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Re: [alsa-devel] [PATCH 2/3] ASoC: DaVinci: pcm, rename variables in prep for ping/pong
by Nori, Sekhar 09 Sep '09

09 Sep '09

On Wed, Sep 09, 2009 at 18:03:58, Caglar Akyuz wrote: > On Wednesday 09 September 2009 15:08:42 Nori, Sekhar wrote: [...] > > > > However, with or without your patch, I noticed a segmentation fault for the > > first time the 'arecord | aplay' loop is created. There is no fault the > > second time around. I haven't spent time debugging this yet. > > > > This is due to these lines in davinci_pcm_enqueue_dma: > > data_type = prtd->params->data_type; > count = period_size / data_type; > > First time running data_type is set to 0. This is a long time issue, I > remember modifying my kernels to check for zero before this division.. Ah, thanks! Planning to submit a fix for this? We can work on the patch if you are busy otherwise. Regards, Sekhar

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[alsa-devel] Layla 3G does not recover from ACPI Suspend
by Mark Hills 09 Sep '09

09 Sep '09

I have an Echo Layla 3G in my workstation. It works for audio, but does not recover from ACPI suspend to RAM. On recovery the system is fine, and the Layla exists in /proc/asound/cards. But when the Layla is used it prints this message to dmesg, multiple times: wait_handshake(): Timeout waiting for DSP Here are the relevant dmesg lines after awakening: Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0xf (was 0x100, writing 0x104) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x4 (was 0x0, writing 0xe7d00000) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x3 (was 0x0, writing 0xc010) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x1 (was 0x2800000, writing 0x2800112) Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 I couldn't see anything loading the firmware. I thought unloading and loading snd-echo3g after recovery would help. This shows the firmware being loaded, but then no ALSA device is shown in /proc/asound/cards. On loading snd-echo3g: Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/echo3g_dsp.fw Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/loader_dsp.fw Echoaudio Echo3G 0000:03:03.0: PCI INT A disabled Echoaudio Echo3G: probe of 0000:03:03.0 failed with error -5 Is the firmware not being loaded when it should be? Or is there some extra initialisation not being done (eg. init_hw() in echo3g_dsp.c)? All other devices in the system recover fine, but as far as I'm aware no other PCI devices require firmware. It's a Dell x86 system, kernel 2.6.31-rc7. Thanks for any help, -- Mark

3 3

[alsa-devel] [PATCH v2] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- The previous patch missed a git add for psc.h, sorry! sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..3f474e8 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 -- 1.6.4.2

3 3

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 87831cb660954356d68cebdb1406f3be09e784e9: Mark Brown (1): ASoC: Fix WM835x Out4 capture enumeration are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ 1 files changed, 96 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); -- 1.6.4.2

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[alsa-devel] Update to problem with Creative X-fi mixer problems
by Susan Cragin 08 Sep '09

08 Sep '09

I saw that someone was going to try to get all the mixer controls to work properly. I updated and opened alsamixer this morning, and found that both the incoming/recording controls were there! (I should have quit while I was ahead.) Then I re-booted and recompiled wine and opened winecfg and found that oss could find the proper mixer but alsa did not. (The oss mixer is listed as Creative CA0110-IBG) ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave fixme:mixer:ALSA_MixerInit No master control found on HD-Audio Generic, disabling mixer susan@ubuntu:~$ alsamixer No mixer elems found

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Re: [alsa-devel] snd_dummy on Centos - Redhat
by Takashi Iwai 08 Sep '09

