September 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] cmipci CMI8738-MC6 - left channel no sound
by Clemens Ladisch 17 Sep '10

17 Sep '10

Tom Corner wrote: > Simple mixer control 'Line-In Mode',0 > Items: 'Line-In' 'Rear Output' 'Bass Output' > Item0: 'Line-In' OK. > Simple mixer control 'Mic-In Mode',0 > Items: 'Mic-In' 'Center/LFE Output' > Item0: 'Center/LFE Output' This should be Min-In (I guess your card doesn't have shared mic-in jack). > Simple mixer control 'Phone',0 > Mono: Playback 7 [100%] [on] I'd guess you don't need this. > Simple mixer control 'Four Channel Mode',0 > Mono: Playback [off] This might have some effect, but I don't know which setting should be the correct one. Regards, Clemens

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[alsa-devel] Emulated IEC958 Card
by Jassi Brar 17 Sep '10

17 Sep '10

Hi, Ok, so there is already the iec958 plugin, but wouldn't it be nice to have an 'Emulated IEC958-Card' driver that implements implements all the possible components of the IEC958-cum-AES/EBU specification. The user could have the look and feel of a real card via ALSA control and audio i/f .... User Data can be provided too (which lacks in the iec958 plugin ?) This driver might be boon for low-end embedded devices which have neither IEC958 controller in the SoC nor a codec chip that supports it. In fact the SPDIF controller of one SoC that I know implements only a subset of the standard with limitations. The only requirement for the emulated-iec958 card would be a shifter with deep enough fifo, that can shift out bits at around 6Mbps (for 48KHz Stereo). And most SoCs have such a resource available that can be put to the use -- OMAP has McBSP, Samsung's and other SoCs can leverage a SPI controller. Any suggestions welcome. Regards, Jassi PS: I start the topic upon insistence from Angat <yadi.brar01(a)gmail.com>, a grad student who wanted to implement the feature for his univ project on his beagle board.

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[alsa-devel] Realtek ALC888 recognized as having 2 channels instead of 8
by Munzir Taha (منذر طه) 17 Sep '10

17 Sep '10

Hi, A new laptop and as usual I need your help. I cannot see the controls for the 7.1 channels supported by the laptop. I tried to compile a new alsa snapshot just in case it's fixed but I lost the audio device altogether after rebooting. I also tried https://launchpad.net/~ubuntu-audio-dev/+archive/ppa which should be the proper way to update alsa in ubuntu but again the audio device disappears altogether. I tried to use options snd-hda-intel model=targa-8ch-dig and though I can see alsa controls now for the surround system but only 1 jack is working and the sound is coming from the wrong jack (rear instead of front/headphone) beside other problems. Attached is alsa-info -- Munzir Taha Telecommunications and Electronics Engineer Maintainer of Fedora Arabic Translation Project https://listman.redhat.com/mailman/listinfo/fedora-trans-ar Master CIW Designer, ICDL, MOUS, Linux+, LPI, SCSA Riyadh, SA

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[alsa-devel] [PATCH] Two cleanup patches
by David Henningsson 17 Sep '10

17 Sep '10

It's just a boring day today. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH 2/2] ALSA: HDA: Sort CXT5066 quirk table
by David Henningsson 17 Sep '10

17 Sep '10

It was just a boring day. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a12a9f6..e501a85 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3093,19 +3093,19 @@ static const char *cxt5066_models[CXT5066_MODELS] = { }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5066_LAPTOP), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", - CXT5066_DELL_LAPTOP), - SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), -- 1.7.1 --------------080203050801010400030502--

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[PATCH 1/2] ALSA: HDA: Fix spelling (change VOSTO to VOSTRO)
by David Henningsson 17 Sep '10

