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Re: [alsa-devel] [PATCH 2.6.26.2] hda_intel: ALSA HD Audio patch for Intel Ibex Peak DeviceIDs
by Takashi Iwai 11 Aug '08

11 Aug '08

At Fri, 8 Aug 2008 15:56:39 -0700, Seth Heasley wrote: > > This patch adds the Intel Ibex Peak (PCH) HD Audio Controller DeviceIDs. > > Signed-off by: Seth Heasley <seth.heasley(a)intel.com> Your embedded patch couldn't be applied properly, maybe because of broken spaces. Check your MUA setting. Anyway, I applied it manually. Thanks! Takashi > > --- linux-2.6.26.2/sound/pci/hda/hda_intel.c.orig 2008-08-08 11:25:05.000000000 -0700 > +++ linux-2.6.26.2/sound/pci/hda/hda_intel.c 2008-08-08 11:27:24.000000000 -0700 > @@ -98,6 +98,7 @@ > "{Intel, ICH8}," > "{Intel, ICH9}," > "{Intel, ICH10}," > + "{Intel, PCH}," > "{Intel, SCH}," > "{ATI, SB450}," > "{ATI, SB600}," > @@ -2181,6 +2182,8 @@ > { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, > { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, > + /* PCH */ > + { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* ATI SB 450/600 */ > -- > 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] powerpc/drivers: use linux/of_device.h instead of asm/of_device.h
by Stephen Rothwell 11 Aug '08

11 Aug '08

Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- drivers/hwmon/ams/ams.h | 2 +- sound/aoa/soundbus/soundbus.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) These are the last two users of asm/of_device.h. If everyone is happy, we can put this through the powerpc tree. I have build tested for ppc64_defconfig and ppc6xx_defconfig (which uses the relevant drivers). After this we can remove the include of linux/of_device.h from asm/of_device.h diff --git a/drivers/hwmon/ams/ams.h b/drivers/hwmon/ams/ams.h index a6221e5..221ef69 100644 --- a/drivers/hwmon/ams/ams.h +++ b/drivers/hwmon/ams/ams.h @@ -4,7 +4,7 @@ #include <linux/mutex.h> #include <linux/spinlock.h> #include <linux/types.h> -#include <asm/of_device.h> +#include <linux/of_device.h> enum ams_irq { AMS_IRQ_FREEFALL = 0x01, diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37..a0f223c 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include <asm/of_device.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <linux/list.h> -- 1.5.6.3 -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au http://www.canb.auug.org.au/~sfr/

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[alsa-devel] clicking/noise when upsampling with ESI juli@
by Amaury De Ganseman 10 Aug '08

10 Aug '08

Hi, There's a problem when I start to play a file the sound is bad during 1sec or less (clicking,cracking,...) or when I switch between tracks I hear clicks. Any idea ? Note: It works very well with OSS driver (last mercurial) Thanks for help. Here my asoundrc: defaults.pcm.rate_converter "speexrate_best" pcm.!default { type plug slave.pcm { type dmix ipc_key 1478 slave { pcm "hw:0,1" format S32_LE #period_time 0 period_size 2048 buffer_size 8192 rate 48000 } } #rate_converter "speexrate_best" #rate_converter "samplerate_best" }

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[alsa-devel] [PATCH] wm8750: it's MONO1, not MONO
by Dmitry Baryshkov 09 Aug '08

09 Aug '08

Since first commit wm8750 contained output named MONO, but all routes mentioned MONO1. Correct MONO to be MONO1. Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-ac97-lib.c | 9 ++++----- sound/soc/codecs/wm8750.c | 2 +- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index be500d3..3802d68 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -8,11 +8,10 @@ #include <sound/pxa2xx-lib.h> #include <asm/irq.h> -#include <asm/hardware.h> - -#include <asm/arch/pxa-regs.h> -#include <asm/arch/pxa2xx-gpio.h> -#include <asm/arch/audio.h> +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/pxa2xx-gpio.h> +#include <mach/audio.h> static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e23cb09..2e71394 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -348,7 +348,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("ROUT2"), - SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), SND_SOC_DAPM_INPUT("LINPUT1"), -- 1.5.6.3

