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

[alsa-devel] CONFIG_SND_HDA_INTEL regression on 2.6.37-rc3 - no audio
by Luis R. Rodriguez 02 Jan '11

02 Jan '11

I just booted 2.6.37-rc3 and noticed my sound with CONFIG_SND_HDA_INTEL stopped working, I get no audio at all. Is this a known regression? I can bisect if not. $ lsmod| grep snd snd_hda_codec_analog 79955 1 snd_hda_intel 25446 3 snd_hda_codec 101539 2 snd_hda_codec_analog,snd_hda_intel snd_hwdep 6546 1 snd_hda_codec snd_pcm 88139 3 snd_hda_intel,snd_hda_codec snd_seq_midi 5900 0 snd_rawmidi 21916 1 snd_seq_midi snd_seq_midi_event 7291 1 snd_seq_midi snd_seq 56979 2 snd_seq_midi,snd_seq_midi_event snd_timer 23148 2 snd_pcm,snd_seq snd_seq_device 6912 3 snd_seq_midi,snd_rawmidi,snd_seq snd 63192 15 snd_hda_codec_analog,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,thinkpad_acpi,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 1240 1 snd snd_page_alloc 8716 2 snd_hda_intel,snd_pcm Luis

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[alsa-devel] mapping externally allocated Scatter Gather DMA buffers
by Manu Abraham 15 Dec '10

15 Dec '10

Hi, I have been wondering how to interface a Scatter Gather region as a DMA buffer for an Audio Grabbing device. The device is a NOC type device, similar to a SOC, the audio grabber is a part of it, for realizing the audio capabilities on the device. The hardware DMA is mostly a Scatter Gather based engine, and it can be described thus, briefly. struct dmabuf { void *vmalloc; dma_addr_t phys; void *virt; int pages; int offset; struct scatterlist *sg_list; struct list_head list; }; where: dmabuf->virt = __get_free_page(GFP_KERNEL); dmabuf->phys = dma_map_single(dev, PAGE_SIZE, DMA_FROM_DEVICE); dmabuf->sg_list = kzalloc(sizeof (struct scatterlist) * pages, GFP_KERNEL); sg_init_table(sg_list, pages); dmabuf->vmalloc = vmalloc((pages + 1) * PAGE_SIZE); In a loop with the pages involved, I do populate the pages into the list; Now I have a stream engine with the hardware, where 8 such dmabufs are allocated. On an interrupt, the device can know from status registers, which dmabuf the device driver needs to read the data from. I wonder what would be the best possible way to interface the stream engine to the Alsa DMA capture engine. I have been eyeballing the alsa sources and whatever documents that are available, but found very little details on the alsa core, but nevertheless found that, it might be possible to pass the vmalloc'd region to substream->runtime->dma_area. I wonder whether this is possible practically and would like to hear your comments on it, whatever feedback, or ideas you have. Any helpful comments appreciated very much. Thanks, Manu

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Re: [alsa-devel] [RFC/PATCH v6 03/12] media: Entities, pads and links
by Clemens Ladisch 13 Dec '10

