April 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Lenovo 3000 N100 ALC861-VD quirk
by Chris Pemberton 25 Apr '08

25 Apr '08

Hello, I'm revisiting an unresolved bug report I opened several months ago: bug # 0003468 The bug still exists as of ALSA v1.0.15 (Ubuntu 8.04). A quick recap: The following line in patch_realtek.c is my machine: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), If I edit this line as follows, I get sound: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo" ALC861VD_3ST), (or I can just pass model=3stack to get sound). Inserting headphones doesn't mute the speakers or power the headphones. But I can manually unmute the headphones via alsamixer (but speakers are still not muted). I do see that the special keyboard keys for speaker mute, volume up and volume down are working correctly; but I don't know if this is something ALSA is responsible for, or X, or Ubuntu. So if this is ALSA, thanks! Anything more I can do to help resolve this? Thanks, Chris

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[alsa-devel] 2.6.25-rc8: why #define SND_HDA_NEEDS_RESUME is not set ?
by warped 25 Apr '08

25 Apr '08

Hi all, I have issue with 2.6.24/25 kernels and analog outs after s3 resume. Simply: after s3 resume analog outs are dead (no sound). SPDIF works OK. With 2.6.23.xx all was OK I browse ALSA code in 2.6.25-rc8 and see that in hda_codec.h define SND_HDA_NEEDS_RESUME is commented. Q is: -might uncommenting this line help me with my issue ? -why it is commented ? My hw is Abit AN-M2HD with ALC889. During boot Kernel reports: user.info kernel: hda_codec: Unknown model for ALC883, trying auto-probe from BIOS...

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: add a private field for ac97-device drivers and let ucb1400 be its first user
by Takashi Iwai 25 Apr '08

25 Apr '08

At Fri, 25 Apr 2008 08:29:08 +0200 (CEST), Jaroslav Kysela wrote: > > changeset: 6120:e2ff47e8771b > tag: tip > user: perex > date: Fri Apr 25 08:29:05 2008 +0200 > files: include/ac97_codec.h kernel/drivers/input/touchscreen/ucb1400_ts.c > description: > add a private field for ac97-device drivers and let ucb1400 be its first user > From: Sebastian Siewior <bigeasy(a)linutronix.de> > > Currently the UCB1400 driver discovers the interrupt via probing. This works > probably only on x86. This patch adds a private field to the ac97 struct > where the ac97 driver can deposit informations for the device driver that > serves a device which is attached to the ac97 bus. > This patch also converts the UCB1400 driver to use this information if > available. > > Signed-off-by: Sebastian Siewior <bigeasy(a)linutronix.de> Jaroslav, as I posted yesterday, I don't think it's good to use a void pointer blindly there. In this case, it's different from a normal private_data, which is asigned and used by the same driver. This is a data passed by ac97 controller driver to another device driver. Thus, it should be a strong type. Also, a cast from a pointer to an int causes a compile warning. So, could you revert it until all issues are solved? thanks, Takashi > > diff -r 9c74d781f388 -r e2ff47e8771b include/ac97_codec.h > --- a/include/ac97_codec.h Fri Apr 25 08:15:12 2008 +0200 > +++ b/include/ac97_codec.h Fri Apr 25 08:29:05 2008 +0200 > @@ -475,6 +475,10 @@ struct snd_ac97 { > struct snd_ac97_build_ops * build_ops; > void *private_data; > void (*private_free) (struct snd_ac97 *ac97); > + /* This field is used by device drivers which serve devices which are > + * attached to the AC97 bus. > + */ > + void *device_private_data; > /* --- */ > struct snd_ac97_bus *bus; > struct pci_dev *pci; /* assigned PCI device - used for quirks */ > diff -r 9c74d781f388 -r e2ff47e8771b kernel/drivers/input/touchscreen/ucb1400_ts.c > --- a/kernel/drivers/input/touchscreen/ucb1400_ts.c Fri Apr 25 08:15:12 2008 +0200 > +++ b/kernel/drivers/input/touchscreen/ucb1400_ts.c Fri Apr 25 08:29:05 2008 +0200 > @@ -492,10 +492,14 @@ static int ucb1400_ts_probe(struct devic > goto err_free_devs; > } > > - error = ucb1400_detect_irq(ucb); > - if (error) { > - printk(KERN_ERR "UCB1400: IRQ probe failed\n"); > - goto err_free_devs; > + if (!ucb->ac97->device_private_data) { > + error = ucb1400_detect_irq(ucb); > + if (error) { > + printk(KERN_ERR "UCB1400: IRQ probe failed\n"); > + goto err_free_devs; > + } > + } else { > + ucb->irq = (int) ucb->ac97->device_private_data; > } > > error = request_irq(ucb->irq, ucb1400_hard_irq, IRQF_TRIGGER_RISING, > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by the 2008 JavaOne(SM) Conference > Don't miss this year's exciting event. There's still time to save $100. > Use priority code J8TL2D2. > http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/java… > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