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[alsa-devel] ALSA OSS Emulation - Read - EAGAIN Errors
by Rahul Iyer 09 Aug '08

09 Aug '08

Hi ALSA Developers I have a full duplex voice application, that records voice and sends (over a network) as well as receives and plays voice. OSS APIs are used to capture and play voice samples. The voice quality is noticeably poor in SUSE 10. After further investigation, I see most of the 'read' system calls fail with EAGAIN (resource unavailable) error. This is not the case in SUSE 9.2. This results in poor voice quality -- broken voice, due to loss of voice samples. Please take a look at the strace stats below for SUSE 9.2 and 10. Check the errors column for SUSE 10. SUSE 9.2 Process 17533 detached % time seconds usecs/call calls errors syscall ------ ----------- ----------- --------- --------- ---------------- 51.54 0.824285 48 17152 gettimeofday 14.36 0.229609 52 4393 693 select 9.85 0.157454 76 2081 692 sigreturn 7.89 0.126199 14 8784 rt_sigprocmask 7.74 0.123765 20 6201 ioctl 4.31 0.068929 33 2067 read 4.31 0.068898 33 2067 write 0.01 0.000083 6 14 alarm 0.00 0.000034 11 3 munmap 0.00 0.000007 4 2 rt_sigaction 0.00 0.000003 3 1 1 open ------ ----------- ----------- --------- --------- ---------------- 100.00 1.599266 42765 1386 total SUSE 10 Process 11342 detached % time seconds usecs/call calls errors syscall ------ ----------- ----------- --------- --------- ---------------- 49.58 1.006722 1 1191090 1189103 read 49.44 1.003873 1 1194607 ioctl 0.95 0.019285 11 1698 write 0.02 0.000453 0 1152 551 select 0.00 0.000038 13 3 munmap 0.00 0.000015 0 9230 gettimeofday 0.00 0.000000 0 1 1 open 0.00 0.000000 0 11 alarm 0.00 0.000000 0 0.00 0.000000 0 793 624 sigreturn 0.00 0.000000 0 6 sched_yield 0.00 0.000000 0 2 rt_sigaction 0.00 0.000000 0 2303 rt_sigprocmask ------ ----------- ----------- --------- --------- ---------------- 100.00 2.030386 2400896 1190279 total BTW, the SUSE 10 machine runs on newer hardware with on board Intel HDA Sound (Realtek ALC262). Any thoughts on what causes so many read failures? Could this be hardware/driver related? There is nothing fancy about this sound card, its a regular on board card. It would be great if someone can confirm their experience with this card. Any suggestions are also most welcome... Thanks for the help.-- Rahul Iyer

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[alsa-devel] Suggestion for http://git.alsa-project.org/
by Timur Tabi 08 Aug '08

08 Aug '08

If you go to http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git, you will see this text: description Sound development tree owner Takashi Iwai last change Fri, 8 Aug 2008 14:50:58 +0000 URL git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git http://www.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git This is very helpful information. It makes git-clone much easier, because I can just copy/psate the URL. Can we have the same thing at http://git.alsa-project.org/? I don't want to have to keep figuring out the http URL for cloning alsa-lib, every time I need to do it. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH] ALSA: wss_lib: missing closing brace in (ifdeffed out) debug function.
by Rene Herman 08 Aug '08

08 Aug '08

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[PATCH] ALSA: wss_lib: missing closing brace in (ifdeffed out) debug function.
by Rene Herman 08 Aug '08

08 Aug '08

Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/wss/wss_lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 011da7a..6b7a0fc 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -282,7 +282,7 @@ static void snd_wss_debug(struct snd_wss *chip) printk(KERN_DEBUG "CS4231 REGS: INDEX = 0x%02x " " STATUS = 0x%02x\n", - wss_inb(chip, CS4231P(REGSEL), + wss_inb(chip, CS4231P(REGSEL)), wss_inb(chip, CS4231P(STATUS))); printk(KERN_DEBUG " 0x00: left input = 0x%02x " -- 1.5.5

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Re: [alsa-devel] [PATCH 03/11] wss_lib: rename cs4321_foo to wss_foo
by Rene Herman 08 Aug '08

