I'm running a BBB straight out of the box running debian. Kernel version is 3.8.13-bone-47.
I'm working with a cape that is very similar to the one here. The difference is that I'm using a TLV320AIC3106 instead of the AIC3104, and I only have enabled the audio out, I'm not interested in recording audio in this application.
My pinout for my application is identical to the cape in the link above.
I've followed the link here to get the cape up and running. Everything that I have matches the output of the tutorial up until I try and play a sample wave file.
When I play a sample wave file, I get the following message: aplay: pcm_write:1710: write error: Input/output error
Running dmesg gives me ALSA sound/core/pcm_lib.c:1010 playback write error (DMA or IRQ trouble?)
Where I'm having trouble is I don't understand how the DMA is coming in to play. Is this a DMA problem? Is it a symptom of something else going wrong like my I2C? Am I missing a configuration somewhere else?
Any thoughts on how to track this down are appreciated.
I realize it has been covered in multiple places before, but it can never be stressed enough. Make sure that when you're sending out information, make sure it goes to the right address over the I2C. I figured out this morning that the audio codec was at address 0x1B, while the driver was addressed to 0x18. Small but critical difference.
The easy fix is to edit the BB-BONE-AUDI-02-00A0.dts file.
Edit line 65 to <0x1B>. Re-compile using the line: dtc -O dtb -o BB-BONE-AUDI-02-00A0.dtbo -b 0 -# BB-BONE-AUDI-02-00A0.dts
move the generated file to the /lib/firmware directory
Insert it using echo BB-BONE-AUDI-02 > /sys/devices/bone_capemgr*/slots
After applying this simple fix it seems to work. I can't say for certain because I have to get the audio amplifier circuit up and running still. At least aplay will play the file without crashing out on me, which is a start.
Related
Let me start off with the components used:
GPIO extension board to connect the raspberry pi to components
Bme280
Pi 3B
Problem description:
This problem was intermittent at first. We were receiving reading and the error occured out of nowhere, made adjustment to the sensor(taking out and putting it back in) which solved it but when we touch the sensor the error showed up again.
We decided to change the sensor and the problem turned from intermittent to persistent. Now we have this as permanent issue.
We removed the package for BME280 and SmBus2 and decided to reinstall it again.
RPI Bme280: https://pypi.org/project/RPi.bme280/
I followed this BME280 installation guide, however when prompting for 'i2cdetect -y 1' there were lines but no number showing up.
What has been confirmed so far:
Patching up of circuits was fine, tested with my lecturer pi and it
works fine.
Codes is fine, we even use the sample code provided by RPi.bme280
I2c was enabled everytime
Please help me because I am really lost. I needa submit this by next week so any help is appreciated.
Problem pic --> Picture of the problem
I've tried to respond, but there is a few pitfalls in SMBus and i2c, so to break the question into smaller bits I've quoted your original message with an indent. The response is provided beneath.
This problem was intermittent at first. We were receiving reading and the error occurred out of nowhere, made adjustment to the sensor(taking out and putting it back in) which solved it but when we touch the sensor the error showed up again.
I locked onto the statement, "Touched the sensor". Is there a chance the BME280 experienced an ESD discharge? And therefore is now partially or fully broken?
Are you using an ESD safe room? or at least ESD bracelet. You will not notice small ESD discharges, and for small electronics even a sub 1000V discharge can cause damage instantly or over time. (Fundamentals of Electrostatic Discarge - incompliancemag.com)
I followed this BME280 installation guide, however when prompting for 'i2cdetect -y 1' there were lines but no number showing up.
The I2Cdetect -y 1
Will send a ReadBit message to each possible slave 0-127. Then listens for the ACK on the slave address.
The message is very simple, but does the trick for detecting working i2c slaves.
As you stated there is no "Lines", i would suggest the slave either has issues responding, if the BME280 has the correct 3v3+GND supplied.
As another possability, could you redo this step dtparam=i2c_arm=on,i2c_baudrate=400000
(Set the baudrate way down to 10000 /10k. Should this solve the issue and you need higher bandwidth, attempt on "50000 /50k" or "100000 /100k") This is just to eliminate the possibility the slave cannot opperate on this high frequency. 400k Baudrate for SMBus is high for a prototype/testing scenario. Very small amounts of interference can corrupt the data over SDA.
Another possibility is the Sink-Current which i've learned the hard way is not the same on i2c and SMBus over i2c.
Is you Pull-up resistor appropriate for the limited sink-current?
The SMBus specifies a minimum sink current of 100µA, and a maximum of 350µA, compared to 3mA for the I²C bus (Ampere=Voltage/Resistance Ohm).
Now to the Software issue.
From your supplied picture on imgur.
File "newcodes_''''''''''''''.py" line 2, in
Your compiler cannot find "bme" where you attempt to import bme280. Reference/path issue?
This can be the cause for other issues further into the code calls.
I don't know what SW you are writing the code in, but if there is an option to insert a breakpoint and run a debug build. I would suggest starting there :)
Alright I have resolved the issue, I changed the wire from jumper wire to hard wire. Thank you guys for your response. I appreciate all of them :D
I'm having a problem with LIRC breaking audio on the OS scale after firing a command.
For example, I'd do:
irsend send_once Samsung_BN59-01224C KEY_VOLUMEUP --count=5
and afterwards, play an audio file, and the program governing that file would seize up and not play any sound. Same goes for a script I've written that uses the pygame library for python.
What's worse is that LIRC also stops firing correctly after this bug occurs. I can see infrared light being shot out of the diode, but there might be something off with the timing.
This happens both ways, so, after playing an audio file, LIRC will stop working but further playing of audio is possible.