17 Sep '10

It was just a boring day. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7e22ed1..a12a9f6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3075,7 +3075,7 @@ enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ - CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ + CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_HP_LAPTOP, /* HP Laptop */ @@ -3086,7 +3086,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", - [CXT5066_DELL_VOSTO] = "dell-vostro", + [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", [CXT5066_HP_LAPTOP] = "hp-laptop", @@ -3099,8 +3099,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), + SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), @@ -3207,7 +3207,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->capture_prepare = cxt5066_olpc_capture_prepare; spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; - case CXT5066_DELL_VOSTO: + case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; -- 1.7.1 --------------080203050801010400030502 Content-Type: text/x-patch; name="0002-ALSA-HDA-Sort-CXT5066-quirk-table.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="0002-ALSA-HDA-Sort-CXT5066-quirk-table.patch"

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[alsa-devel] Programming ALSA sound device.
by Bankim Bhavsar 17 Sep '10

17 Sep '10

Producer thread writes sound bytes worth 10 ms every 10 ms to a ring buffer. Consumer thread is responsible for writing sound bytes to ALSA sound device in a strictly non-blocking fashion. Consumer thread checks snd_pcm_avail/snd_pcm_avail_update, fetches minimum of sound bytes available in ring buffer and availFrames(returned by snd_pcm_avail_update) and writes to sound device using snd_pcm_writei(). Consumer thread polls every 20 ms. However on consumer thread snd_pcm_avail/snd_pcm_avail_update() returns 0 at least 3-4 times consecutively (total 60-80 ms) and on every 5th-6th poll snd_pcm_avail/snd_pcm_avail_update returns full buffer size occasionally reporting an under-run. On an average, under-run is reported at least 10-15 times a minute which is not desirable. Following are the hardware and software parameter settings. hardware params: 16-bit 44100 stereo buffer time 200 ms period time 25 ms software params: start_threshold: full buffer size no setting for avail_min Alternatively I've tried using wait_for_poll() technique as mentioned in the example. http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html However poll() unblocks very rapidly and producer ring buffer doesn't have enough bytes. What's the best technique for PCM playback for such use cases? Is it okay to skip calling snd_pcm_avail/snd_pcm_avail_update and directly attempt writing to ALSA sound device instead? Let me know if more information is required. Thanks, Bankim.

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Re: [alsa-devel] [patch] ALSA: usb/mixer: remove bogus cast
by Takashi Iwai 17 Sep '10

17 Sep '10

At Thu, 16 Sep 2010 20:13:47 +0200, Dan Carpenter wrote: > > "uinfo->value.enumerated.item" is an unsigned int. If it's negative > when we do the comparison: > if ((int)uinfo->value.enumerated.item >= cval->max) > then we would read past the end of the array on the next line. This cast was for the old crappy gcc who spews bogus warnings when compared with signed and unsigned ints. The kernel sets a compile flag for avoiding this since long time ago, so we can get rid of such cast gracefully now, indeed. > I also changed the strcpy() to strlcpy() out of paranoia. > > Signed-off-by: Dan Carpenter <error27(a)gmail.com> Applied now. Thanks. Takashi > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > index 3ed3901..93c95c5 100644 > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -1642,9 +1642,10 @@ static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl > uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; > uinfo->count = 1; > uinfo->value.enumerated.items = cval->max; > - if ((int)uinfo->value.enumerated.item >= cval->max) > + if (uinfo->value.enumerated.item >= cval->max) > uinfo->value.enumerated.item = cval->max - 1; > - strcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item]); > + strlcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item], > + sizeof(uinfo->value.enumerated.name)); > return 0; > } > >

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[alsa-devel] [PATCH 0/4] ASoC: fsi: DMA support 1st step
by Kuninori Morimoto 16 Sep '10

16 Sep '10

Dear Mark, Liam These patches are 1st step for DMA support of FSI driver Kuninori Morimoto (4): ASoC: fsi: Add fsi_dma_get_area ASoC: fsi: Add fsi_dma_soft_push/pop function ASoC: fsi: modify variable name to easy to understand ASoC: fsi: merge fsi_data_push/pop to fsi_fifo_data_ctrl These patches didn't add new feature. FSI driver is still using PIO. And above patches indicate it as "soft dma". I will add DMAEngine support for FSI in future Best regards -- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: ak4642: code clean up
by Kuninori Morimoto 16 Sep '10

