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

[alsa-devel] some questions about sound system
by loody 17 Mar '11

17 Mar '11

Dear all: I just install Ubuntu9.10 on my laptop and I try to update my sound system and recompile the utility. there are some questions I met: 1. when configure the utility it says checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found and config.log is "conftest.c:10:42: error: CoreFoundation/CFPreferences.h: No such file or directory conftest.c: In function 'main': conftest.c:14: error: 'NULL' undeclared (first use in this function) conftest.c:14: error: (Each undeclared identifier is repor" the above error seems I miss install some package before. I google the name but I have no improvement. "conftest.c:20:28: error: ac_nonexistent.h: No such file or directory configure:7397: $? = 1" the above error seems the header located where the conftest.c doesn't expect. how could solve them all? 2. there are some configure file in share when I compile advance linux sound architecture. like alsa.conf. have anyone know where I can find the naming rule? appreciate your help, miloody

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[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

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[alsa-devel] Problem with VIA VT1708S and git versions of alsa-driver.
by Mark Goldstein 31 Jan '11

31 Jan '11

Hello, Recently I'm having some strange issues with my VIA VT1708S on-board sound card. The description might be a bit long, so I'll give a brief summary first. I have ASUS P5QL/EPU motherboard with VT1708S 8-channel HDA Codec. There are 3 OSs installed on this machine: openSUSE 11.1 (my main system), openSUSE 11.3, Kubuntu 10.04. Audio worked fine until some updates at the beginning of September. Since then I have 2 different issues. 1).On 11.1 - Line In control seems inactive, instead Front Mic control controls the level from Line In jack. Front Mic is constanly disabled. alsaconf does not show hda-intel in the list of detected cards. Latest version it happened with is git20100925. When I compiled stable 1.0.23 sources, everything started working as expected again. I reported this issue on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5133 2) On oS 11.3 I start experiencing problems recording audio either from Mic or from Line. After some experiments I found out that the Capture1 control is somehow broken. Mute switch is reversed. When it is unmuted, there is no audio recorded. When it is muted, recording works. The same happened to at least one more user of openSUSE, see the thread "How to investigate the problem with MIC" on alsa-user mailing list. I've also added my comments to issue https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5130 but now I think it is different issue. Even when I have audio recorded normally when Capture1 is muted, I still have very distorted voice from Mic in Skype. It worked fine before the problem started. More details: On oS 11.1 I had my audio working fine after updating to 1.0.23 versions of alsa driver. I'm using opensuse/multimedia: repository that provides binary rpm of alsa-driver git snapshots. I have only one current kernel 2.6.27.48 on 11.1. On oS 11.3 audio worked right out of the box, since it uses kernels with alsa-driver 1.0.23. I have 2-3 kernels installed at the same time - normal 2.6.34 from official repository and 2.6.35 from kernel build service. I'm not sure when exactly the problems started, but definitely at the beginning of September. I've initially saw that on oS 11.1 I can't hear the audio from my TV card connected to Line In. Then I noticed that Mic does not work in Skype on both 11.1 and 11.3, while it still works on Kubuntu. When I unmuted Front Mic on 11.1 I've suddenly found that it actually enabled Line In. I think now that something went bad with card detecting. Probably on 11.1 with latest git versions my card is not properly detected and some "defaults" are used. I'm attaching here 2 outputs of alsa-info.sh - 11.1_stable captured with alsa-driver compiled from stable 1.0.23 sources. 11.1_git25 captured with driver compiled with git20100925 version from opensuse/multimedia: repository. For me the problem with 11.1 is resolved, but probably it's worth checking why latest version from git behave this way. I'll continue with 11.3 investigation later on. Regards, -- Mark Goldstein

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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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[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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[alsa-devel] [PATCHv3] ASoC: Overhaul Samsung drivers
by Jassi Brar 23 Nov '10