The following extremely rarely but sometimes I'm able to play audio after LIRC finishes a command, and the result is heavily pitched down version of the original sound that cuts out after around a second or so.
Tested with different remotes, same results occur. I'm not sure if the fix that a user proposed in this thread could cause this (https://github.com/raspberrypi/linux/issues/2993) but I'm putting it out there that I used it, since unmodified LIRC has problems with both the receiver and transmitter turned on in /boot/config.txt. The rest of my installation is standard.
Fixed this by reverting the fix I posted in the last paragraph. Apparently using PWM for Infrared causes issues with onboard audio on raspbian. I commented out the lines responsible for the receiver and left the gpio-ir-tx option uncommented. Works fine with just the transmitter on.
I am a beginner to arduino and trying to follow one of the videos from youtube to make a project- Master And Slave using HC-05 modules. I followed every step from that video carefully. Here's the link to the video ->
https://www.youtube.com/watch?v=BXXAcFOTnBo
For Slave Module when I type AT commands in the serial monitor nothing shows up.
I searched in Internet for the this problem and still didn't fix it. Here's What i have done:
tired restarting my pc
tried Changing arduino, At first i used arduino nano and then uno
I tried changing the baud rate to all other values and nothing happened
And there is no change in light blink, it stays on AT command mode until power is turned off and on again.
This is how is looks
Thank You For the Answer
Your problem is not actually related to arduino in any way. This tutorial is utilizing onboard USB->UART transceiver IC. So in theory you could even remove Atmel chip from the board. Its more like using module like this FT232 converter. Anyways sorry if I confused you but you can research the subject.
But anyways to the problem. Couple of questions:
Did you press reset button on module while powerin up the circuit?
Thats the way it enters to AT command mode
Did you try to reverse RX-TX lines, don't worry connecting these guys across eatch other doesn't harm your board.
(They are so often plugged in wrong..)
Did you make sure you have connected EN pin of module also. (To arduino 3.3V not 5V)
Did you triple check all connections
Try to upload simple Sketch like blinky to Arduino to make sure you are connected to USB-UART transeiver and you have correct drivers.
Make sure you upload empty sketch or remove the chip when you try to apply tutorial steps.
Change jumper wires. Sometimes cheap jumper wires are really bad quality.
I have been sometimes scratching head for long time because of broken jumper cable.
Please let me know if you have checked all of those so we can think for next step.
I've been playing around with some speech-to-text and text-to-speech systems, and am running into the problem that when the computer makes sounds that it can recognize, it starts taking commands from itself. To avoid this, I'd like a stream of all sounds picked up by the microphone that were not produced by the computer itself.
I see that PulseAudio has an echo cancellation module, but so far I have been unable to distinguish between its output and the raw microphone output: it still contains all the sounds picked up by the microphone that came from the computer speakers. I wonder if the default echo canceller is doing the opposite of what I want (i.e., it removes sounds heard by the microphone from being sent to the speakers).
Any idea how I can do this (preferably with pacmd)? I have thoroughly confused myself trying to specify non-default sources for the echo canceller, and have wandered into loopback modules and other things that are probably irrelevant. I know very little about PulseAudio, haven't found a good introduction to it (I've looked through much of the PulseAudio documentation but didn't see anything relevant), and might just be missing something simple. I feel frustrated that echo cancellation apparently doesn't work, I can't find documentation on it, and I can't find examples of it working from other people.
Thanks in advance for the help!
Other details that might be relevant: I'm running Ubuntu Saucy on a Lenovo Thinkpad T410. I'm using the built-in microphone and speakers (so, I'm pretty sure they're using the same sound card and I won't have clock drift issues). My actual application gets its sound through GStreamer, but GStreamer gets it from PulseAudio, and I don't think GStreamer itself has AEC capabilities. If there's a different way of doing this, I'd gladly switch to that.
Ah, I've got it! Merely loading the echo cancellation plugin isn't enough; you then need to start using it. In particular, it will only cancel echos of sounds passed into it, and if no sounds go through it, nothing will be cancelled. So, open /etc/pulse/default.pa and add the line
load-module module-echo-cancel
towards the bottom (I put it right after the line that loads module-filter-apply). Then, restart the PulseAudio daemon by running (as a non-root user) pulseaudio -k. Next, run pacmd to get a command line interface to PulseAudio, and give it the commands list-sources and list-sinks. Note the indices of the echo canceller in the responses. Edit /etc/pulse/default.pa again, and uncomment the two lines at the end about set-default, replacing the words input and output with the indices of the echo canceller's source and sink. Finally, restart PulseAudio again with pulseaudio -k (again, run as a non-root user).
Now, by default all sounds to be output get sent through the echo canceller before heading to the speakers, and all sounds to be input get pulled from the echo canceller after coming in through the microphone, and things actually work. You can verify that it's working by running pavucontrol and looking at the sound levels on the Input Devices screen (try playing some music and speaking, and note that the echo cancelled input shows normal sound levels when you speak but very low levels (verging on nothing) when you're silent but the music is playing).
This answer mostly comes from this post, which I wish I'd found weeks ago.
I'm working on an audio recognition project.
For testing, I'd like to be able to have a program:
load audio data from a file
provide it to the Linux kernel, as if it were coming from a microphone
have any user-space program sampling the microphone be obtaining data sourced
from my file.
Is that possible in Linux without having to write a new kernel module?
EDIT: i guess that solution won't work .. but see my comment below.
this shoud be simple under linux.
here are the steps:
make a named pipe with mkfifo (mkfifo ~/audio_out.pipe)
cat the audiofile into this pipe (cat test.wav > ~/audio_out.pipe)
get the program you want to listen, to get input from this pipe. maybe you have to make a symlink for programs not flexible enough to read from any device.
I hope I got your question right.