16 Sep '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 22 +++++++++++----------- 1 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c84cc9c..a95ff4f 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -422,7 +422,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION); codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = ak4642->control_data; + codec->control_data = ak4642->control_data; snd_soc_add_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); @@ -431,12 +431,12 @@ static int ak4642_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { - .probe = ak4642_probe, - .resume = ak4642_resume, - .read = ak4642_read_reg_cache, - .write = ak4642_write, - .reg_cache_size = ARRAY_SIZE(ak4642_reg), - .reg_word_size = sizeof(u8), + .probe = ak4642_probe, + .resume = ak4642_resume, + .read = ak4642_read_reg_cache, + .write = ak4642_write, + .reg_cache_size = ARRAY_SIZE(ak4642_reg), + .reg_word_size = sizeof(u8), .reg_cache_default = ak4642_reg, }; @@ -448,7 +448,7 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c, int ret; ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); - if (ak4642 == NULL) + if (!ak4642) return -ENOMEM; i2c_set_clientdata(i2c, ak4642); @@ -481,9 +481,9 @@ static struct i2c_driver ak4642_i2c_driver = { .name = "ak4642-codec", .owner = THIS_MODULE, }, - .probe = ak4642_i2c_probe, - .remove = __devexit_p(ak4642_i2c_remove), - .id_table = ak4642_i2c_id, + .probe = ak4642_i2c_probe, + .remove = __devexit_p(ak4642_i2c_remove), + .id_table = ak4642_i2c_id, }; #endif -- 1.7.0.4

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[alsa-devel] [PATCH 1/3] ASoC: Add debugfs listing of registered CODECs
by Mark Brown 16 Sep '10

16 Sep '10

Help with diagnostics for machine driver setup by listing all the registered CODECs in debugfs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 59b99fb..5e4a1f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -271,6 +271,33 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) debugfs_remove_recursive(codec->debugfs_codec_root); } +static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + ssize_t ret = 0; + struct snd_soc_codec *codec; + + if (!buf) + return -ENOMEM; + + list_for_each_entry(codec, &codec_list, list) + ret += snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", + codec->name); + + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + + return ret; +} + +static const struct file_operations codec_list_fops = { + .read = codec_list_read_file, + .llseek = default_llseek,/* read accesses f_pos */ +}; + #else static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) @@ -3192,6 +3219,11 @@ static int __init snd_soc_init(void) "ASoC: Failed to create debugfs directory\n"); debugfs_root = NULL; } + + if (!debugfs_create_file("codecs", 0444, debugfs_root, NULL, + &codec_list_fops)) + pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); + #endif return platform_driver_register(&soc_driver); -- 1.7.1

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Re: [alsa-devel] cmipci surround sound front-left plays on lfe
by Clemens Ladisch 16 Sep '10

16 Sep '10

Tom Corner wrote: > On 09/10/10 09:10, Clemens Ladisch wrote: >> Tom Corner wrote: >>> The surround sound 5.1 is playing fine except channel 0 (front-left) >>> is playing on the woofer. >> >> And the software's front left channel is also playing on the front left >> speaker? And the subwoofer channel isn't playing on any speaker? > > No the subwoofer is playing correctly. The front-left is playing *only* > on the subwoofer. There is no sound at all coming out of the front-left > speaker. In other words, what happens is this? speaker output ---------------------- front L silence front R front R center center bass mix of front L + bass surround L surround L surround R surround R What happens when you run speaker-test with 4 and 2 channels? Regards, Clemens

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[alsa-devel] [PATCH] ALSA: HDA: Enable internal speaker on Dell M101z
by David Henningsson 16 Sep '10

16 Sep '10

BugLink: http://launchpad.net/bugs/640254 In some cases a magic processing coefficient is needed to enable the internal speaker on Dell M101z. According to Realtek, this processing coefficient is only present on ALC269vb. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] ALSA: core: Allow card id change to the same string
by Peter Ujfalusi 16 Sep '10

