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Dsd P25 Decoder Software Download: A Guide to Digital Speech Decoding with SDR


Dsd P25 Decoder Software Download: How to Decode Digital Voice Signals with an SDR




If you are interested in listening to digital voice communications such as police, fire, emergency, or ham radio, you may have heard of Dsd, a software package that can decode several digital speech formats with a software defined radio (SDR) device. In this article, we will explain what Dsd is, what it can do, what you need to use it, how to set it up and run it on Windows and Linux, how to decode different digital speech formats with it, and what are some tips and tricks for using it. We will also discuss some limitations and challenges of using Dsd and provide some useful links and resources for further information.




Dsd P25 Decoder Software Download



What is Dsd and what can it do?




Dsd is an open source software package that decodes several digital speech formats




Digital Speech Decoder (DSD) is an open source software package that decodes several digital speech formats using a software defined radio (SDR) device. It was originally developed by Darrin Keene (KB3EWY) and has been maintained and improved by various contributors over the years. Dsd is written in C and can run on Windows, Linux, and Mac OS X platforms.


Dsd supports formats such as P25, DMR, NXDN, D-STAR, and more




Dsd can decode many popular digital speech formats that are used by public safety, amateur radio, commercial, and military communications. Some of the supported formats are:



  • P25: Project 25 is a standard for digital radio communications for public safety agencies in North America. It uses the IMBE and AMBE vocoders for voice encoding and decoding. P25 has two phases: Phase 1 uses FDMA modulation and operates on 12.5 kHz channels, while Phase 2 uses TDMA modulation and operates on 6.25 kHz channels.



  • DMR: Digital Mobile Radio is a standard for digital radio communications for commercial and amateur radio users. It uses the AMBE+2 vocoder for voice encoding and decoding. DMR has three tiers: Tier 1 is for license-free use on 446 MHz band, Tier 2 is for conventional use on licensed frequencies, and Tier 3 is for trunked use on licensed frequencies.



  • NXDN: Next Generation Digital Narrowband is a standard for digital radio communications for commercial and public safety users. It uses the AMBE+2 vocoder for voice encoding and decoding. NXDN has two modes: NXDN48 uses 6.25 kHz channels with FDMA modulation, while NXDN96 uses 12.5 kHz channels with FDMA or TDMA modulation.



  • D-STAR: Digital Smart Technologies for Amateur Radio is a standard for digital radio communications for amateur radio users. It uses the AMBE vocoder for voice encoding and decoding. D-STAR has three modes: DV mode uses GMSK modulation and operates on 6.25 kHz channels, DD mode uses QPSK modulation and operates on 128 kbps data streams, and DV Fast Data mode uses GMSK modulation and operates on 96 kbps data streams.



  • Yaesu System Fusion: Yaesu System Fusion is a proprietary standard for digital radio communications for amateur radio users. It uses the AMBE+2 vocoder for voice encoding and decoding. Yaesu System Fusion has two modes: C4FM mode uses C4FM modulation and operates on 12.5 kHz channels, while VW mode uses V/D mode and operates on 12.5 kHz channels.



  • ProVoice: ProVoice is a proprietary standard for digital radio communications for public safety users. It uses the IMBE vocoder for voice encoding and decoding. ProVoice operates on 12.5 kHz channels with CQPSK modulation.



  • dPMR: Digital Private Mobile Radio is a standard for digital radio communications for commercial users. It uses the AMBE+2 vocoder for voice encoding and decoding. dPMR operates on 6.25 kHz channels with FDMA modulation.



Dsd uses the mbelib library to synthesize the decoded digital speech




Dsd relies on the mbelib library to synthesize the decoded digital speech from the various formats. mbelib is an open source library that implements the IMBE, AMBE, and AMBE+2 vocoders that are used by most of the digital speech formats. mbelib can convert the encoded bits into PCM samples that can be played back or recorded by Dsd or other software.


What do you need to use Dsd?




You need a software defined radio (SDR) device such as the RTL-SDR




To use Dsd, you need a software defined radio (SDR) device that can receive the digital voice signals that you want to decode. An SDR device is a hardware device that can tune to different frequencies and demodulate different types of signals using software. There are many types of SDR devices available in the market, but one of the most popular and affordable ones is the RTL-SDR dongle. The RTL-SDR dongle is a USB device that can receive signals from about 24 MHz to 1766 MHz with a bandwidth of up to 3.2 MHz.