23 Nov '10

From: Jassi Brar <jassi.brar(a)samsung.com> Hi, Submitting revised patch series. Since v2: a) Divided 'ASoC Samsung Add common I2S driver' into two. b) 'ARM Samsung Rename s3c64xx I2S platform device' based on plat tree As Kgene agreed to prepare a branch for Mark, platform related changes are based on his tree wherever possible. Patches for Kgene will be posted on Arm-Linux ml and those for Mark Brown, on ALSA ml, now that people have already 'seen' two versions of the patch series with same code but different segregation. Here is overview of patches:- For Kgene:- ************ S5PV310: DMA: Add platform devices for PL330 DMACs ARM: Samsung: Rename s3c64xx I2S platform device ARM: Samsung: Prepare audio header for new I2S driver ARM: S3C64XX: I2S: Upgrade platform device ARM: S5P64X0: I2S: Upgrade platform device ARM: S5P6442: I2S: Upgrade platform device ARM: S5PC100: I2S: Upgrade platform device ARM: S5PV210: I2S: Upgrade platform device ARM: S5PV310: Add audio platform devices For Mark :- *********** ASoC: Samsung: Remove redundant AQUILA driver ASoC: Samsung: Rename DMA device ARM: Samsung: Define common audio-dma device ASoC: Samsung: Rename ASoC DMA driver ASoC: Samsung: Rename AC97 platform device ASoC: Samsung: Rename AC97 driver ASoC: Samsung: Rename PCM driver ASoC: Samsung: Generalize DMA driver namespace ASoC: Samsung: Rename s3c64xx I2S device ASoC: Samsung: Add common I2S driver ASoC: SMARTQ: Move to use new I2S driver ASoC: GONI: Move to use new I2S driver ASoC: SMDK64XX: Move to use new I2S driver ASoC: S3C64XX: Remove obsoleted I2S drivers ASoC: SMDK64XX: Rename for other platforms ASoC: SMDK_WM8580: Enable for SMDKC100 ASoC: Samsung: Generalize Kconfig symbols ASoC: Samsung: Rename from s3c24xx to samsung Thanks. ************************************************* Since v1: a) Use case 16 in set_bfs and default for error b) Better code comments. c) Use switch case in i2sv2_i2s_set_clkdiv d) Use SNDRV_PCM_RATE_8000_96000 e) MAX_I2S -> SAMSUNG_I2S_SECOFF f) Local namespace cleanup with i2s driver. ------------ Here come patches overhauling the Samsung ASoC support. The AC97, PCM, SPDIF and ASoC-DMA related code has merely undergone cosmetic changes - symbol, device & driver renaming and file moving. A new I2S controller driver has been added to manage Samsung SoCs since S3C64XX. Older driver has been changed only as is necessary. Regarding I2S, I have left the s3c24xx series untouched because the controllers are too different to run by a common driver. S3C2412's I2S does have some similarities, but not important enough to warrant impact on new driver design. The new I2S driver has been designed around the notion of features/quirks. The newer SoCs have revised version of I2S_0(usually) than previous ones. So, I2S block is 'specified' in the platform code and the driver works on that configuration. For now, Secondary I2S DAI works only with System-DMAC. I2S-Internal DMA and SRAM will be enabled later. Testing:- Simple testing of 5.1 channel and hardware mixing seems to work (though there seems to be some issue with symmetric_rates in soc-core.c which would be tried to fix later as saperate issue). I would appreciate any test-results for GONI and SMARTQ. I suggest the patches 13,14,16 & 17 go via Kgene's Tree, and the rest via ASoC tree due to tight dependencies. 1). ASoC: Samsung: Remove redundant AQUILA driver 2). ASoC: Samsung: Rename DMA device 3). ARM: Samsung: Define common audio-dma device 4). ASoC: Samsung: Rename ASoC DMA driver 5). ASoC: Samsung: Rename AC97 platform device 6). ASoC: Samsung: Rename AC97 driver 7). ASoC: Samsung: Rename PCM driver 8). ASoC: Samsung: Generalize DMA driver namespace 9). ASoC: Samsung: Rename s3c64xx I2S device 10). ASoC: Samsung: Add common I2S driver 11). ARM: S3C64XX: I2S: Upgrade platform device 12). ARM: S5P6440: I2S: Upgrade platform device 13). ARM: S5P6442: I2S: Upgrade platform device 14). ARM: S5PC100: I2S: Upgrade platform device 15). ARM: S5PV210: I2S: Upgrade platform device 16). ARM: S5PV310: Add audio platform devices 17). ASoC: SMARTQ: Move to use new I2S driver 18). ASoC: GONI: Move to use new I2S driver 19). ASoC: SMDK64XX: Move to use new I2S driver 20). ASoC: S3C64XX: Remove obsoleted I2S drivers 21). ASoC: SMDK64XX: Rename for other platforms 22). ASoC: SMDK_WM8580: Enable for SMDKC100 23). ASoC: Samsung: Generalize Kconfig symbols 24). ASoC: Samsung: Rename from s3c24xx to samsung