16 Sep '10

When user want to change the card id to the same string on the card via /sys/class/sound/cardX/id, do not report error. Instead return with success without doing anything. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/core/init.c | 9 +++++++-- 1 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index ec4a50c..2de45fb 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -607,11 +607,16 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return -EEXIST; } for (idx = 0; idx < snd_ecards_limit; idx++) { - if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) - goto __exist; + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) { + if (card == snd_cards[idx]) + goto __ok; + else + goto __exist; + } } strcpy(card->id, buf1); snd_info_card_id_change(card); +__ok: mutex_unlock(&snd_card_mutex); return count; -- 1.7.2.2

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[alsa-devel] [PATCH] configure.in: Create missing include/pcmcia
by Ozan Çağlayan 16 Sep '10

16 Sep '10

With kernels >= 2.6.36, the include/pcmcia/cs_types.h file can not be created because of the missing parent directory. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- configure.in | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/configure.in b/configure.in index c8fefef..82ef3c0 100644 --- a/configure.in +++ b/configure.in @@ -461,7 +461,7 @@ AC_DEFUN([CHECK_KERNEL_HEADER], [ AC_MSG_RESULT(no) if test ! -f include/$1; then mkdir -p include/linux include/asm include/media - mkdir -p include/linux/regulator include/linux/usb + mkdir -p include/linux/regulator include/linux/usb include/pcmcia if test -z "$2" ; then echo "Creating a dummy <$1>..." touch include/$1 -- 1.7.2.1

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[PATCH] ALSA: HDA: Enable internal speaker on Dell M101z
by David Henningsson 16 Sep '10

16 Sep '10

BugLink: http://launchpad.net/bugs/640254 In some cases a magic processing coefficient is needed to enable the internal speaker on Dell M101z. According to Realtek, this processing coefficient is only present on ALC269vb. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab2947d..2063d0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14671,6 +14671,7 @@ static int alc269_resume(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, + ALC269_FIXUP_DELL_M101Z, }; static const struct alc_fixup alc269_fixups[] = { @@ -14680,11 +14681,20 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, + [ALC269_FIXUP_DELL_M101Z] = { + .verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 13}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, + {} + } + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; -- 1.7.1 --------------000100000009090600000802--

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[alsa-devel] [PATCH] ASoC: da7210: code clean up
by Kuninori Morimoto 16 Sep '10

16 Sep '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 18 +++++++++--------- 1 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eabf3c0..58bb9b9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -454,7 +454,7 @@ static int da7210_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->control_data; + codec->control_data = da7210->control_data; codec->hw_write = (hw_write_t)i2c_master_send; /* FIXME @@ -547,11 +547,11 @@ static int da7210_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_da7210 = { - .probe = da7210_probe, - .read = da7210_read, - .write = da7210_write, - .reg_cache_size = ARRAY_SIZE(da7210_reg), - .reg_word_size = sizeof(u8), + .probe = da7210_probe, + .read = da7210_read, + .write = da7210_write, + .reg_cache_size = ARRAY_SIZE(da7210_reg), + .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, }; @@ -597,9 +597,9 @@ static struct i2c_driver da7210_i2c_driver = { .name = "da7210-codec", .owner = THIS_MODULE, }, - .probe = da7210_i2c_probe, - .remove = __devexit_p(da7210_i2c_remove), - .id_table = da7210_i2c_id, + .probe = da7210_i2c_probe, + .remove = __devexit_p(da7210_i2c_remove), + .id_table = da7210_i2c_id, }; #endif -- 1.7.0.4

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[alsa-devel] [PATCH] ALSA nvhdmi: Add missing codec IDs
by Stephen Warren 15 Sep '10

