I'm using the component https://github.com/Axios-Engineering/acquisition-components called "RTLTcpSource" from Axios to listen using a RTL2820T USB Dongle. The uses port ComplexIQ_Float plots the port data (and FFT) without any problems.
The data is obviously there but I'm not able to get any sort of audio out. I've tested it against AudioTestSource component, which gives the desired audio output. I have also used DataWriter to output the audio to a file, and then played it through VLC (which plays fine).
I was wondering what sort of adjustments could I make to RTLTcpSource to give it this capability. I have also used the method in this video to try to get an audio output but was not able to play the port data: https://www.youtube.com/watch?v=wN9p8EjiQs4
I'm using RH 1.84 on CentOS 6.4.
Thanks in advance for any tips.
It's not clear from your description what your processing waveform is doing, but you need to have decimation, demod, and data format conversion components in the data path. Something like this:
RTL TCP -> TuneFilterDecimate -> FM Demod -> TuneFilterDecimate -> Reformat.
The Reformat component is needed because the Java audio playback is somewhat picky about it's format and sample rate.
You also may find that a later version of the REDHAWK IDE will be more forgiving of your SDR implementation . Some additional capabilities for playing audio data were added since 1.8.4, and some bug fixes were made as well. In particular, 1.9.X makes certain assumptions about your data stream (e.g. sample rate, number of channels) that must be explicitly specified with keywords using 1.8.4.
Related
How to get file information like sampling rate, bit rate etc of .raw audio files using terminal in linux? Soxi works for .wav files but it isn't working for .raw.
If your life depended on discovering an answer you could make some assumption to tease apart the unknowns ... however there is no automated way since the missing header would give you the easy answers ...
The audio analysis tool called audacity allows you to open up a RAW file, make some guesses and play the track
http://www.audacityteam.org
In audacity goto File -> Import -> Raw Data...
Above settings are typical for audio ripped from a CD ... toy with trying stereo vs mono for starters.
Those picklist widgets give you wiggle room to discover the format of your PCM audio given that the source audio is something when properly rendered is recognizable ... would be harder if the actual audio was noise
However if you need a programmatic method then rolling your own solution to ask those same questions which appear in above window is possible ... is that what you need or will audacity work for you ? We can go down the road of writing code to play off the unknowns mentioned in #Frank Lauterwald's comment
To kick start discovering this information programmatically, if the binary raw audio is 16 bit then each audio sample (point on the audio curve) will consume two bytes of your PCM file. For mono audio then the following two bytes would be your next sample, however if its stereo then these two following bytes would be the sample from the other channel. If more than two channels then just repeat. Typical audio is little endian. Sampling rate is important when rendering the audio, not when programmatically parsing raw bytes. One approach would be to create an output file with a WAV header followed by your source PCM data. Populate the header with answers from your guesswork. This way you could listen to this output file to help confirm your guesses.
Here is a sample 500k mono PCM audio file signed 16 bit which can be imported into audacity or used as input to rolling your own identification code
The_Constructus_Corporation_Long_Street-ycexQvMy03k_excerpt_mono.pcm
I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a 'line in' jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.
The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.
So I can do this today:
Take the good sound and the camera captured sound and load each into
audacity and pretty quickly use the timeshift toot to perfectly fit
the good audio to the questionable audio from the HDMI capture and
cut the good audio to the exact size of the video. Then I can use
ffmpeg or other video editing software to replace the questionable
audio with the better audio.
But while somewhat quick and easy, it always carries with it a bit of human error and time. I'd like to automate this if possible as this process is repeated at least weekly throughout the year.
Does anyone have a suggestion if any of these ideas have merit or could suggest another approach?
I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn't see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats?
if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now-- I've never tried either of these filters yet-- just trying to optimize against blind alleys by asking for advice up front.
If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I'd take it, if it turns out reasonably reliable
Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible-- but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.
any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI?
Thanks in advance for any pointers or suggestinons!
I have a bunch of video clips from a webcam (duration is 5, 10, 60 seconds), and I'm looking for a way to detect "does this video clip have movement", to decide whether the file should be saved or discarded in a future processing phase.
