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 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]> 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 received this message 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