I am trying, and failing, to connect an I2S microphone (Invensense ICS43432) to my Raspberry Pi (B+) running Arch Linux. I have asked for specific advice in the relevant Arch Linux ARM forum but my question is really more general than that: how does one go about debugging Linux audio input issues?
I have verified with a logic analyser that the I2S microphone is sending sensible data in the correct channel (left) and the correct pin of the Raspberry Pi. The I2S microphone appears under ALSA as a "sound card". arecord is perfectly happy to record from that device and I have boosted the gain of that device using alsamixer by 30 dB. Yet all the data bytes of the recorded file are zero.
How does one go about checking the flow of audio data, the operation of DMA, under Linux?
I had the same problem trying to record in stereo, using 2 Adafruit I2S MEMS breakout mics: arecord worked fine, but zeros when using ALSA to write to a bin file. Choosing a 32 bit word format (Little Endian 32 bits, Signed) made it work. Only I end up with 64 bit stereo Frames.
Related
During the installation, we are supposed to check the sound card by pressing ‘arecord -L’ to obtain a certain output like shown below,
pi#raspberrypi:~ $ arecord -L
null
Discard all samples (playback) or generate zero samples (capture)
jack
JACK Audio Connection Kit
pulse
PulseAudio Sound Server
default
playback
ac108
sysdefault:CARD=seeed4micvoicec
seeed-4mic-voicecard,
Default Audio Device
dmix:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct sample mixing device
dsnoop:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct sample snooping device
hw:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Direct hardware device without any conversions
plughw:CARD=seeed4micvoicec,DEV=0
seeed-4mic-voicecard,
Hardware device with all software conversions
usbstream:CARD=seeed4micvoicec
seeed-4mic-voicecard
USB Stream Output
usbstream:CARD=ALSA
bcm2835 ALSA
USB Stream Output
However, the output that I have received is as shown below,
Screenshot of Output
It basically shows that I don’t have the ALSA soundcard, and I cant move on to the sound localization process. Please show how can I move forward, thanks!
We have a setup with a Windows 7 machine where we installed Dante Virtual Soundcard and start that soundcard with ASIO capabilities. The soundcard will receive audio over the network from a Tesira server. We want to capture the audio to files (highly preferring each channel to a separate file). The files will be played back on a later moment. There will likely be 6 channels or more.
In the same setup we use ffmpeg to capture some video which is working fine, with Direct Show. So for audio we wanted to use the same setup, since ffmpeg is able to record audio as well. However, there seems to be no option to select the ASIO devices which the virtual soundcard probably creates. So the question is what command line to use for ffmpeg, or what to install? Or which other program can record ASIO from command line?
I already tried installing:
Asio4all (actually wrong way around)
sox (don't know why actually)
HiFi Cable Asio Bridge (from VB-audio, not enough channels even with donate version)
Voicemeeter (from VB-Audio, not enough channels and actually mixes down)
O Deus Asio link, this might be an interesting option but it did not let me configure any route, any suggestions?
One thing I noticed is that the virtual soundcard can also be set to use WDM. Then I can see the devices with ffmpeg -list_devices true -f dshow -i duymmy, but recording does not yield any result, I have to ctrl-c to make it stop instead of q, and the file is zero bytes. Supposedly this is because the data over the network is all ASIO formatted and the Tesira Server cannot send "WDM data". FFmpeg stops at selecting the capture pin for audio only
EDIT:
I ran ffmpeg with high verbosity and when selecting the WDM soundcard it stops at Selecting pin Capture on audio only. Also when requesting the options it gives the same line for 22 times: min ch=1 bits=8 rate= 11025 max ch=2 bits=16 rate= 44100
You might use Voicemeeter instead of HIFI-Cable / ASIO-Bridge. Voicemeeter is a virtual audio device mixer able to connect everything together, any audio point, in any interface and any app together (including ASIO DAW)... Download & User Manual on www.voicemeeter.com
To answer my own question: it is not possible to capture sound from an ASIO device with ffmpeg. Maybe I will write the code for it if I need it...
I could however solve my issues by separating the two streams of audio data we have (AVB and Dante). These where on the same switch and maybe it is a bug in the firmware, maybe misconfiguration.
Thanks for your help!
