Hi Pablo, Sure, that should be no problem.
My files are actually already online. The license for it is CC-BY-SA (https://creativecommons.org/licenses/by-sa/4.0/legalcode). I need to update the readme to make this clear though. This means you’re free to use it, as long as proper attribution is done. The repository is https://github.com/LCAV/CompactSix I used Eagle CAD to design the board. Let me know if you have any question. Cheers, Robin > On 27 mars 2016, at 01:22, Pablo Fonovich <[email protected]> wrote: > > Hi Robin: > > After working hard, i'm sure that i'm having some hardware problems, cause > driver seems to work but get no output. This led me to trying to communicate > with the codec via i2c and i could not... So either my chip is dead or i'm > having hardware trouble that i could not detect in the whole past week... > > So i had the idea of using the same codec you did, to get something to > work... if it's not much trouble, could you share with me the schematics of > your hardware? and could i use the drivers you wrote? > > It's very frustrating working so much and having hardware fails... > > another thing i wanted to ask, do you know if alsamixer may not work if i2c > communication is not possible? Cause i always get I/O error... although i > wrote al needed (i think) structures > > Thanks for everything > > Pablo > > 2016-02-13 12:59 GMT-03:00 Pablo Fonovich <[email protected] > <mailto:[email protected]>>: > Hi robin! > > By this time, i was writting a guide to get a custom linux os to boot in the > beaglebone... and i was quite busy with that... > > In my project, the drivers seems to be working properly, although i can't use > regmap_update_bits function... i'm not sure, but i think it has something to > do with the i2c communications that seems not to be working... If it has to > update registers through i2c and this is not working, it makes sense that > that functions returns error... i think that for that same reazon, alsamixer > is not working... i've made the structures you told me, but i keep getting > "input output error" when i try to "alsamixer" > > Anyway, the university is closed till march here (we are on holydays) so i > can't get the equipmet to test the hardware, and i don't own any right now > (too expensive for me). > > One main issue i'm having and that i can't solve is that i can record only 4 > channels with arecord... > I configured 2 serializers as input, and 4 tdm slots on each in the device > tree... but if i "arecord -c 6" it says "no channel count avaliable"... i'm > starting to think that arecord recors from only one serializer... i would > need to record an arbitrary channel if it's possible, and i think i should > write an alsa application for this... > > But more important right now seems to be this: Do you think is possible that > alsamixer and regmap_update_bits are not working cause of i2c problem? > > Thanks for everything > Pablo > > > > 2016-02-12 12:17 GMT-03:00 Robin Scheibler <[email protected] > <mailto:[email protected]>>: > Hi Pablo, > > Any progress with your board ? > > Also, to answer your question, I got the alsamixer working. You have to add a > number of macros for predefined controls corresponding to register to update > in the codec. You can see it directly in my code: > > https://github.com/fakufaku/linux/blob/66e5bfa74df723ff64b48d2ae14f3543d6c47861/sound/soc/codecs/sta321mp.c#L74 > > <https://github.com/fakufaku/linux/blob/66e5bfa74df723ff64b48d2ae14f3543d6c47861/sound/soc/codecs/sta321mp.c#L74> > > > static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); > static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); > static