08 Aug '08

On 31-07-08 21:03, Krzysztof Helt wrote: Takashi, you already applied these but is it possible to still get my Acked-by (or Reviewed-by, whatever you want) on them? When a problem is bisected to one of these, people tend to put the people in the patch tags in CC and I looked at them and would like to be informed of any trouble. I tested most users by now, and we'll get the azt2320 fixed before this hits the streets but I still like to be in the loop on any traffic concerning them. Acked-by: Rene Herman <rene.herman(a)gmail.com> (or Reviewed-by, Tested-by, ...) Below comments not important: > @@ -679,23 +679,25 @@ static int __devinit snd_opl3sa2_probe(s [ ... ] > - chip->cs4231 = cs4231; > - if ((err = snd_cs4231_pcm(cs4231, 0, NULL)) < 0) > + chip->wss = wss; > + err = snd_wss_pcm(wss, 0, NULL); > + if (err < 0) > return err; > - if ((err = snd_cs4231_mixer(cs4231)) < 0) > + err = snd_wss_mixer(wss); > + if (err < 0) > return err; > if ((err = snd_opl3sa2_mixer(chip)) < 0) > return err; <mumble> > - if ((err = snd_cs4231_timer(cs4231, 0, NULL)) < 0) > + err = snd_wss_timer(wss, 0, NULL); > + if (err < 0) > return err; [ ... ] > +static void snd_wss_debug(struct snd_wss *chip) > + printk(KERN_DEBUG > + " 0x00: left input = 0x%02x " > + " 0x10: alt 1 (CFIG 2) = 0x%02x\n", > + snd_wss_in(chip, 0x00), > + snd_wss_in(chip, 0x10)); > + printk(KERN_DEBUG > + " 0x01: right input = 0x%02x " > + " 0x11: alt 2 (CFIG 3) = 0x%02x\n", > + snd_wss_in(chip, 0x01), > + snd_wss_in(chip, 0x11)); I'll admit that's actually half-way passable... > @@ -553,79 +628,88 @@ static void snd_wss_playback_format(s [ ... ] > if (full_calib) { > - snd_cs4231_mce_up(chip); > + snd_wss_mce_up(chip); > spin_lock_irqsave(&chip->reg_lock, flags); > - if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { > - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, > - (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? > - (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : > - pdfr); > + if (chip->hardware != WSS_HW_INTERWAVE && !chip->single_dma) { > + if (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) > + pdfr = (pdfr & 0xf0) | > + (chip->image[CS4231_REC_FORMAT] & 0x0f); > } else { > - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); > + chip->image[CS4231_PLAYBK_FORMAT] = pdfr; > } > + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, pdfr); > spin_unlock_irqrestore(&chip->reg_lock, flags); Much beter. > +static void snd_wss_capture_format(struct snd_wss *chip, [ ... ] > - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); > - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); > - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); > + snd_wss_out(chip, CS4231_ALT_FEATURE_1, > + chip->image[CS4231_ALT_FEATURE_1] | 0x20); > + snd_wss_out(chip, CS4231_REC_FORMAT, > + chip->image[CS4231_REC_FORMAT] = cdfr); Not much better... > + snd_wss_out(chip, CS4231_ALT_FEATURE_1, > + chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); Same level of betterness. Rene.

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[alsa-devel] Application hangs with different period sizes
by anuj aggarwal 08 Aug '08

08 Aug '08

My audio driver supports buffer sizes from 128 to 32768 and period sizes from 8 to 2048. I am trying to set the period size as 1030 (just a random number between 8 & 2048, but not multiple of 2^n) and buffer size as 16*buffer_size (i.e. 16480). The output is as follows: Buffer size range from 128 to 32768 Period size range from 8 to 2048 Request period size 1030 and got 1030 Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16480 period_size : 1030 period_time : 23356 tstamp_mode : NONE period_step : 1 avail_min : 1030 start_threshold : 16480 stop_threshold : 16480 silence_threshold: 0 silence_size : 0 boundary : 1080033280 The problem with this setup is the application just hangs without playing anything. If I use period size as 2048, the app plays the song but clips the last part of it. If I use 1024/512, it works fine. I have tried alsa lib version 1.0.15 & 1.0.16 but the problem persists. Is there anything wrong with my app or audio driver? Please help. Thanks in advance, Anuj Aggarwal

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