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[alsa-devel] [PATCH v2][ASoC v2] Replace snd_soc_free_codec with snd_soc_unregister_codec
by Timur Tabi 24 Apr '08

24 Apr '08

The functions snd_soc_register_codec() and snd_soc_free_codec() are companion functions, but snd_soc_free_codec() releases resources (via kfree) that were not allocated by snd_soc_register_codec(). This means that the caller of snd_soc_register_codec() has to use a specific method to allocate these resources. So we replace snd_soc_free_codec() with snd_soc_unregister_codec(). This new function only unregisters the codec but does not free the memory resources. The caller can now allocate and free the memory resources as it sees fit. Also updated all callers of snd_soc_free_codec() to call snd_soc_unregister_codec() instead and free the resources themselves. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- I cannot get ASoC V2 to compile on my machine, so this code is untested. include/sound/soc-codec.h | 2 +- sound/soc/codecs/wm8350.c | 8 ++++++-- sound/soc/codecs/wm8731.c | 8 ++++++-- sound/soc/codecs/wm8750.c | 8 ++++++-- sound/soc/codecs/wm8753.c | 8 ++++++-- sound/soc/codecs/wm9712.c | 8 ++++++-- sound/soc/codecs/wm9713.c | 8 ++++++-- sound/soc/soc-core.c | 11 ++++------- 8 files changed, 41 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index 3cf4819..a809160 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -160,7 +160,7 @@ struct snd_soc_codec *snd_soc_new_codec( int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev); -void snd_soc_free_codec(struct snd_soc_codec *codec); +void snd_soc_unregister_codec(struct snd_soc_codec *codec); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, struct snd_card *card, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3195d13..4ea8ced 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,7 +1436,9 @@ static int wm8350_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8350); prv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1446,7 +1448,9 @@ static int wm8350_codec_remove(struct platform_device *pdev) struct wm8350_data *wm8350 = codec->private_data; snd_soc_unregister_codec_dai(wm8350->dai); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); kfree(wm8350); return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0fe8766..2195c77 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -595,7 +595,9 @@ static int wm8731_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8731); wm8731_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -606,7 +608,9 @@ static int wm8731_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8731->dai); kfree(wm8731); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1b0296f..23e9126 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -898,7 +898,9 @@ static int wm8750_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8750); wm8750_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -909,7 +911,9 @@ static int wm8750_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8750->dai); kfree(wm8750); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } static struct platform_driver wm8750_codec_driver = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0d795f4..1cb2d64 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1584,7 +1584,9 @@ voice_err: codec_err: kfree(codec->private_data); err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1603,7 +1605,9 @@ static int wm8753_codec_remove(struct platform_device *pdev) /* free codec */ kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 16b6a18..7014190 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -715,7 +715,9 @@ aux_dai_err: codec_err: kfree(wm9712); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -728,7 +730,9 @@ static int wm9712_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9712->aux_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5e99bbe..d9ded51 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1154,7 +1154,9 @@ aux_dai_err: codec_err: kfree(wm9713); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1168,7 +1170,9 @@ static int wm9713_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9713->voice_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 939b543..7dc6d68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1856,21 +1856,18 @@ int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** - * snd_soc_free_codec - unregister and free codec. + * snd_soc_unregister_codec - unregister a codec. * @codec: codec driver * - * Unregisters a codec driver with the core and frees all its resources. + * Unregisters a codec driver with the core. */ -void snd_soc_free_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct snd_soc_codec *codec) { mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - if (codec->reg_cache) - kfree(codec->reg_cache); - kfree(codec); } -EXPORT_SYMBOL_GPL(snd_soc_free_codec); +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); /** * snd_soc_register_platform_dai - registers a platform DAI. -- 1.5.5.1