15 Sep '10

Adds codec IDs for various other NVIDIA GPUs to the NVIDIA HDMI audio driver. This is a backport of 9cf2657d05f9f9e04c3e113d68bf7cef5a942070 suitable for inclusion into 2.6.34 and 2.6.35. ... but subset so that it solely adds new IDs, and doesn't reformat nor rename any existing IDs. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/pci/hda/patch_nvhdmi.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index b0652ac..dccfcf4 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -559,6 +559,21 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ @@ -573,6 +588,21 @@ MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0010"); +MODULE_ALIAS("snd-hda-codec-id:10de0011"); +MODULE_ALIAS("snd-hda-codec-id:10de0012"); +MODULE_ALIAS("snd-hda-codec-id:10de0013"); +MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0018"); +MODULE_ALIAS("snd-hda-codec-id:10de0019"); +MODULE_ALIAS("snd-hda-codec-id:10de001a"); +MODULE_ALIAS("snd-hda-codec-id:10de001b"); +MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0040"); +MODULE_ALIAS("snd-hda-codec-id:10de0041"); +MODULE_ALIAS("snd-hda-codec-id:10de0042"); +MODULE_ALIAS("snd-hda-codec-id:10de0043"); +MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); -- 1.6.0.6 nvpublic

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Re: [alsa-devel] [PATCH] ASoC: WM8985: Initial driver
by Mark Brown 15 Sep '10

15 Sep '10

On Wed, Sep 15, 2010 at 03:23:20PM +0100, Dimitris Papastamos wrote: > The WM8985 is a low power, high quality, feature-rich stereo > CODEC designed for portable multimedia applications that > require low power consumption and high quality audio. > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Applied, thanks. > sound/soc/codecs/wm8985.c | 1195 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/wm8985.h | 1046 +++++++++++++++++++++++++++++++++++++++ > 2 files changed, 2241 insertions(+), 0 deletions(-) As discussed I added the Makefile and Kconfig entries as I did so.

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Re: [alsa-devel] [PATCH] ASoC: WM8985: Initial driver
by Liam Girdwood 15 Sep '10

15 Sep '10

On Wed, 2010-09-15 at 15:23 +0100, Dimitris Papastamos wrote: > The WM8985 is a low power, high quality, feature-rich stereo > CODEC designed for portable multimedia applications that > require low power consumption and high quality audio. > > Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> > --- > sound/soc/codecs/wm8985.c | 1195 +++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/codecs/wm8985.h | 1046 +++++++++++++++++++++++++++++++++++++++ > 2 files changed, 2241 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/codecs/wm8985.c > create mode 100644 sound/soc/codecs/wm8985.h Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0

[alsa-devel] [PATCH 1/3] ASoC: tlv320aic3x: Use snd_soc_read and snd_soc_write
by Jarkko Nikula 15 Sep '10