How do I get the output from an ASIO device to IceCast2 or FFMpeg?
Duplicate?
And if not, Place the output for ffmpeg -f dshow -i "audio=your_device_name_in_dshow" -list_options
I am working with digital TV in Linux platform. Currently I am facing with one issue in audio. When I give stereo audio to
snd_pcm_write_i
Function and after long time running the audio channels get swapped. That is, right channel audio hearing in Left channel and Left in Right. I dumped the PCM data in to a file before giving to alsa in issue case and played using 'aplay' and audio is good.So I think the PCM data is OK. In my system,'AK4643' audio codec device is used. Does any one faces this issue? If so please help me.
The issue was associated with the I2S driver .
Fixed the issue with updated driver from chip vendor.
I am bit stuck, how can I make my arduino record into .wav files?
The arduino is connected with a microphone, and am using the Arduino ADC.
Any ideas? Will I be able to play them back using my pc?
many question cross my head
1- Is this possible using an arduino Uno
2- Is this possile using just a microphone connected to the Arduino ADC
3- if yes how can i get the wav format.
The idea gonna be like this
Ardiuno microphone-->Uno ADC -->arduino (library making wav sound)--> Storing data to a an SD card connected via SPI or maybe (connecting a Raspberry as a storage device)
also another question:
4- Do I need an amplifier due to the act that analog output from the microphone is very weak so the ADC couldn't detect the variation
In another log i had seen that i should connect the microphone to a level shifter.And that cause of the analog output is AC so i have to make the negative wave as 0 (for 10 it ADC)
the zero point as 512 and the positive as 1024 (10 bit ADC).(really i'm not sure about this part)
doing some research i got this library "https://github.com/TMRh20/TMRpcm/wiki/Advanced-Features#recording-audio" which is supposed to do the job, I mean making some wav file from the analog input.
So any help would be appreciated
Thx in advance,
Salah Laaroussi
Yes, although a bit complex it is very possible to do this via an uno.
The biggest hurdles to overcome is the limited amount of RAM and the clock speed. You will have to setup twin buffers to handle writing to the SD card. Make sure the card has a high enough write speed or the entire program will come to a screeching halt as you will run out of memory.
apc mag has a great article detailing out the circuit and code.
http://apcmag.com/arduino-projects-digital-audio-recorder.htm/
There are many things you haven't prepared yet:
output of microphone (assuming you know about electronics: still requires a biasing circuit e.g. a resistor + capacitor).
the output of the microphone is still very weak (in the magnitude of mV), which Arduino is incapable of capturing so you need a pre-amplifier
the design of the pre-amplifier will also include DC offset which makes the output of the microphone all above 0VDC which is in the range of the Arduino ADC otherwise the arduino will capture only those above 0VDC.
I am trying to O/P audio to a USB soundcard (Lindy PnP SoundCard device) via Python/PySide/PyQT by the use of Phonon and/or QTMultimedia.
I can O/P the aduio (mp3/wav) which is no problem - the issue is that I want to control the USB's output sample clock rate, I need to be able to change this from 44.1 to 48 kHz. The soundcard comes with its own s/w that allows this so it is possible.
I can play Audio through Phonon like so..
self.mediaObj=phonon.Phonon.MediaObject(self)
self.audioSink=Phonon.AudioOutput(Phonon.MusicCategory, self)
self.audioPath=Phonon.createPath(self.mediaObj, self.audioSink)
self.audioSink.setVolume(0.3)
However I do not see any way to change the sample clock rate of the USB device from looking at the Class Reference doc's it seems its not possible.
http://www.pyside.org/docs/pyside/PySide/phonon/index.html
So then I have tried to use Qt Multimedia to change the USB soundcards O/P clock rate..
format = QtMultimedia.QAudioFormat()
format.setChannels(2)
format.setFrequency(44100)
format.setSampleSize(16)
format.setByteOrder(QtMultimedia.QAudioFormat.LittleEndian)
format.setSampleType(QtMultimedia.QAudioFormat.SignedInt)
This has no effect. Does anyone know how I would do this and if it is even possible with Phonon/PyQT? I am guessing I need to go lower and try find the USB Soundcard directly which will be messy..
Much appeciate any help!!
Alan