const struct snd_kcontrol_new sta321mp_snd_controls[] = { > SOC_SINGLE("Master Switch", STA321MP_MMUTE, 0, 1, 1), > SOC_SINGLE_TLV("Master Volume", STA321MP_MVOL, 0, 0xff, 1, mvol_tlv), > SOC_DOUBLE_R("Ch1 Capture Switch", STA321MP_C1VTMB, STA321MP_C2VTMB, 7, 1, 1), > SOC_DOUBLE_R("Ch2 Capture Switch", STA321MP_C3VTMB, STA321MP_C4VTMB, 7, 1, 1), > SOC_DOUBLE_R("Ch3 Capture Switch", STA321MP_C5VTMB, STA321MP_C6VTMB, 7, 1, 1), > SOC_DOUBLE_R_TLV("Ch1 Capture Volume", STA321MP_C1VOL, STA321MP_C2VOL, 0, > 0xff, 1, chvol_tlv), > SOC_DOUBLE_R_TLV("Ch2 Capture Volume", STA321MP_C3VOL, STA321MP_C4VOL, 0, > 0xff, 1, chvol_tlv), > SOC_DOUBLE_R_TLV("Ch3 Capture Volume", STA321MP_C5VOL, STA321MP_C6VOL, 0, > 0xff, 1, chvol_tlv), > }; > > This implements volume control for all six capture channels, as well as mute > switches. > > Best, > Robin > > > >> On 12 janv. 2016, at 21:50, Pablo Fonovich <[email protected] >> <mailto:[email protected]>> wrote: >> >> Hi Robin: >> >> I realized that although i was commenting the line you said, i was using and >> old image of the compiled kernel, and that was why i kept getting the same >> error! My bad... Now that is working, but i have an error in the line: >> >> ret = regmap_update_bits(priv->regmap, PCM3168_DAC_FMT_CONTROL, 0x0f, val); >> >> on my pcm3168.c file... ret gets the value -128 (error code). I use this >> function to update some register on the codec... i suspected that my i2c >> connection is not working properly, so i commented this line (althoug i'm >> not sure if an error in i2c causes this function to crash) and finally the >> arecord command works gives no error, but records nothing :( >> >> That was somehow expected, as my codec is not geting it's regmap updated... >> One of this days i will use an ozscope to see if the signals out of the i2c >> of the beaglebone are right... If you came up with some idea please tell >> me... you've been of great help... >> >> P.D: have you got alsamixer working in your project? >> >> 2016-01-11 17:06 GMT-03:00 Pablo Fonovich <[email protected] >> <mailto:[email protected]>>: >> Hi: >> Thanks for your suggestions! >> i've already commented that line, as i said in the mail when i isolated the >> problem to that function... the error persist but now in: >> >> ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT); >> if (ret < 0) >> return ret; >> >> I'm using the linux-cross-reference, that's how i find out what each >> parameter means, but perhaps i still dont quite understand what the function >> really does. >> >> I've also read the mcasp section of the am335x datasheet when i was doing >> the PCB for the codec, but i should re-read it, as many things where not >> clear for me at that moment, and perhaps now they make a little more sense. >> Anyway, the problem and same error code persists, now in that other line... >> >> Thanks for your help! i'll post any progress i make >> >> >> 2016-01-11 5:12 GMT-03:00 Robin Scheibler <[email protected] >> <mailto:[email protected]>>: >> Hi Pablo, >> >> I took a look at your code. >> >>> static int pcm3168_hw_params(struct snd_pcm_substream *substream, >>> struct snd_pcm_hw_params *params) >>> { >>> struct snd_soc_pcm_runtime *rtd = substream->private_data; >>> struct snd_soc_dai *codec_dai = rtd->codec_dai; >>> struct snd_soc_dai *cpu_dai = rtd->cpu_dai; >>> struct snd_soc_codec *codec = rtd->codec; >>> struct snd_soc_card *soc_card = rtd->card; >>> struct platform_device *pdev = to_platform_device(soc_card->dev); >>> int ret = 