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[alsa-devel] [PATCH][ASoC v2] Replace snd_soc_free_codec with snd_soc_unregister_codec
by Timur Tabi 24 Apr '08

24 Apr '08

The functions snd_soc_register_codec() and snd_soc_free_codec() are companion functions, but snd_soc_free_codec() releases resources (via kfree) that were not allocated by snd_soc_register_codec(). This means that the caller of snd_soc_register_codec() has to use a specific method to allocate these resources. So we replace snd_soc_free_codec() with snd_soc_unregister_codec(). This new function only unregisters the codec but does not free the memory resources. The caller can now allocate and free the memory resources as it sees fit. Also updated all callers of snd_soc_free_codec() to call snd_soc_unregister_codec() instead and free the resources themselves. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- include/sound/soc-codec.h | 2 +- sound/soc/codecs/wm8350.c | 8 ++++++-- sound/soc/codecs/wm8731.c | 8 ++++++-- sound/soc/codecs/wm8750.c | 8 ++++++-- sound/soc/codecs/wm8753.c | 8 ++++++-- sound/soc/codecs/wm9712.c | 8 ++++++-- sound/soc/codecs/wm9713.c | 8 ++++++-- sound/soc/soc-core.c | 11 ++++------- 8 files changed, 41 insertions(+), 20 deletions(-) diff --git a/include/sound/soc-codec.h b/include/sound/soc-codec.h index 3cf4819..6f132ba 100644 --- a/include/sound/soc-codec.h +++ b/include/sound/soc-codec.h @@ -160,7 +160,7 @@ struct snd_soc_codec *snd_soc_new_codec( int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev); -void snd_soc_free_codec(struct snd_soc_codec *codec); +void snd_soc_deregister_codec(struct snd_soc_codec *codec); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, struct snd_card *card, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3195d13..4ea8ced 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1436,7 +1436,9 @@ static int wm8350_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8350); prv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1446,7 +1448,9 @@ static int wm8350_codec_remove(struct platform_device *pdev) struct wm8350_data *wm8350 = codec->private_data; snd_soc_unregister_codec_dai(wm8350->dai); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); kfree(wm8350); return 0; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0fe8766..2195c77 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -595,7 +595,9 @@ static int wm8731_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8731); wm8731_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -606,7 +608,9 @@ static int wm8731_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8731->dai); kfree(wm8731); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1b0296f..23e9126 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -898,7 +898,9 @@ static int wm8750_codec_probe(struct platform_device *pdev) codec_err: kfree(wm8750); wm8750_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -909,7 +911,9 @@ static int wm8750_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm8750->dai); kfree(wm8750); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } static struct platform_driver wm8750_codec_driver = { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0d795f4..1cb2d64 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1584,7 +1584,9 @@ voice_err: codec_err: kfree(codec->private_data); err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1603,7 +1605,9 @@ static int wm8753_codec_remove(struct platform_device *pdev) /* free codec */ kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 16b6a18..7014190 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -715,7 +715,9 @@ aux_dai_err: codec_err: kfree(wm9712); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -728,7 +730,9 @@ static int wm9712_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9712->aux_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 5e99bbe..d9ded51 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1154,7 +1154,9 @@ aux_dai_err: codec_err: kfree(wm9713); priv_err: - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return ret; } @@ -1168,7 +1170,9 @@ static int wm9713_codec_remove(struct platform_device *pdev) snd_soc_unregister_codec_dai(wm9713->voice_dai); kfree(codec->private_data); - snd_soc_free_codec(codec); + snd_soc_unregister_codec(codec); + kfree(codec->reg_cache); + kfree(codec); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 939b543..7dc6d68 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1856,21 +1856,18 @@ int snd_soc_register_codec(struct snd_soc_codec *codec, struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** - * snd_soc_free_codec - unregister and free codec. + * snd_soc_unregister_codec - unregister a codec. * @codec: codec driver * - * Unregisters a codec driver with the core and frees all its resources. + * Unregisters a codec driver with the core. */ -void snd_soc_free_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct snd_soc_codec *codec) { mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); - if (codec->reg_cache) - kfree(codec->reg_cache); - kfree(codec); } -EXPORT_SYMBOL_GPL(snd_soc_free_codec); +EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); /** * snd_soc_register_platform_dai - registers a platform DAI. -- 1.5.5.1