08 Sep '09

At Fri, 4 Sep 2009 00:43:09 +0200, Giovanni Maruzzelli wrote: > > On Fri, Sep 4, 2009 at 12:03 AM, Giovanni > Maruzzelli<gmaruzz.lists(a)gmail.com> wrote: > > I'll test it and report back here. > > Still we have big problems.... OK, then upgrade your alsa driver stack to 1.0.21. There have been some changes in PCM core. Then, get dummy.c again from the latest alsa-driver-unstable tarball. It already contains my latest fix. If 1.0.21+dummy.c doesn't work, apply the patch below to get position logs. If this still causes the invalid backward jump to position 0, you should check xrun debug, e.g. # echo 9 > /proc/asound/card0/pcm0p/xrun_debug (assuming snd-dummy is card0). Takashi --- diff --git a/dummy.c b/dummy.c index e8e29bf..f7c21d5 100644 --- a/dummy.c +++ b/dummy.c @@ -239,6 +239,10 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) dpcm->frac_period_rest += dpcm->frac_period_size; } dpcm->frac_period_rest -= delta; + printk(KERN_DEBUG "xxx delta=%ld, pos=%d(%d), rest=%d(%d)\n", + delta / dpcm->rate, + dpcm->frac_pos, dpcm->frac_pos / HZ, + dpcm->frac_period_rest, dpcm->frac_period_rest / HZ); } static int dummy_systimer_start(struct snd_pcm_substream *substream) @@ -248,6 +252,10 @@ static int dummy_systimer_start(struct snd_pcm_substream *substream) dpcm->base_time = jiffies; dummy_systimer_rearm(dpcm); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** START: buffer=%d, period=%d, rate=%d\n", + (int)substream->runtime->buffer_size, + (int)substream->runtime->period_size, + substream->runtime->rate); return 0; } @@ -257,6 +265,7 @@ static int dummy_systimer_stop(struct snd_pcm_substream *substream) spin_lock(&dpcm->lock); del_timer(&dpcm->timer); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** STOP\n"); return 0; }

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[alsa-devel] Audio cutoff when playing using dmix
by ruchi sirauthiya 08 Sep '09

08 Sep '09

Hi, I am facing audio clipping issues when dmix is used for playback. For the test we used a full scale sine wave tone generated using software and encoded as "Signed 16 bit Little Endian, Rate 44100 Hz, Mono". The audio is getting cut out at a 2-3 second interval during playback. This happens only for a fraction of second but it is consistent. This does not happen when audio is played back on the device directly but only when going through dmix. We have observed that there were differences in audio cutting off when the sampling rate was set to 8000Hz and 32000 Hz in asound.conf. The contents of asound.conf is pasted below for your reference: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { pcm hw:0 rate 8000 channels 2 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.rateConvert { type plug slave.pcm "asymed" } The configuration given above works fine with dmix in 2.6.29 kernel, I am using kernel 2.6.22.18. I have compared the configuration differences (buffers and periods) with and without using dmix. I tried applying some parameters (like buffer_size, period_time and period_size) which was in use when audio was played directly to the hardware device by aplay using the following command: aplay -D rateConvert --period-time=92879 --buffer-size=65536 --period-size=4096 /mnt/test.wav -vvv -d 5 There was come improvement in the audio cutting off issue but it was not eliminated completely. Given below are the logs for aplay (-vvv) with and without using dmix (with audio cutoff issue on kernel 2.6.22): Without using dmix: [root@OMAP3 tests]# aplay /mnt/test.wav -vvv -d 5 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono 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 : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 65536 period_size : 4096 period_time : 92879 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 65536 stop_threshold : 65536 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% ...Same text repeats 50 times... Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% Using dmix: [root@OMAP3 tests]# aplay -D rateConvert /mnt/test.wav -vvv -d 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5512 xfer_align : 5512 start_threshold : 22048 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 16000 period_size : 1000 period_time : 125000 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 2097152000 silence_threshold: 0 silence_size : 2097152000 boundary : 2097152000 Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% ...Same text repeats 36 times... Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00005f1a ############### 74% Please help me fix this issue. Does this occur due to non-optimal settings of the parameters (buffers, periods) in the configuration or is it some other issue with the ALSA library or the kernel? Please ask if any more setup/configuration settings are required for reference. Thanks and Regards, Ruchi Sirauthiya

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 367da1527aff2a4f789d03082f9fb191cdf332c4: Robert Schwebel (1): ASoC: fix pxa2xx-ac97.c breakage are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (2): ASoC: Remove unuused hw_read_t ASoC: Fix WM835x Out4 capture enumeration include/sound/soc.h | 1 - sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 1 insertions(+), 2 deletions(-)

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Re: [alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Raymond Yau 08 Sep '09