0; >>> unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) >>> snd_soc_card_get_drvdata(soc_card))->sysclk; >>> >>> /* set the codec system clock */ >>> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); >>> if (ret < 0) >>> return ret; >>> >>> /* set the CPU system clock */ >>> ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT); >>> if (ret < 0) >>> return ret; >>> >>> return 0; >>> } >> >> I see that you are calling snd_soc_dai_set_sysclk for both codec_dai and >> cpu_dai. I think that is the error. In your case codec_dai is the PCM3168 >> chip and cpu_dai is the processor McASP module. Sysclk only makes sense for >> McASP as far as I understand. Try commenting out >> >>> /* set the codec system clock */ >>> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); >>> if (ret < 0) >>> return ret; >> >> I would also suggest you read up about the McASP clock generation structure >> in the sitara documentation (~7000 pages document available from TI). This >> helped me understand how the McASP module works. >> >> Another resource I used is the linux cross-reference >> http://lxr.free-electrons.com/ <http://lxr.free-electrons.com/> >> I used it to look up where functions are defined and what they do. >> >> Cheers, >> Robin >> >> >>> On 10 janv. 2016, at 18:02, Pablo Fonovich <[email protected] >>> <mailto:[email protected]>> wrote: >>> >>> Hi: im using 24Mhz, that is provided by de beaglebone... i used printk to >>> see that sysclk value and it is correct... i get the value as is stated in >>> the evm_hw_params function. >>> I dont see why the function returns error... but is there a list of error >>> codes to see what -524 means? >>> Thanks very much >>> >>> El ene 10, 2016 1:55 PM, "Robin Scheibler" <[email protected] >>> <mailto:[email protected]>> escribió: >>> Hi Pablo, >>> >>>> On 10 janv. 2016, at 01:48, Pablo Fonovich <[email protected] >>>> <mailto:[email protected]>> wrote: >>>> >>>> Hi Robin: >>>> with printk statements, i've isolated the error to this instruction: >>>> >>>> ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); >>>> >>>> but i don't know what is wrong... the sysclk variable is: >>>> >>>> unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *) >>>> snd_soc_card_get_drvdata(soc_card))->sysclk; >>>> Just linke in the evm_hw_params >>> >>> What value of sysclk are you providing ? >>> >>>> >>>> I've noticed that you used: >>>> >>>> ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_IN); >>>> >>>> If i'm getting it right, this is to set the cpu_dai sysclock, but i dont >>>> get why you used 0 in the frequency parameter. Also you never used the >>>> function to set the codec_dai sysclk... Is it not necessary? I'll try >>>> commenting this line >>> >>> In my case, MCLK is provided to the chipset by an external crystal. Thus >>> there is no sysclk for me. BCLK and LRCLK are also provided by the chipset >>> to McASP. >>> >>> Best, >>> Robin >>> >>>> >>>> >>>> >>>> >>>> 2016-01-09 11:55 GMT-03:00 Pablo Fonovich <[email protected] >>>> <mailto:[email protected]>>: >>>> Hi Robin: >>>> I intend to implement a portable multitrack recorder, that's why i choose >>>> this codec, it has many inputs and outputs, good sample rate and bit >>>> resolution, and all in a single chip. >>>> I can't find wendelin's mail direction, so i think i'll just write to that >>>> posts, but they are a bit old i think. I have already looked at them, but >>>> i was more confused by that time, perhaps now i can understand a bit more. >>>> >>>> If i find out anything i will continue posting here, so if someone is >>>> interested in this project can look at ti... >>>> >>>> Please if you find