I'm new to USB development, and i'm quite confused about what data rates are realistic.
I'm trying to develop an external sound card connected on an AVR32 processor, which supports USB Full Speed(12 Mb/s). I'll use USB audio class 1 to send the audio data to a PC. I need to send 24 bit, 48kHz, 2 channels as INput to the computer, but also 24 bit, 48kHz, 1 channel OUTput from the computer. Streaming both ways.
That gives me a data rate of: 24 bit * 48kHz * 3 channels = 3,5 Mb/s, which should be possible by using USB Full Speed?
I understand that the Audio Class sends data via an Isochronous transfer, but i'm confused about how many transactions ( e.g. IN = 256 bytes ) it is possible to make in one frame? according to the USB specification (http://www.usb.org/developers/docs/usb20_docs/#usb20spec - > table 5-4) it seems to be possible to send more than one transaction per frame?
Is it possible to send both IN and OUT packets within one frame?
Thanks in advance!
Related
I am implementing an USB audio input device (STM32 with two microphones), following the Universal Serial Bus Device Class Definition for Audio Devices, Release 2.0.
I want to use USB High Speed, not Full Speed. The Standard AS Isochronous Audio Data Endpoint Descriptor has the field bInterval.
How should I choose the value for bInterval? Can I use a value of 0x01 (interval of 125us) or should I use an interval of 1ms (0x04, which would be the same interval as in Full Speed)? Does a shorter interval has advantages for the amount of samples that can be transferred (later I would like to use up to 16 microphones)?
I am using python sockets to connect to a bluetooth HC-05 module with my PC. I want to send music to the HC-05 by converting a wav file to a string array which will later by converted to integers ranging from 0-65535 on an arduino. The arduino and HC-05 communicate via serial at 9600 baud. Then those ints will be passed into a DAC via I2C. I am wondering if there could be a memory issue sending an enormous number of strings from my PC. Is it possible the original quality of the sound will be distorted as a result of the different rates of sending/receiving data across the devices? Or will the sound signal just be delayed on the DAC?
The arduino and HC-05 communicate via serial at 9600 baud.
This is going to be too low to be usable for audio.
9600 baud gives you 7680 bits of data per second. At 16 bits per sample, you're looking at a sample rate of 481 Hz, which is too low for intelligible audio. It's barely even high enough to reproduce sound at all.
You need to:
Increase the baud rate. Ideally you'll want at least 57600 baud, for 46 kbit/sec of data. If higher baud rates are available, use them.
Use fewer bits for each sample. Using 4 bits for each sample at 56 kbaud will give you a respectable sample rate of 11.5 kHz. The audio will sound a little tinny at 4 bits, but it'll be intelligible.
I'm working on a project that streams data to a laptop. The data is an analog signal with 5 kHz bandwidth (almost like audio) that is digitized at first and should be transmitted over a bluetooth module to a laptop. I've searched a lot about modules that use this protocol to stream data. I figured out simple bluetooth modules like hc-05 or hc-06 because of their limitation of packet's size and interval time can't be used for this application. It has been suggested to use audio bluetooth module like bc127 and csr because of their appropriate sample rate (I want more than 20 kS/s) and their applications so I want to use them but not for an audio signal for my signal. Now I want to ask you:
1- Can I use these modules to acquire my signal (that is not an audio signal) wireless?
2- Do these modules compress signal for transmission and should I decompress it in receiver side (I know they have some audio DSP but I don't know what are they and their function)?
3- Can laptop's bluetooth hardware receive this data without any problem? If not, what are alternatives?
4- is there any filtering in the proccess? i mean filter about voise band(300 Hz ~ 4 kHz)
thank you.
I'm implementing a LIN protocol on a Linux SBC that transmits over a UART. I don't have time to develop a complete LIN stack, so I'm just implementing a frame structure for messages as defined by the protocol. The problem is that the protocol requires a "Break" field which makes the slave devices on the bus listen. This field consists of zeros for 13 bit-times. Any ideas how to send zeros 13 bit-times over UART, when serial data transmission requires complete bytes?
Per Wiki:
LIN (Local Interconnect Network) is a serial network protocol used for
communication between components in vehicles. The need for a cheap
serial network arose as the technologies and the facilities
implemented in the car grew, while the CAN bus was too expensive to
implement for every component in the car. European car manufacturers
started using different serial communication topologies, which led to
compatibility problems.
If you would have paid attention at class you would have known that:
Data is transferred across the bus in fixed form messages of
selectable lengths. The master task transmits a header that consists
of a break signal followed by synchronization and identifier fields.
The slaves respond with a data frame that consists of between 2, 4 and
8 data bytes plus 3 bytes of control information.
You should just send an echo of 0x0000 following by CR/LF.
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.