15 Sep '10

Start phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read and aic3x_write calls in order to switch to soc-cache helpers. This patch replaces aic3x_read_reg_cache and aic3x_write with snd_soc_read and snd_soc_write. This is basically null-op since .read and .write in soc_codec_dev_aic3x points to aic3x_read_reg_cache and aic3x_write. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 194 ++++++++++++++++++++-------------------- 1 files changed, 98 insertions(+), 96 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 94dc707..138165c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -839,8 +839,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int clk; /* select data word length */ - data = - aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); + data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4)); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -854,7 +853,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (0x03 << 4); break; } - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, data); /* Fsref can be 44100 or 48000 */ fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000; @@ -869,17 +868,17 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, if (bypass_pll) { pll_q &= 0xf; - aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); - aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); /* disable PLL if it is bypassed */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); } else { - aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); + snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); /* enable PLL when it is used */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); } /* Route Left DAC to left channel input and @@ -888,7 +887,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; if (params_rate(params) >= 64000) data |= DUAL_RATE_MODE; - aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data); + snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); /* codec sample rate select */ data = (fsref * 20) / params_rate(params); @@ -897,7 +896,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, data /= 5; data -= 2; data |= (data << 4); - aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); + snd_soc_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data); if (bypass_pll) return 0; @@ -966,13 +965,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); - aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGC_REG, (pll_d >> 6) << PLLD_MSB_SHIFT); - aic3x_write(codec, AIC3X_PLL_PROGD_REG, - (pll_d & 0x3F) << PLLD_LSB_SHIFT); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + data | (pll_p << PLLP_SHIFT)); + snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, + pll_r << PLLR_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGC_REG, + (pll_d >> 6) << PLLD_MSB_SHIFT); + snd_soc_write(codec, AIC3X_PLL_PROGD_REG, + (pll_d & 0x3F) << PLLD_LSB_SHIFT); return 0; } @@ -980,15 +982,15 @@ found: static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; - u8 rdac_reg = aic3x_read_reg_cache(codec, RDAC_VOL) & ~MUTE_ON; + u8 ldac_reg = snd_soc_read(codec, LDAC_VOL) & ~MUTE_ON; + u8 rdac_reg = snd_soc_read(codec, RDAC_VOL) & ~MUTE_ON; if (mute) { - aic3x_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); - aic3x_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); + snd_soc_write(codec, LDAC_VOL, ldac_reg | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, rdac_reg | MUTE_ON); } else { - aic3x_write(codec, LDAC_VOL, ldac_reg); - aic3x_write(codec, RDAC_VOL, rdac_reg); + snd_soc_write(codec, LDAC_VOL, ldac_reg); + snd_soc_write(codec, RDAC_VOL, rdac_reg); } return 0; @@ -1012,8 +1014,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 iface_areg, iface_breg; int delay = 0; - iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; - iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; + iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; + iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1052,9 +1054,9 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); - aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); - aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } @@ -1072,18 +1074,18 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + reg | PLL_ENABLE); } break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ - reg = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG); - aic3x_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); + reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + snd_soc_write(codec, AIC3X_PLL_PROGA_REG, + reg & ~PLL_ENABLE); } break; case SND_SOC_BIAS_OFF: @@ -1098,8 +1100,8 @@ void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) { u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; u8 bit = gpio ? 3: 0; - u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit); - aic3x_write(codec, reg, val | (!!state << bit)); + u8 val = snd_soc_read(codec, reg) & ~(1 << bit); + snd_soc_write(codec, reg, val | (!!state << bit)); } EXPORT_SYMBOL_GPL(aic3x_set_gpio); @@ -1128,7 +1130,7 @@ void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, if (detect & AIC3X_HEADSET_DETECT_MASK) val |= AIC3X_HEADSET_DETECT_ENABLED; - aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); } EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); @@ -1211,81 +1213,81 @@ static int aic3x_init(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int reg; - aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); - aic3x_write(codec, AIC3X_RESET, SOFT_RESET); + snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); /* DAC default volume and mute */ - aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); - aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); + snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); + snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); /* DAC to HP default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Line Out default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - aic3x_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - aic3x_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ - reg = aic3x_read_reg_cache(codec, LLOPM_CTRL); - aic3x_write(codec, LLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, RLOPM_CTRL); - aic3x_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, MONOLOPM_CTRL); - aic3x_write(codec, MONOLOPM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPLOUT_CTRL); - aic3x_write(codec, HPLOUT_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPROUT_CTRL); - aic3x_write(codec, HPROUT_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPLCOM_CTRL); - aic3x_write(codec, HPLCOM_CTRL, reg | UNMUTE); - reg = aic3x_read_reg_cache(codec, HPRCOM_CTRL); - aic3x_write(codec, HPRCOM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, LLOPM_CTRL); + snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, RLOPM_CTRL); + snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, MONOLOPM_CTRL); + snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPLOUT_CTRL); + snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPROUT_CTRL); + snd_soc_write(codec, HPROUT_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPLCOM_CTRL); + snd_soc_write(codec, HPLCOM_CTRL, reg | UNMUTE); + reg = snd_soc_read(codec, HPRCOM_CTRL); + snd_soc_write(codec, HPRCOM_CTRL, reg | UNMUTE); /* ADC default volume and unmute */ - aic3x_write(codec, LADC_VOL, DEFAULT_GAIN); - aic3x_write(codec, RADC_VOL, DEFAULT_GAIN); + snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); + snd_soc_write(codec, RADC_VOL, DEFAULT_GAIN); /* By default route Line1 to ADC PGA mixer */ - aic3x_write(codec, LINE1L_2_LADC_CTRL, 0x0); - aic3x_write(codec, LINE1R_2_RADC_CTRL, 0x0); + snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); + snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL); /* PGA to Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); /* PGA to Mono Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL); /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); /* Line2 Line Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); /* Line2 to Mono Out default volume, disconnect from Output Mixer */ - aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); - aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { /* Class-D speaker driver init; datasheet p. 46 */ - aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D); - aic3x_write(codec, 0xD, 0x0D); - aic3x_write(codec, 0x8, 0x5C); - aic3x_write(codec, 0x8, 0x5D); - aic3x_write(codec, 0x8, 0x5C); - aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00); - aic3x_write(codec, CLASSD_CTRL, 0); + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); + snd_soc_write(codec, 0xD, 0x0D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, 0x8, 0x5D); + snd_soc_write(codec, 0x8, 0x5C); + snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); + snd_soc_write(codec, CLASSD_CTRL, 0); } /* off, with power on */ @@ -1305,10 +1307,10 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->setup) { /* setup GPIO functions */ - aic3x_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - aic3x_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); } snd_soc_add_controls(codec, aic3x_snd_controls, -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Remove needless codec->bias_level assignment to SND_SOC_BIAS_OFF
by Jarkko Nikula 15 Sep '10

