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[alsa-devel] [PATCH v2] add channel order map interface in asoc core
by Barry Song 13 Sep '09

13 Sep '09

The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 7 +++++++ sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ 2 files changed, 31 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..5c84566 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +152,9 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_channel_map)(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..ba1f8e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

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[alsa-devel] How to test suspend ?
by Giuliano Pochini 12 Sep '09

12 Sep '09

I couldn't make s2ram work on my machine and /sys/pci/.../power/state virtual files were removed one year ago. How can I test suspend/resume ? -- Giuliano.

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[alsa-devel] SPDIF issues
by Ben 12 Sep '09

12 Sep '09

All, I have been happily using a small setup for the purpose of MythTV. I recently decided to upgrade the graphics so I could enjoy VPDAU which has been integrated into the 0.22 branch of MythTV. There is just one snag, I don't have any sound. I am using an Asus P5E-VM HDMI board, it has onboard sound & graphics. The HDMI interface outputs to my TV and it works fine. I then came across an nvidia 9500GT (gainwood) card which has a VGA/DVI/HDMI out and also takes in sound via SPDIF. I connected the cable to my spdif headers and I can't seem to get any sound. I'm running Ubuntu Karmic so I thought I would upgrade the Alsa drivers/lib/utils to the latest, this has not helped matters. I have checked Alsa mixer and nothing is muted. I have tried: speaker-test -Dplughw:0,1 - (I think this is correct). I'm sure it's something pretty trivial I could be doing wrong but someone mentioned this kind of setup can be quite problematic and suggested I posted the details to this list. Details here: http://www.alsa-project.org/db/?f=ffb9e4b76f37a87f21fec2a4db18295479608d24 I'd appreciate your thoughts! Thanks, Ben

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[alsa-devel] [PATCH 8/8, retry] sound/soc: introduce missing kfree
by Julia Lawall 12 Sep '09

12 Sep '09

From: Julia Lawall <julia(a)diku.dk> Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++-------- 1 files changed, 20 insertions(+), 13 deletions(-) diff --git a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..6096d22 100644 --- a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c +++ b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c @@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op) int size, irq, rc; const __be32 *prop; void __iomem *regs; + int ret; /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); @@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { - iounmap(regs); - return -ENOMEM; + ret = -ENOMEM; + goto out_unmap; } /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; + if (!prop || size < sizeof *prop) { + ret = -ENODEV; + goto out_free; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex); @@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op) if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + ret = -ENODEV; + goto out_free; } /* Disable all interrupts and reset the PSC */ @@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op) &psc_dma_bcom_irq_tx, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; + ret = -ENODEV; + goto out_irq; } /* Save what we've done so it can be found again later */ @@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Tell the ASoC OF helpers about it */ return snd_soc_register_platform(&mpc5200_audio_dma_platform); +out_irq: + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); +out_free: + kfree(psc_dma); +out_unmap: + iounmap(regs); + return ret; } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);

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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 12 Sep '09

12 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

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[alsa-devel] [PATCH 8/8] sound/soc: introduce missing kfree
by Julia Lawall 12 Sep '09

12 Sep '09

From: Julia Lawall <julia(a)diku.dk> Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc5200_dma.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..41ffd57 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -469,8 +469,11 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) + if (!prop || size < sizeof *prop) { + iounmap(regs); + kfree(psc_dma); return -ENODEV; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex);

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[alsa-devel] WM8580 DAI: Debugged
by jassi brar 12 Sep '09

12 Sep '09

Hi, I have been making SMDK6410 I2S work with WM8580 codec. In the process i debugged the wm8580 driver. The patch is:- Last patch(Minor WM8580 enhancements) _minus_ the MCLKRATIO/BCLKRATIO stuff _plus_ the snd_soc_dai definition debugged. The snd_soc_dai debugging is necessary if the Audio Capture is to work with the CODEC driver. Next comes the patch.....

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[alsa-devel] ALSA 1.0.21 release
by Jaroslav Kysela 11 Sep '09

11 Sep '09

Hello all, all ALSA packages except alsa-oss and alsa-firmware (no changes from 1.0.20) are available for download at http://www.alsa-project.org . Changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.20_v1.0.21 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 11 Sep '09

11 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, who is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

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[alsa-devel] [PATCH] add channel order setting interface in asoc core
by Barry Song 11 Sep '09

11 Sep '09

Hi Mark, As discussed, I am adding a patch to add an interface to set the relationship between audio channel No. and slot No. The interface should be really useful because nobody can ensure audio channel n always uses slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1. After the patch is applied, I can change bfin stuff based on it. Thanks, Barry Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 5 +++++ sound/soc/soc-core.c | 18 ++++++++++++++++++ 2 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..7439425 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,9 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_channel_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +151,8 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_channel_order)(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..30d0143 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,24 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_order - configure DAI audio channel order + * @dai: DAI + * @num: how many channels + * @slot: pointer to an array which imply the slot number channel 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_order(dai, num, slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_order); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

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