having to install any special drivers or software at all. Under linux, plugging them in instantly freezes the usb sound card up, and locks it at 100% volume, in such a way that there's absolutely no way to turn the volume down: not from alsamixer, and not from the speaker's hardware control dial. Sound does play, but again, it's unbearably loud, to the point where I don't do it for risk of damaging my speakers and/or ears. I filed the bug away on the bug-tracker, but haven't heard much out of it since. The bug contains my lsusb -v information, the "contents" output from amixer for this card, and some preliminary results showing some ability to tweak the volume with amixer, but not reliably. Any suggestions are welcome. I'm happy to experiment with this set more if it can help, run experimental code, etc. Heck, I'd go at the thing with a screwdriver for chipset information if I know what I was looking for or thought it could help.

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[alsa-devel] [PATCH] ALSA: HDSP: check for io box before uploading firmware
by Tim Blechmann 10 Nov '08

10 Nov '08

currently the hdsp driver tries to upload the firmware, even if the io box is not connected. this patch adds a check for the io box before trying to upload the firmware. thus instead of messages complaining about the fifo status and firmware loading failure, the driver gives a message that no multiface or digiface is connected. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543..cf9eb41 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5045,6 +5045,9 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); + if ((err = hdsp_check_for_iobox(hdsp)) < 0) + return err; + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { #ifdef HDSP_FW_LOADER if ((err = hdsp_request_fw_loader(hdsp)) < 0) @@ -5057,7 +5060,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* init is complete, we return */ return 0; #endif - /* no iobox connected, we defer initialization */ + /* we defer initialization */ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n"); if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; -- 1.5.6.3

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[alsa-devel] Stuttering 100mA and 500mA MaxPower usb audio investigation on bus powered and external powererd usb hubs
by Jelle de Jong 10 Nov '08

10 Nov '08

Hello everybody, I have been testing several scenarios to find some glues why some usb audio devices did not fully work and gave stuttering sounds I found the following: usb devices define a max power attribute during there enumeration process. I have some usb devices that have this at 500mA and some that have this on 100mA. I also have some usb hubs that have external extra power supply and some that are only a hub with bus power. Here it comes: Make sure the 4 port usb hub is full with for example: - usb keyboard (100mA) - usb mouse (100mA) - usb audio (100mA or 500mA) - usb stick (200mA/500mA) - usb hub with external power (500mA MaxPower) (AC adapter delivers 1A) - usb hub without external power (100mA MaxPower) Make sure to use audio with several tones low tones (most dvds will do) When connecting a 100mA usb audio devices on a hub with external power it will give stuttering sound!! (most usb audo devices are 100mA and most hubs are external powered due to more then one usb stick) When connecting a 500mA usb audio device on a hub with external power it will work perfectly. When connecting a 100mA or a 500mA usb audio device on a hub without external power it will work perfectly. I have the idea there is something wrong with the power distribution system when using external powered usb hubs... Any ideas how to correctly address this problem/bug? Best regards, Jelle

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[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Romain 10 Nov '08

10 Nov '08

Hi, I've tested the sbxfi driver and it works great on my setup. Had little problems before getting it running : i had to comment out the snd_printk stuff, otherwise the module wouldn't load (snd_printk* symbols not found, in dmesg) This was probably not the cleanest way to deal but it's just debug messages and i'm new wrt kernel driver hacking. I also had to disable my onboard audio in BIOS, otherwise the X-Fi would not get picked up. i'm using Fedora 9 with Linux 2.6.26.6-79.fc9.i686 kernel (standard supply from fedora repos),and pulseaudio I'm using with standard 2.1 jack output, i do not yet own home cinema speakers so i couldn't test all the cabling options. Thanks for this great work, Best Regards, Romain.

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[alsa-devel] [PATCH] ALSA: hda - Add another HP model (6730s) for AD1884A.
by Michel Marti 09 Nov '08

09 Nov '08

Signed-off-by: Michel Marti <mma(a)objectxp.com> --- sound/pci/hda/patch_analog.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d3fd432..ee62e31 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3861,6 +3861,7 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: hdsp: check for iobox and upload firmware during ioctl
by Tim Blechmann 09 Nov '08

09 Nov '08

currently, the error message when trying to run hdspmixer or hdspconf if the breakout box is not connected is somehow misleading, since it asks the user to upload the firmware. this patch adds a test, whether the breakout box is connected and tries to upload the firmware in the case, that it is not present, e.g. because of power failures of the breakout box. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 17 +++++++++++++---- 1 files changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cf9eb41..5bd0be6 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4548,11 +4548,18 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne { struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; + int err; switch (cmd) { case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: { struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg; + if ((err = hdsp_check_for_iobox (hdsp)) < 0) + return err; + + if ((err = hdsp_check_for_firmware(hdsp, 1)) < 0) + return err; + if (!(hdsp->state & HDSP_FirmwareLoaded)) { snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n"); return -EINVAL; @@ -4572,10 +4579,12 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne unsigned long flags; int i; - if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); - return -EINVAL; - } + if ((err = hdsp_check_for_iobox (hdsp)) < 0) + return err; + + if ((err = hdsp_check_for_firmware(hdsp, 1)) < 0) + return err; + spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); -- 1.5.6.3

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[alsa-devel] Intel G45 HDMI audio - missing samples
by Shane W 08 Nov '08

08 Nov '08

Hi, One issue with Intel audio over HDMI on a DG45-ID motherboard. When playing a short sample via HDMI, it misses the first quarter second or so of the audio. This is particularly noticable when using text to speech as the first word is lost. I can get around this by having a seperate program hold the audio device open while using text to speech, that seems to get the whole sample but it's a cludgy workaround. The same issue does not exist on the optical port. Best, Shane

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