15 Sep '10

This assignment is done by the snd_soc_register_codec so there is no need to redo it in probe function of a codec driver. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- I can separate this patch between TI and other codecs but I think this is trivial enough to be carried either through Mark's or Liam's tree. --- sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/twl4030.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8731.c | 2 -- sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8990.c | 1 - 8 files changed, 0 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 239f056..24f5f49 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -642,7 +642,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) int ret; codec->hw_write = (hw_write_t)i2c_master_send; - codec->bias_level = SND_SOC_BIAS_OFF; ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); if (ret < 0) { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 67d8c04..6f38d61 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -560,7 +560,6 @@ static int ssm2602_probe(struct snd_soc_codec *codec) pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - codec->bias_level = SND_SOC_BIAS_OFF, codec->control_data = ssm2602->control_data; ssm2602_reset(codec); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index a3c5b52..bf6d01f 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1385,7 +1385,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->bias_level = SND_SOC_BIAS_OFF; codec->idle_bias_off = 1; dac33->codec = codec; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c7ee1a4..cbebec6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2245,7 +2245,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_codec_get_mclk() / 1000; - codec->bias_level = SND_SOC_BIAS_OFF; codec->idle_bias_off = 1; twl4030_init_chip(codec); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index d00da20..02ecf54 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -569,7 +569,6 @@ static int wm8510_probe(struct snd_soc_codec *codec) wm8510_reset(codec); /* power on device */ - codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, wm8510_snd_controls, ARRAY_SIZE(wm8510_snd_controls)); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 56f5408..6980ff8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -488,8 +488,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - codec->bias_level = SND_SOC_BIAS_OFF, - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8731->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 4a945d3..6c0ff35 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1550,7 +1550,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); int ret = 0, reg; - codec->bias_level = SND_SOC_BIAS_OFF; INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ffecbd..264828e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1354,7 +1354,6 @@ static int wm8990_probe(struct snd_soc_codec *codec) wm8990_reset(codec); /* charge output caps */ - codec->bias_level = SND_SOC_BIAS_OFF; wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); -- 1.7.1

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[alsa-devel] need help diagnosing USB MIDI latency problem with MPD16
by Krzysztof Foltman 15 Sep '10