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[alsa-devel] HDA Analyzer - Dell M4500 Internal Mic
by Tom Burger 22 Nov '10

22 Nov '10

Hi all! My mic didn't work (Dell M4500) . Using HDA Analyzer I create a configuration that does work. I read that I could send my working codec to here. So here it is. Maybe you can also tell me how te make this change permanent? Good luck, Tom Diff for codec 0/0 (0x111d76d5): --- +++ @@ -100,16 +100,17 @@ Node 0x11 [Pin Complex] wcaps 0x400483: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x90a60130: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE + Pin-ctls: 0x20: IN Unsolicited: tag=0x00, enabled=0 Power: setting=D0, actual=D0 Node 0x12 [Vendor Defined Widget] wcaps 0xf00503: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 1 0x20 @@ -121,37 +122,37 @@ Delay: 13 samples Node 0x14 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Amp-Out vals: [0x79 0x79] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x15 [Audio Input] wcaps 0x1d0541: Stereo - Converter: stream=0, channel=0 + Converter: stream=1, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x17 Processing caps: benign=0, ncoeff=0 Node 0x16 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x18 Processing caps: benign=0, ncoeff=0 Node 0x17 [Audio Selector] wcaps 0x300d0d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 - Amp-Out vals: [0x0f 0x0f] + Amp-Out vals: [0x00 0x00] Power: setting=D0, actual=D0 Connection: 7 - 0x0c 0x0e 0x0f 0x1b 0x11 0x12 0x0a* + 0x0c 0x0e 0x0f 0x1b 0x11* 0x12 0x0a Node 0x18 [Audio Selector] wcaps 0x300d0d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Power: setting=D0, actual=D0 Connection: 7 0x0c 0x0e 0x0f 0x1b 0x11 0x12 0x0a* Node 0x19 [Audio Selector] wcaps 0x300501: Stereo Power: setting=D0, actual=D -- Tom Burger | tvburger(a)gmail.com | +31 (0)62 412 20 27 http://www.linkedin.com/in/tvburger

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[alsa-devel] About CS42L52---Where to find CS42L52 ASoC driver
by James Liu 19 Nov '10

19 Nov '10

Hello ,guys where to find CS42L52 ASoC driver? I have cloned the alsa-kernel.git and alsa-asoc.git tree,but no CS42L52 device driver. Please tell me the download site. Thanks James Liu

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[alsa-devel] [RFC i-iv] ASoC: Add support for cross-device paths without dai and without name collision
by Jarkko Nikula 05 Nov '10

05 Nov '10

Hi I'm sending here four set of patches that are build on top of another. They are trying to solve issues with following setup: CPU <-> [Codec x.0] -> [Codec x.1] -> out Here only Codec x.0 has DAI link between codec and CPU, Codec x.1 is used as an amplifier and the same codec driver is used for both codec instances. Currently this setup is not possible because ASoC doesn't support cross-device paths, i.e. DAPM is per codec, there is no support to register dailess codecs and no resolution for name collisions that would occur when multiple codec instances tries to register same kcontrols and when trying to build paths between widgets. I've separated patches to these problems to separate sets so that reviewing would be easier. However, they depend on previous sets so they must be applied sequentially and thus each set is named as RFC_[i-iv]/iv. I've based this work on of top of Liam's DAPM decoupling patches. Those must be squashed together but I'm sending here them separately because they include core change from Liam and trivial conversions and minor fixes from me. i Decoupling DAPM from codecs. Core implementation from Liam Girdwood and some minor bits from me, conversion of new codecs and all the platforms. I've build tested the codecs and OMAP platforms and if some platform doesn't build, it's my fault. ii Extend ASoC core to handle cross-device paths. Decoupling paths and widgets from DAPM context and handling cross-device bias and widget power changes. iii Support for dailess codecs. This is achieved by renaming dai_link as dev_map, i.e. the former struct snd_soc_dai_link becomes struct snd_soc_dev_map so it does not only describe platform, cpu dai and codec binding but also registration of dailess codecs. iv Optional name_prefix to struct snd_soc_dev_map. With this a machine driver can specify a name that is used to prefix codec kcontrols, widgets and internal route names for avoiding name collision that would occur if multiple codec instances are registered. -- Jarkko