I've looked into motion and OpenCV, but motion seems to only want to work on the raw video stream, and OpenCV seems to be way too advanced for my use.
My ideal solution would be a linux command-line tool that I can feed video files into, and get a simple "does/doesn't contain movement" answer back, so I can discard the irrelevant files. False positives (in a reasonable quantity) are perfectly acceptable for my use.
Does such a tool exist? Or any simple examples of doing this with other tools?
You can check dvr-scan which is simple cross-platform command line tool based on OpenCV.
To just list motion events in csv format (scan only):
dvr-scan -i some_video.mp4 -so
To extract motion in single video:
dvr-scan -i some_video.mp4 -o some_video_motion_only.avi
For more examples and various other parameters see:
https://dvr-scan.readthedocs.io/en/latest/guide/examples/
I had the same problem and wrote the solution: https://github.com/jooray/motion-detection
Should be fairly easy to use from command-line.
If you would like to post-process already-captured video then motion can be useful.
VLC allow you to stream or convert your media for use locally, on your private network, or on the Internet. So an already-captured video can be streamed over HTTP, RTSP, etc. and motion can handle it as a network camera.
Furthermore:
How to Stream using VLC Media Player
If OpenCv is to advanced for you, maybe you should consider something easier which is... SimpleCV (wrapper for OpenCV) "This is computer vision made easy". There is even an example of motion detection using SimpleCV - https://github.com/sightmachine/simplecv-examples/blob/master/code/motion-detection.py Unfortunetely i can't test it(because my OpenCv version isn't compatible with SimpleCV), but generally it looks fine (and isn't complicated) - it just substract previous frame from current and calculate mean of the result. If this value is bigger than some threshold (which most likely you will have to adjust) than we can assume that there were some motion between those 2 frames. Note that setting threshold to 0 is really a bad idea, because always there is some difference between 2 consecuitve frames (changes of lighting, noises, etc).
I'm gonna use the TuneFilterDecimate of Redhawk 1.10 to isolate the RDS data stream of WBFM transmissions.
I wonder why it transforms a real stream of data in a complex one when it is not required from the elaboration and if it is possible to exploit it to make a frequency shift of the signal from 57kHz to the baseband.
I followed this youtube video http://www.youtube.com/watch?v=wN9p8EjiQs4 to try to build a Fm waveform receiver to hear the audio stream but I heard only a distorted audio voice. Can you suggest me some settings?
Thanks for your help.
At present, TuneFilterDecimate will only output complex. You may want to use the FastFilter component instead to perform your filtering. For an example of REDHAWK doing a WBFM RDS demod, check out the Sub100 dollar project.
The documentation is here: http://sourceforge.net/projects/redhawksdr/files/redhawk-doc/1.10.0/
The Waveform used is here: https://github.com/RedhawkSDR/RBDS_wf
You'll need to install the components used within the waveform, those are located in the git repositories.
I'm wondering if it's possible to draw an audio channel of a video or audio file as an image using ffmpeg, or if there's another tool that would do it on Win2k8 x64. I'm doing this as part of an encoding process after a user uploads a video or audio file.
I'm using ColdFusion 10 to handle the upload and calling cfexecute to run ffmpeg.
I need the image to look something like this (without the horizontal lines):
You can do this programmatically very easily.
Study the basics of FFmpeg. I suggest you to compile this sample. It explains how to open a video/audio, identify the streams and loop over the packets.
Once you have the data packet (in this case you are interested only in the audio packets). You will decode it (line 87 of this document) and obtain the raw data of an audio. It's the waveform itself (the analogue "bitmap" for an audio).
You could also study this sample. This second example is how to write a video/audio file. You don't want to write any video, but with this sample you can easily understand how the audio raw data packet works, if you see the functions get_audio_frame() and write_audio_frame().
You need to have some knowledge about creating a bitmap. Any platform has an easy way to do that.
So, the answer for you: YES, IT IS POSSIBLE TO DO THIS WITH FFMPEG! But you have to code a little bit in order to get what you want...
UPDATE:
Sorry, there are ALSO built-in features for this:
You could use those filters... or
showspectrum, showwaves, avectorscope
Here are some examples on how to use it: FFmpeg Filters - 12.22 showwaves.