13 Dec '10

Laurent Pinchart wrote: > A link is a point-to-point oriented connection between two pads, either > on the same entity or on different entities. Data flows from a source > pad to a sink pad. > > Links are stored in the source entity. In the descriptors of USB Audio and HDAudio devices, the links are stored only in the sink. But AFAICS this doesn't matter in the media driver API. > +Links have flags that describe the link capabilities and state. > + > + MEDIA_LINK_FLAG_ACTIVE indicates that the link is active and can be > + used to transfer media data. When two or more links target a sink pad, > + only one of them can be active at a time. > + MEDIA_LINK_FLAG_IMMUTABLE indicates that the link active state can't > + be modified at runtime. If MEDIA_LINK_FLAG_IMMUTABLE is set, then > + MEDIA_LINK_FLAG_ACTIVE must also be set since an immutable link is > + always active. So the intended userspace API for controlling routing is to change the ACTIVE flag of links? In USB and HD audio devices, all links are immutable, and the routing is controlled by 'selector' entities that activate exactly one of their input pads. In userspace, this entity shows up as a mixer control. I guess it would be possible to map the ACTIVE flag onto these controls. Alternatively, entities can have 'mute' mixer controls associated with their pads. In this case, multiple unmuted inputs would be mixed together. > +#define MEDIA_ENTITY_TYPE_MASK 0x00ff0000 > +#define MEDIA_ENTITY_SUBTYPE_MASK 0x0000ffff > + > +#define MEDIA_ENTITY_TYPE_NODE (1 << MEDIA_ENTITY_TYPE_SHIFT) > ... > +#define MEDIA_ENTITY_TYPE_NODE_ALSA (MEDIA_ENTITY_TYPE_NODE + 3) ALSA has PCM and MIDI devices, and several types of mixer controls. (It also has hardware dependent and timer devices, but I don't think these would need topology information.) So we need at least these: MEDIA_ENTITY_TYPE_NODE_ALSA_PCM MEDIA_ENTITY_TYPE_NODE_ALSA_MIDI MEDIA_ENTITY_TYPE_SUBDEV_ALSA_CONTROL Furthermore, topology information is also needed for entities not associated with a mixer control, such as microphones, speakers, jacks/ connectors, and effect units. These entities are defined in the USB and HD audio specifications, but are not yet handled by ALSA. > +struct media_entity { > ... > + union { > + /* Node specifications */ > + struct { > + u32 major; > + u32 minor; > + } v4l; > + struct { > + u32 major; > + u32 minor; > + } fb; > + int alsa; > + int dvb; > + > + /* Sub-device specifications */ > + /* Nothing needed yet */ > + }; ALSA devices are not addressed by their device node but with card/device/ subdevice numbers; mixer controls have numeric IDs, unique per card: struct { int card; int device; int subdevice; } alsa_device; struct { int card; int numid; } alsa_control; Regards, Clemens

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[alsa-devel] low volume with hda base card (see attached alsa-info output
by Halim Sahin 13 Dec '10

13 Dec '10

Hi, I need to set all mixer stuff to 100% and the output isn't ok. Volume is too low. When I switch to another input on my amp. it's about 500% louder. Please have a look to attached ala-info output. I tested with opensuse 11.2 and also with a newer kernel for the distro 2.6.34. Can you please help? Thx. Halim -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

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Re: [alsa-devel] poor sound from ALC 850/880 in alsa 1.0.21 and 1.0.23
by Colin Kincaid Williams 05 Dec '10

05 Dec '10

On Fri, Nov 26, 2010 at 8:35 PM, Daniel Chen <seven.steps(a)gmail.com> wrote: > Please use http://www.alsa-project.org/alsa-info.sh >

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[alsa-devel] netbook: pcspeaker doesn't work since 2.6.35 see attached alsa-info-files
by Halim Sahin 03 Dec '10

03 Dec '10

Hi, The subject sais it :-(. In alsamixer I can't umute or set any pc speaker related settings since 2.6.35. The control for pc speaker is gone :-(. Now I started the last working kernel 2.6.34, run alsa-info. Then repeated same with 2.6.36. can you please help to solve this? For accessibility software a working pc-speaker is really useful :-). Regards Halim

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[alsa-devel] Does anyone really have success experience with SPI codecs with new name-based SOC API?
by Alexander 02 Dec '10

02 Dec '10

Does anyone really have success experience with SPI codecs with new name-based SOC API? There are really few SPI codecs in ASoC subsystem, so I wonder if only I cannot bind my SPI codec to machine driver. The codec is registered with name spi0.0 it's the behavior of SPI subsystem, I suppose. So the form is %s%d.%d and not %s.%d as supposed in fmt_single_name() in soc_core.c For me dev_name(dev)==spi0.0 dev->driver->name==cs4271-codec Is this really working code? ----- strncpy(name, dev_name(dev), NAME_SIZE); /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); if (found) { /* get ID */ if (sscanf(&found[strlen(dev->driver->name)], ".%d", id) == 1) { /* discard ID from name if ID == -1 */ if (*id == -1) found[strlen(dev->driver->name)] = '\0'; } } else { -----