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Re: [alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Thank you. I tried that but on starting a playback it gives me an error that the device is in use and does not playback. Should i be enabling anything else for my .copy function to be invoked and successful playback to happen? On Thu, Apr 24, 2008 at 10:53 AM, Eliot Blennerhassett < linux(a)audioscience.com> wrote: > Harsha priya gupta wrote: > > Hi, > > > > Can anyone help me - I need to modify an existing driver such that a new > > .copy function is invoked. how to make sure that mmap is diabled? > > > > Remove SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID > > from the struct snd_pcm_hardware.info initialisation. > > -- > Eliot > > > > > -- -Harsha

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[alsa-devel] how to disable mmap and allow .copy function to be called
by Harsha priya gupta 24 Apr '08

24 Apr '08

Hi, Can anyone help me - I need to modify an existing driver such that a new .copy function is invoked. how to make sure that mmap is diabled? Thanks, Harsha

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[alsa-devel] snd_pcm_rewind() weirdness
by Lennart Poettering 24 Apr '08

24 Apr '08

Hey! On ALSA 1.0.16 with the HDA driver I am experiencing some weird things with snd_pcm_rewind() which I think are the result of a kernel bug, but I am not sure. Here's what I do to trigger this: I play some PCM. Eventually I want to rewrite what I already have written. I call snd_pcm_hwsync() and then snd_pcm_avail_update() to determine how "empty" the playback buffer is. I subtract that value from the total buffer size. The resulting value should be something resembling the number of frames that I might be able to rewind. I then subtract enough frames to keep away from the playback pointer far enough (50ms, the playback buffer is 370ms, period is 185ms). Now I pass that value to snd_pcm_rewind() and it returns something that is higher than what I passed in. Which I find quite surprising. I had expected that the returned value could be equal or smaller than what I passed in, but never higher! So, my interpretation of the API was apparently wrong. It looks like on HDA the snd_pcm_rewind() parameter is apparently rounded *up* to the next period boundary (which in this case was equivalent to the *entire* buffer)? Of course, rewinding that far immediately caused an XRUN. Which is why I think rounding up in this cases is a bad idea and the API shouldn't work this way. Rounding down is OK, rounding up is call for trouble. To tape over this I added some code to immediately call snd_pcm_forward() with the difference if I noticed that the device forwarded more than I requested. This value also got rounded up -- to twice the period size. So with this backward+forward operation I ended up at a frame index that is farther ahead than where I originally came from! So, what's the expected behaviour of snd_pcm_rewind()? Always round up? What's the rationale behind this? If this is not a bug but expected behaviour, could we at least add something for passing the direction to round to? What I am working on is the "glitch-free" playback model in PulseAudio (http://0pointer.de/blog/projects/pulse-glitch-free.html). To do this I am writing blocks smaller than the period size to the device via the mmap interface all the time. Given that on HDA _rewind() is apparently not granular to the frame size, is writing via mmap in small blocks like that OK? If it's not, how can I learn about this in advance? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [RFC/PATCH 0/1] ALSA SoC support for OMAP
by Jarkko Nikula 24 Apr '08

24 Apr '08

Hi Here is the second version of ASoC support for OMAP. Now based to dev branch of git://opensource.wolfsonmicro.com/linux-2.6-asoc but applies to linux-omap by modifying Kconfig and Makefile in sound/soc/. Code hasn't changed a very much. Some minor clean-ups etc and support for OMAP2430 and 3430 is not included until we have updated arch/arm/plat-omap/mcbsp.c module in place supporting them (work in progress). This have the same issues than first version [1] but TODO list is different: - Pin multiplexing: This belongs to .../plat-omap/mcbsp.c, not here - Multiformat & multilink McBSP DAI: ASoC v2 - omap_mcbsp_start & _stop API change: Probably with or after updated mcbsp.c Note that include/asm-arm/arch-omap/control.h is missing from vanilla kernel at the moment but that's queued for 2.6.26. -- Jarkko 1. http://marc.info/?l=linux-omap&m=120643770810683&w=2

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[alsa-devel] [PATCH 01/11] soc - s3c24xx-i2s - Use linux/io.h
by Mark Brown 23 Apr '08

23 Apr '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0de211c..c95ebe2 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/jiffies.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,7 +33,6 @@ #include <sound/soc.h> #include <asm/hardware.h> -#include <asm/io.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> -- 1.5.5.1

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