any documentation tell me! >>>> >>>> Thanks >>>> >>>> 2016-01-08 19:51 GMT-03:00 Robin Scheibler <[email protected] >>>> <mailto:[email protected]>>: >>>> Maybe also this can help: >>>> http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059433.html >>>> >>>> <http://mailman.alsa-project.org/pipermail/alsa-devel/2013-February/059433.html> >>>> >>>> R >>>> >>>>> On 8 janv. 2016, at 23:49, Robin Scheibler <[email protected] >>>>> <mailto:[email protected]>> wrote: >>>>> >>>>> Hi Pablo, >>>>> >>>>> I don’t know for 4.2. Try to read the >>>>> sound/alsa/soc/davinci/davinci-mcasp.c file to understand which >>>>> functionalities are implemented. I used the kernel from Robert C Nelson’s >>>>> repo on github. >>>>> >>>>> After a quick search on google, I saw you are not the first person to >>>>> attempt to connect this codec to the BBB. >>>>> http://www.spinics.net/lists/alsa-devel/msg41291.html >>>>> <http://www.spinics.net/lists/alsa-devel/msg41291.html> >>>>> Maybe you can try to contact this Wendelin Klimann. >>>>> >>>>> BTW, this PCM3168 seems like a very neat chip. What exactly are you >>>>> building ? >>>>> >>>>> Cheers, >>>>> Robin >>>>> >>>>>> On 8 janv. 2016, at 23:17, Pablo Fonovich <[email protected] >>>>>> <mailto:[email protected]>> wrote: >>>>>> >>>>>> Thanks Robin, >>>>>> Im using latest stable kernel, think is 4.2, i can check when i get >>>>>> home... i will add printk statements, as you suggested... >>>>>> Another question. I inted to use one mcasp serializer for 4 channels... >>>>>> in the am335x datasheet it seems possible, so i configured 4 tdm slots >>>>>> in the device tree instead of 2, but if i try to record with -c 6 it >>>>>> says again the "channels count non available". Do you know if its >>>>>> possible? its quite frustrating to work without propper documentation, >>>>>> but in need this project to get my title... >>>>>> >>>>>> And last thing, where you able to get alsamixer working? >>>>>> Thanks very much >>>>>> >>>>>> 2016-01-08 19:03 GMT-03:00 Robin Scheibler <[email protected] >>>>>> <mailto:[email protected]>>: >>>>>> Hi Pablo, >>>>>> >>>>>> What is the kernel version you are using ? I have finally used kernel >>>>>> 3.14 because in 3.8 it seems that the mcasp driver implementation could >>>>>> not support more than two channels. >>>>>> >>>>>> I was in the same position as you. I mostly dug as much as possible from >>>>>> TI documentation and also read the drivers for other chipsets in the >>>>>> kernel code. Then, I also used >>>>>> the sitara am355xx data sheet to understand how McASP works. It took me >>>>>> quite some time to piece together how things work. Still now I do not >>>>>> fully understand. The problem is that there is no proper documentation >>>>>> for the ALSA SoC wrt embedded platforms such as the BBB. >>>>>> >>>>>> Note that the TI example code might not be for exactly the same kernel >>>>>> version you are using. There might be discrepancies. >>>>>> >>>>>> I suggest you run a search on the source code to find where the error >>>>>> “ASoC: machine hw_params failed” is generated. You can add printk >>>>>> statements in your code so that you can later check in the kernel log >>>>>> that things are happening as you expect. >>>>>> >>>>>> Cheers, >>>>>> Robin >>>>>> >>>>>>> On 8 janv. 2016, at 01:25, Pablo Fonovich <[email protected] >>>>>>> <mailto:[email protected]>> wrote: >>>>>>> >>>>>>> sorry, the message also states: >>>>>>> >>>>>>> arecord: set_params:1297: Unable to install hw params: >>>>>>> ACCESS: RW_INTERLEAVED >>>>>>> FORMAT: S24_LE >>>>>>> SUBFORMAT: STD >>>>>>> SAMPLE_BITS: 32 >>>>>>> FRAME_BITS: 