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[alsa-devel] Hardware events
by Rolf Theunissen 02 Dec '10

02 Dec '10

Hello, For some hardware devices (for instance the ThinkPad EC) the volume can be changed by (only) hardware buttons. Changes in the the hardware mixers are notified in the driver via the function "snd_ctl_notify" with the mask SNDRV_CLT_EVENT_MASK_VALUE. For notification of volume changes to the users of the system, an OSD can be used. To implement such a OSD, it must be possible to respond to changes done by the hardware. This raises the following question: is it possible to detect the events originating from the hardware in the functions that handle the callbacks (set via snd_mixer_elem_set_callback/snd_hctl_elem_set_callback)? This would allow to implement an OSD that only responds to the hardware buttons and that would ignore events originating from "software". Thanks

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[alsa-devel] [PATCH 0/5] ASoC: tpa6130a2: pop removal (mostly) changes
by Peter Ujfalusi 02 Dec '10

02 Dec '10

Hello, The following series improves the tpa driver to have better pop filtering capabilities. New stereo DAPM route can be used to reduce left/right power time differences. Machine drivers can opt to not use the built in DAPM routes, and use direct call to enable the stereo path. --- Peter Ujfalusi (5): ASoC: tpa6130a2: Simplify power state management ASoC: tpa6130a2: Defer SW enable from power enable ASoC: tpa6130a2: Use one event handler for PGA_E ASoC: tpa6130a2: Add stereo DAPM path ASoC: tpa6130a2: Make DAPM registration optional, and direct interface sound/soc/codecs/tpa6130a2.c | 77 ++++++++++++++++++++++++------------------ sound/soc/codecs/tpa6130a2.h | 3 +- 2 files changed, 46 insertions(+), 34 deletions(-) -- 1.7.3.2

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[alsa-devel] SND-SOC - how to test drivers ?
by Vaclav Peroutka 02 Dec '10

02 Dec '10

Hello all, I am trying to develop ALSA driver for audio subsystem for one Android mobile phone. I already have some messages in the kernel log - please see below. It means that I have s3c-i2s connected to MAX9880 (source code was not IMHO tested). And then the output from MAX9880 goes into unknown switching matrix ( for which I even don't have a datasheet - only register list and names of bits in registers - I can guess what purpose they serve to). What I want is to make some testbench driver and an application which can set switching matrix registers through IOCTL. In MAX9880 I want to preset microphone path to output and switch matrix until I hear my voice from microphone in speaker or headphones. Or, is there any better method how to do that ? AFAIK there is no alsamixer application for Android. What do you use for path and gain settings in this case ? And last question so far - where to look to fix this message: "Failed to add route LMICIN->Mic Input" ? All other routes were added so far. Thank you very much in advance, Vaclav <6>[ 2.932154] Advanced Linux Sound Architecture Driver Version 1.0.18a. <4>[ 2.953782] No device for DAI MAX9880 <4>[ 2.962009] No device for DAI s3c-i2s <4>[ 3.003635] I2S off <4>[ 3.019939] I2S off <4>[ 3.027948] Maxim MAX9880 Audio CODEC 0.12 <6>[ 3.054919] max9880: max9880_init <4>[ 3.063481] Entered s3c24xx_pcm_new <4>[ 3.071566] Entered s3c24xx_pcm_preallocate_dma_buffer <4>[ 3.080354] Entered s3c24xx_pcm_preallocate_dma_buffer <6>[ 3.088901] asoc: MAX9880 <-> s3c-i2s mapping ok <6>[ 3.097441] max9880: max9880_dapm_event 769 <6>[ 3.124311] max9880: max9880_dapm_event 0 <6>[ 3.140593] max9880: max9880_add_widgets <3>[ 3.156166] Failed to add route LMICIN->Mic Input <6>[ 3.164204] max9880: max9880_add_controls <6>[ 3.203234] ALSA device list: <6>[ 3.210962] #0: S3C6410 MAX9880 (MAX9880)

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