I'm trying to use pyaudio to analyze audio playing on my computer's system. To do this I am selecting a recording device called Stereo Mix as my input signal into pyaudio. Stereo Mix seems to be a recording device provided by my computer's soundcard, VIA High Definition Audio.
The problem is the signal I get from pyaudio changes in amplitude when I increase the volume on my computer. I would like to get an absolute signal which does not change with increased volume (ideally the raw signal generated from the computer's system). Does anyone have a solution this problem?
Thanks
Related
I'm trying to record raw composite video siganl to an audio file by connecting the yellow rca cable from a player to the mic input in my pc so I can then put the cable in my audio output and connect it with the video input in an old crt tv and play back the signal to the tv so that I can view the original video.
But that didn't work and I can only see random white lines.
Is that due to frequency limits in the audio format or in the onboard audio chip, or is analog-digital conversion and the other way when recording and playing back damaging the signal?
Video signals operate in ranges above 1 Mhz, where high-quality audio signals only max out at ~96Khz. Video signals would likely need to be be encoded in a format that an audio recorder could pick up, then decoded back into a video signal before a television could render it properly. This answer on the Sound Design exchange may be of interest to you.
A very high bitrate uncompressed audio file may be able to store a low-fidelity video signal, a black and white signal could be stored at sub-vhs quality, but could be at least a resolvable image, recording component video may be possible even though syncing the seperate tracks would be hard.
I tried it.
Sampling rate is 192KHz. It can record up to 192/2=96KHz.
I succeed to capture part of luminance signal.
Color signal is in very high frequency.
So we can't record color signal using soundcard.
Video is very distorted.
However we may can caputure more clearly using soundcard more highter sampling rate.
https://m.youtube.com/watch?v=-Q_YraNAGhw&feature=youtu.be
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 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
I have a capture card that captures SDI video with embedded audio. I have source code for a Linux driver, which I am trying to enhance to add video4linux2 support. My changes are based on the vivi example.
The problem I've come up against is that all the example I can find deal with only video or only audio. Even on the client side, everything seems to assume v4l is just video, like ffmpeg's libavdevice.
Do I need to have my driver create two separate devices, a v4l2 device and an alsa device? It seems like this makes the job of keeping audio and video in sync much more difficult.
I would prefer some way for each buffer passed between the driver and the app (through v4l2's mmap interface) contain a frame, plus some audio that matches up (with respect to time) with that frame.
Or perhaps have each buffer contain a flag indicating if it is a video frame, or a chunk of audio. Then the time stamps on the buffers could be used to sync things up.
But I don't see a way to do this with the V4L2 API spec, nor do I see any examples of v4l2-enabled apps (gstreamer, ffmpeg, transcode, etc) reading both audio and video from a single device.
Generally, the audio capture part of a device shows up as a separate device. It's usually a different physical device (posibly sharing a card), which makes sense. I'm not sure how much help that is, but it's how all of the software I'm familiar with works...
There are some spare or reserved fields in the v4l2 buffers that can be used to pass audio or other data from the driver to the calling application via pointers to mmaped buffers.
I modified the BT8x8 driver to use this approach to pass data from an A/D card synchronized to the video on Ubuntu 6.06.
It worked OK, but the effort of maintaining my modified driver caused me to abandon this approach.
If you are still interested I could dig out the details.
IF you want your driver to play with gstreamer etc. a separate audio device generally is what is expected.
Most of the cheap v4l2 capture card's audio is only an analog pass through with a volume control requiring a jumper to capture the audio via the sound card's line input.