08 Sep '09

2009/9/7 Manuel Jander <manuel.jander(a)gmail.com> > Hi Raymond, > > On Mon, Sep 7, 2009 at 3:47 AM, Raymond Yau<superquad.vortex2(a)gmail.com> > wrote: > > On Thursday 03 September 2009 21:10, Manuel Jander wrote: > >> does anybody still have a AU88x0 card ? I know they are quite ancient > >> now, but if anyone could give the attached patch a try ? Would be > >> nice. > >> I made the patch against linux-2.6.30.4 > >> http://micro.homelinux.net/~mjander/au88x0.diff.gz<http://micro.homelinux.net/%7Emjander/au88x0.diff.gz>< > http://micro.homelinux.net/%7Emjander/au88x0.diff.gz> > >> > >> Best Regards, > >> Manuel > > > > Which version of audaicous (1.5.x or 2.1) are you testing ? > > (alsa-gapless or alsa-ng plugin) > > > > au88x0 most likely cannot support 1 sample per period because of the > > FIFO as described in patent 6167465. > > Well, that is clear to me now. Using a DMA transfer that is smaller > than one audio sample is not really a good idea. > I would be better off using PIO instead of DMA. > > > if you look at the source code of alsa-ng and alsa-gapless plugin > > both of them use the own way to calculate the buffer size instead of > > using snd_pcm_hw_params_get_buffer_size() > > > > > > > http://hg.atheme.org/audacious-plugins/file/2c0a9bba77d7/src/alsa-ng/alsa-c… > > > > CHECK_FAIL (snd_pcm_hw_params_set_access (pcm_handle, hwparams, > > SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"); > > CHECK_FAIL (snd_pcm_hw_params_set_format (pcm_handle, hwparams, afmt), > > "snd_pcm_hw_params_set_format"); > > CHECK_FAIL (snd_pcm_hw_params_set_channels (pcm_handle, hwparams, nch), > > "snd_pcm_hw_params_set_channels"); > > CHECK_FAIL (snd_pcm_hw_params_set_rate (pcm_handle, hwparams, rate, 0), > > "snd_pcm_hw_params_set_rate"); > > CHECK_FAIL (snd_pcm_hw_params (pcm_handle, hwparams), > "snd_pcm_hw_params"); > > bitwidth = snd_pcm_format_physical_width(afmt); > > bps = (rate * bitwidth * nch) >> 3; > > buf_size = MAX(aud_cfg->output_buffer_size, 100); > > ringbuf_size = buf_size * bps / 1000; > > I increased the minimum period bytes value to 256 in that patch. > Audacious version 2 (Debian SID) > works for me. > well, audacious.i386 1.5.1-9.fc10 cannot play any music (alsa plugin) on my au8830 with your patch, it only play music with (oss plugin) Use xmms instead of audacious since they have similar UI 256 seem much bigger than the figures mentioned in the patent , may be it is the practical limit on your VIA ITX In one embodiment of the present invention, the FIFO buffer is 32 samples or words in depth, which would withstand a maximum bus latency of about 4 microseconds. By way of example, if 48 DMA channels trigger simultaneously, with a trigger level set at half the buffer size (16 samples), each DMA channel can wait 16 samples (20 microseconds per sample), or approximately 320 microseconds before running out of data. By way of example, CPU 210 may update-sub-buffer tables 380(0)-380(x) routinely every 10 milliseconds. The periodic checks may be initiated by audio processing peripheral 230 by way of interrupts, or may be scheduled independently by the driver > > How do you get the "ADB Ctrl failure" error ? > > I do not know how to reproduce that. Sometimes it happens, and its not > possible > to recover without a hardware reset (unload and load module). Somehow > it semes to > happen when my computer gets really hot (watching an MPEG-4 video, or > compiling the kernel...). > Its a old VIA ITX board, with a 1.3GHz processor (on demand cpufreq). > > The patch tries to reset the audio card if all PCM streams are > released once such a > condition was detected. This is the situation were the windows driver > used to crash the > whole machine. > it work quite well on my two motherbroad using intel chipset > > There are two main problem of the au88x0 driver > > 1) alsa is unable to calculate the correct value for the three > > functions snd_pcm_hw_params_set_buffer_time_near() , > > snd_pcm_hw_params_set_period_time_near() and snd_pcm_set_params() > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1724 > > > > Test case : > > > > alsa-lib/test/pcm_min.c ( change default to hw:0,0 ) > > > > Real world case: > > > > gstreamer and winealsa.drv > > I will try that. I had no idea about that bugs. > A simple workaround is add another constraint snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096 ); however winealsa.drv will be using a larger period bytes than wineoss.drv > > > 2) unable to recover when the number of available src is not enough > > (FATAL: ResManager: resource type 1 exhaust) > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2228 > > > > test case : > > > > running application to use 15 src ( 7 stereo and one mono stream ) and > > open another stereo stream > > Hmm, the good ol' resource manager :) OK, I will try to find a > solution. I have not much > time these days, but maybe it rains again, and I get bored, who knows :D > > Best Regards, > Manuel > simple fix may be adding counts of the resource and check that all resources are enough before allocating them

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