15 Sep '10

Hello, Some time ago, I wrote the Akai MPD16 protocol handlers (input/output) for the USB MIDI driver. Unfortunately, the kernel driver seems to suffer from a rather embarrassing input latency, it looks like the MIDI event timing granularity is >100ms (so the latency is variable, making the pad practically useless except if a rhythm is played with the rate matching the polling rate!). At the same time, a simple Python based contraption using libusb and bulkRead in a while loop, gets much better results. The same USB MIDI driver in the same kernel and PC doesn't have noticeable problems with class-compliant E-Mu Xboard25 connected to the same USB port. Both devices use endpoints in bulk mode. I've attached the excerpts of lsusb -v output for both devices - the only differences are: 1) the Akai device isn't class compliant, 2) the E-mu device has one pair of inputs/outputs, the Akai has two (control/configuration and MIDI proper) 3) the maximum packet length is 32 for E-mu, 64 for Akai. When I enabled logging in midi.c (#define DUMP_PACKETS), it looks like the device is returning data packets rather infrequently, as a single packet often contains multiple events (up to 3) when two pads are pressed in rapid succession. So, it looks like a problem outside of my snd_usbmidi_akai_input. One thing that *does* help, though, is disabling the 'control' port by adding endpoints[0].in_cables = 0; in Akai quirk handling in snd_usbmidi_create - when input from the control port is not polled, the choppiness disappears completely. I'm guessing that the device itself (or something in between) has some problems having to handle the URBs for both endpoints, and that perhaps the IN/NAK exchange on a "control" endpoint causes the device's microcontroller (or something else?) to stall for a while. However, in that case, a proper solution (one that doesn't involve sacrificing the ability to use control port for configuration) would involve submitting URBs for the control input endpoint only when control port is open on ALSA side, and cancelling them when the port is closed. However, that sounds pretty invasive. Any hints? The device in question is rather uncommon and hasn't been supported for a while, so refactoring the whole USB MIDI driver to accommodate it doesn't sound like a great idea - but still, the latency thing sucks pretty bad for a drumming-oriented device! Thanks in advance, Krzysztof

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[alsa-devel] Fwd: Microphone not working on PCI ID 8086:3b56
by Jeff Chua 15 Sep '10

15 Sep '10

Hi, I'm having problem getting the microphone to work on the Lenovo X201s, but the audio output is working fine. It's showing up as: 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) [00:1b.0 0403: 8086:3b56 (rev 06)] # uname -s Linux boston 2.6.36-rc4 #57 SMP PREEMPT Tue Sep 14 05:12:21 EDT 2010 x86_64 GNU/Linux # cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer 160: [ 5] : control # cat /proc/asound/pcm 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 # dmesg thinkpad_acpi: Console audio control enabled, mode: override (read/write) input: ThinkPad Extra Buttons as /devices/platform/thinkpad_acpi/input/input6 HDA Intel 0000:00:1b.0: PCI INT B -> GSI 17 (level, low) -> IRQ 17 HDA Intel 0000:00:1b.0: irq 45 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda-codec: No codec parser is available ALSA device list: #0: HDA Intel at 0xf2520000 irq 45 #5: ThinkPad Console Audio Control at EC reg 0x30, fw 6QHT29WW-1.10 I've alsa-plugins-1.0.23, alsa-utils-1.0.23, alsa-lib-1.0.23 installed. alsamixer is showing Capture at 100%, Analog Mic Boost at 40db and Digital at 100, but captured using "arecord" or "rec" and then playback is all just dead silence! Any pointers on how to make this work would be very much appreciated. Thanks, Jeff

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[alsa-devel] [PATCH] patch_nvhdmi.c: Fix supported sample rate list.
by Stephen Warren 15 Sep '10

15 Sep '10

22050 isn't a valid HDMI sample rate. 32000 is. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-By: Wei Ni <wni(a)nvidia.com> --- sound/pci/hda/patch_nvhdmi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 69b950d..baa108b 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { #else /* support all rates and formats */ #define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 -- 1.7.0.4 nvpublic

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