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[alsa-devel] Balanced output support for Xonar: Patches for Virtuoso driver to add mixer option for balanced mono output (PCM179x dacs)
by Christian Wisner-Carlson 05 Nov '10

05 Nov '10

This patch adds a mixer switch for soundcards using the snd-virtuoso driver and pcm1796/pcm1792(a) DAC(s) which switches between Stereo, Balanced Left, and Balanced Right output modes. The PCM179x series has a "Monaural Mode" that outputs a single channel in differential (and therefore balanced, because of the buffer circuit) mode, which boosts the SNR and allows for common mode rejection (with the proper cable). Simply use the left output as the noninverting signal and the right output as the inverting output. The datasheets for both the pcm1792a and pcm1796 provide application suggestions, which boil down mainly to "connect an XLR jack to the outputs of the normal application circuit". I tested this thoroughly with an Asus Xonar Essence ST, although it should also work with the HDAV, Essence STX, and probably other cards. I get no noticeable interference when running balanced audio from the card over approximately 150ft of unshielded twisted pair cable which I have running through my walls. The driver defaults to stereo output and this patch will not cause issues for users who do not need/want this feature. The patches that I am posting only give one control regardless of whether or not more than one DAC is connected. I have another patchset that gives individual control of all 4 dacs in the Xonar Essence ST+H6 and HDAV+H6 setups (H6 is an optional daughterboard for surround sound). It allows for 4 balanced outputs, 8 unbalanced outputs, or any combination of balanced and unbalanced outputs. HOWEVER, it adds 4 mixer controls and the code isn't very pretty. If people want it, I'll post it as well and/or take suggestions on how to improve it. Here is a diff of xonar_pcm179x.c, which is the only changed file. The diff is against alsa-driver 1.0.23 but should cleanly patch the GIT revision. Please tell me if this is not the correct way to post a patch, and tell me if it seems like a candidate to be merged. Christian Wisner-Carlson --- ./alsa-driver-1.0.23/alsa-kernel/pci/oxygen/xonar_pcm179x.c 2010-04-16 07:10:10.000000000 -0400 +++ ./anothermono/alsa-driver-1.0.23/alsa-kernel/pci/oxygen/xonar_pcm179x.c 2010-10-30 05:28:55.000000000 -0400 @@ -176,6 +176,8 @@ unsigned int current_rate; bool os_128; bool hp_active; + bool monomode; + bool rightmono; s8 hp_gain_offset; bool has_cs2000; u8 cs2000_fun_cfg_1; @@ -532,6 +534,13 @@ reg = PCM1796_OS_64; else reg = PCM1796_OS_32; + + if (data->monomode) + reg = reg | PCM1796_MONO; + + if (data->rightmono) + reg = reg | PCM1796_CHSL_RIGHT; + for (i = 0; i < data->dacs; ++i) pcm1796_write_cached(chip, i, 20, reg); } @@ -719,6 +728,19 @@ return 0; } +static int monomode_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { "Stereo", "Balanced Left", "Balanced Right" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 0; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + static int os_128_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -729,6 +751,18 @@ return 0; } +static int monomode_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + int newvalue; + newvalue = data->monomode + (data->monomode & data->rightmono); + + value->value.enumerated.item[0] = newvalue; + return 0; +} + static int os_128_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { @@ -751,6 +785,25 @@ return changed; } +static int monomode_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + int changed; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != + (data->monomode + (data->monomode & data->rightmono)); + if (changed) { + data->monomode = (value->value.enumerated.item[0]) ? 1 : 0; + data->rightmono = (value->value.enumerated.item[0] == 2) ? 1 : 0; + update_pcm1796_oversampling(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + static const struct snd_kcontrol_new os_128_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DAC Oversampling Playback Enum", @@ -759,6 +812,14 @@ .put = os_128_put, }; +static const struct snd_kcontrol_new monomode_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Mono Mode Playback Enum", + .info = monomode_info, + .get = monomode_get, + .put = monomode_put, +}; + static int st_output_switch_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { @@ -932,6 +993,9 @@ err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); if (err < 0) return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&monomode_control, chip)); + if (err < 0) + return err; return 0; } I am also in the process of adding a Phase Invert switch and (possibly) Deemphasis control for the PCM179x DACs as well, and will post patches if anyone is interested.

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