128 >>>>>>> CHANNELS: 4 >>>>>>> RATE: 96000 >>>>>>> PERIOD_TIME: (21333 21334) >>>>>>> PERIOD_SIZE: 2048 >>>>>>> PERIOD_BYTES: 32768 >>>>>>> PERIODS: 4 >>>>>>> BUFFER_TIME: (85333 85334) >>>>>>> BUFFER_SIZE: 8192 >>>>>>> BUFFER_BYTES: 131072 >>>>>>> TICK_TIME: 0 >>>>>>> # aplay -f S24_LE -r 96000 ../Yam*wav >>>>>>> Warning: format is changed to S16_LE >>>>>>> Playing WAVE '../Yamaha-TG100-Ocarina-C5.wav' : Signed 16 bit Little >>>>>>> Endian, Rate 44100 Hz, Stereo >>>>>>> [ 102.291787] davinci_evm sound: ASoC: machine hw_params failed: -524 >>>>>>> aplay: set_params:1297: Unable to install hw params: >>>>>>> ACCESS: RW_INTERLEAVED >>>>>>> FORMAT: S16_LE >>>>>>> SUBFORMAT: STD >>>>>>> SAMPLE_BITS: 16 >>>>>>> FRAME_BITS: 32 >>>>>>> CHANNELS: 2 >>>>>>> RATE: 44100 >>>>>>> PERIOD_TIME: (125011 125012) >>>>>>> PERIOD_SIZE: 5513 >>>>>>> PERIOD_BYTES: 22052 >>>>>>> PERIODS: 4 >>>>>>> BUFFER_TIME: (500045 500046) >>>>>>> BUFFER_SIZE: 22052 >>>>>>> BUFFER_BYTES: 88208 >>>>>>> TICK_TIME: 0 >>>>>>> >>>>>>> >>>>>>> 2016-01-07 21:24 GMT-03:00 Pablo Fonovich <[email protected] >>>>>>> <mailto:[email protected]>>: >>>>>>> Thanks very much, that was a problem. But now another error appears: >>>>>>> "davinci_evm sound: ASoC: machine hw_params failed: -524". Do you know >>>>>>> what it means, or where can i find the meaning? I assume this means >>>>>>> there is something wrong when i created the pcm3168_hw_params in the >>>>>>> alsa machine layer... but i don't know the problem... where did you >>>>>>> look for info? only in that TI guide? I attach my files, and if is not >>>>>>> much trouble for you i would be grateful if you help me... >>>>>>> >>>>>>> i've noticed that some functions you used in davinci_evm.c where not >>>>>>> explained in that guide... and it's very difficult for a noob like me >>>>>>> in this subject for understainding what to do... I've just based my >>>>>>> pcm3168.c in another driver, and i really don't quite understand much >>>>>>> of what is going on, nor found any simple documentation about it. If >>>>>>> you could give me some hints it would be great! >>>>>>> >>>>>>> Thanks very much.. >>>>>>> >>>>>>> 2016-01-07 17:30 GMT-03:00 Robin Scheibler <[email protected] >>>>>>> <mailto:[email protected]>>: >>>>>>> Hi Pablo, >>>>>>> >>>>>>> With McASP it is not possible to capture less than 2 channels. Try with >>>>>>> -c 2 argument. >>>>>>> >>>>>>> Best, >>>>>>> Robin >>>>>>> >>>>>>>> On 7 janv. 2016, at 20:16, [email protected] >>>>>>>> <mailto:[email protected]> wrote: >>>>>>>> >>>>>>>> Hi Robin: >>>>>>>> >>>>>>>> I'm doing a similar project, but with a PCM3168 codec... I also >>>>>>>> followed the guide >>>>>>>> http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example >>>>>>>> <http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example> >>>>>>>> Anyway there are lots of things that are not very clear for me... >>>>>>>> <http://processors.wiki.ti.com/index.php/Sitara_Linux_Audio_DAC_Example> >>>>>>>> >>>>>>>> In my case, i use codec as clock slave, and synch, so the problem you >>>>>>>> had should not affect to me... But when i try to record it says: >>>>>>>> >>>>>>>> arecord -D hw:0,0 -f S24_LE -c 1 -r 96000 -t wav test.wav >>>>>>>> Recording WAVE 'test.wav' : Signed 24 bit Little Endian, Rate 96000 >>>>>>>> Hz, Mono >>>>>>>> arecord: set_params:1239: Channels count non available >>>>>>>> >>>>>>>> Could you help me? >>>>>>>> >>>>>>>> On Sunday, May 24, 2015 at 4:38:43 AM UTC-3, Robin Scheibler wrote: >>>>>>>> Dear all, >>>>>>>> >>>>>>>> I have created a custom cape around the STA321MP >>>>>>>> <http://www.st.com/web/en/catalog/sense_power/FM125/CL935/SC534/PF251500d.d24> >>>>>>>> chip from ST Microelectronics. It has 6 MEMS microphones and outputs >>>>>>>> them on an I2S bus with 3 data lines. I am trying to use mcasp0 on the >>>>>>>> boneblack side to receive the data. I am now to the point where the >>>>>>>> codec configuration happens correctly (I can see the i2s signals with >>>>>>>> a logic analyzer) and a soundcard is detected. However, when I try to >>>>>>>> record sound with arecord, I get the following error: >>>>>>>> >>>>>>>> root@beaglebone:~# arecord -l >>>>>>>> **** List of CAPTURE Hardware Devices **** >>>>>>>> card 0: EVM [STA321MP EVM], device 0: STA321MP sta321mp-audio-0 [] >>>>>>>> Subdevices: 1/1 >>>>>>>> Subdevice #0: subdevice #0 >>>>>>>> root@beaglebone:~# arecord -D hw:0,0 -f S24_LE -c 6 -r 44100 -t wav >>>>>>>> test.wav >>>>>>>> Recording WAVE 'test.wav' : Signed 24 bit Little Endian, Rate 44100 >>>>>>>> Hz, Channels 6 >>>>>>>> arecord: pcm_read:1801: read error: Input/output error >>>>>>>> >>>>>>>> Let me know provide more details. >>>>>>>> The codec is frame and bit clocks master >>>>>>>> The connection to mcasp0 is the following: >>>>>>>> Bit clock <=> P9_12 (mcasp0_aclkr_mux3) >>>>>>>> Frame sync clock (L/R) <=> P9_27 (mcasp0_fsr) >>>>>>>> Data <=> P9_28 (mcasp0_axr2) >>>>>>>> Data <=> P9_30 (mcasp0_axr0) >>>>>>>> Data <=> P9_25 (mcasp0_axr3) >>>>>>>> I am using kernel 3.14.41 (because 3.8 doesn't have support for mcasp >>>>>>>> with >2 channels) >>>>>>>> I am using dtb-builder to add my device to the device tree >>>>>>>> All the code I have added to the kernel (codec and machine code) and >>>>>>>> to the device tree is on github. Here are the links to the specific >>>>>>>> changes I have made. >>>>>>>> >>>>>>>> kernel: >>>>>>>> https://github.com/fakufaku/linux/commit/3a6680960412ef0fc3fbcc6f2c0021350039cc92 >>>>>>>> >>>>>>>> <https://github.com/fakufaku/linux/commit/3a6680960412ef0fc3fbcc6f2c0021350039cc92> >>>>>>>> device tree: >>>>>>>> https://github.com/fakufaku/dtb-rebuilder/commit/127dbf3c82c78cc602058308ed72978b170c99a6 >>>>>>>> >>>>>>>> <https://github.com/fakufaku/dtb-rebuilder/commit/127dbf3c82c78cc602058308ed72978b170c99a6> >>>>>>>> It is my first time trying to code audio driver so please be indulgent >>>>>>>> :) I have spent a lot of time on this but it seems I am now completely >>>>>>>> stuck. I'd really appreciate some help. I'd be happy to provide more >>>>>>>> clarifications on anything. >>>>>>>> >>>>>>>> One thing I do not understand is if I need to add mixer bindings. And >>>>>>>> also the audio routing (jack vs codec pins, etc). >>>>>>>> >>>>>>>> Thank you in advance! >>>>>>>> Robin >>>>>>>> >>>>>>>> -- >>>>>>>> For more options, visit http://beagleboard.org/discuss >>>>>>>> <http://beagleboard.org/discuss> >>>>>>>> --- >>>>>>>> You received this message because you are subscribed to a topic in the >>>>>>>> Google Groups "BeagleBoard" group. >>>>>>>> To unsubscribe from this topic, visit >>>>>>>> https://groups.google.com/d/topic/beagleboard/pptZ_9PTqMY/unsubscribe >>>>>>>> <https://groups.google.com/d/topic/beagleboard/pptZ_9PTqMY/unsubscribe>. >>>>>>>> To unsubscribe from this group and all its topics, send an email to >>>>>>>> [email protected] >>>>>>>> <mailto:[email protected]>. >>>>>>>> For more options, visit https://groups.google.com/d/optout >>>>>>>> <https://groups.google.com/d/optout>. >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> For more options, visit http://beagleboard.org/discuss >>>>>>> <http://beagleboard.org/discuss> >>>>>>> --- >>>>>>> You 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