You need a discriminator tap or a virtual audio cable to connect the SDR output to the Dsd input




To use Dsd, you also need a way to connect the audio output from the SDR device to the audio input of Dsd. There are two ways to do this: using a discriminator tap or using a virtual audio cable.


A discriminator tap is a physical modification of the SDR device or the radio receiver that allows you to access the raw, unfiltered audio signal before it is processed by the device. This signal contains the full information of the digital voice signal and can be fed into Dsd for decoding. However, a discriminator tap requires some soldering skills and may void the warranty of the device.


A virtual audio cable is a software solution that creates a virtual audio device that can transfer the audio output from one software to the audio input of another software. This way, you can use your SDR software (such as SDR#) to demodulate the digital voice signal and send it to Dsd for decoding. A virtual audio cable does not require any hardware modification and is easier to set up and use. However, a virtual audio cable may introduce some latency or distortion in the audio signal and may not work with some SDR software or devices.


You need to download and install Dsd and mbelib from their official sources or third-party websites




To use Dsd, you also need to download and install Dsd and mbelib on your computer. You can get them from their official sources or from third-party websites that provide pre-compiled binaries or packages for different platforms.


The official source for Dsd is its GitHub repository: https://github.com/szechyjs/dsd. Here you can find the latest source code, documentation, and issues of Dsd. You can also download the latest release of Dsd for Windows from here: https://github.com/szechyjs/dsd/releases. However, this release does not include mbelib, so you need to download it separately from here: https://github.com/szechyjs/mbelib/releases. You need to extract both Dsd and mbelib to the same folder and run Dsd from the command line.


If you prefer a graphical user interface (GUI) for Dsd, you can use DSDPlusUI, which is a third-party application that provides a GUI for Dsd and some additional features. You can download it from here: https://dsdplusui.com/. You need to extract DSDPlusUI to the same folder as Dsd and mbelib and run DSDPlusUI.exe.


If you are using Linux, you can compile Dsd and mbelib from their source code or use a package manager to install them. For example, if you are using Ubuntu, you can use these commands to install them:



sudo apt-get install git build-essential libsndfile1-dev git clone https://github.com/szechyjs/mbelib.git cd mbelib make sudo make install cd .. git clone https://github.com/szechyjs/dsd.git cd dsd make sudo make install


Alternatively, you can use a third-party website such as https://www.rtl-sdr.com/tag/digital-speech-decoder/ to download pre-compiled binaries or packages of Dsd and mbelib for different platforms.


How to set up and run Dsd on Windows?




You need to configure your SDR software (such as SDR#) to output the audio to a virtual audio cable or a discriminator tap




To set up and run Dsd on Windows, you need to configure your SDR software (such as SDR#) to output the audio to a virtual audio cable or a discriminator tap. If you are using a virtual audio cable, you need to install it first. You can use any virtual audio cable software that works with your SDR software, such as VB-CABLE, Virtual Audio Cable, or Voicemeeter. You need to create a virtual audio device and set it as the default playback device in your Windows sound settings.


Then, you need to launch your SDR software (such as SDR#) and tune it to the frequency of the digital voice signal that you want to decode. You need to set the modulation mode to NFM (narrowband FM) and adjust the filter bandwidth to match the channel width of the signal (usually 12.5 kHz or 6.25 kHz). You need to set the audio output device of your SDR software to the virtual audio device that you created earlier.


If you are using a discriminator tap, you need to connect it to the audio input of your computer or use an external sound card. You need to set the audio input device of your computer to the discriminator tap or the sound card. You need to adjust the volume and the sample rate of the audio input to match the Dsd requirements.


You need to launch Dsd from the command line or use a graphical user interface (GUI) such as DSDPlusUI




To run Dsd on Windows, you need to launch it from the command line or use a graphical user interface (GUI) such as DSDPlusUI. If you are using the command line, you need to open a command prompt and navigate to the folder where you extracted Dsd and mbelib. You need to type the following command to run Dsd:



dsd -i /dev/dsp -o /dev/dsp -fd


This command tells Dsd to use the default audio input and output devices and to decode P25 signals with FDMA modulation. You can change the options and arguments of Dsd according to the signal type and modulation mode that you want to decode. You can type dsd -h to see the help message and the list of available options and arguments.


If you are using a GUI such as DSDPlusUI, you need to launch it from the folder where you extracted it. You need to select the audio input and output devices and the mode and options for decoding from the GUI. You can also see the decoded information and control some settings from the GUI.


You need to adjust the settings and parameters of Dsd according to the signal type and modulation mode




To decode different digital speech formats with Dsd, you need to adjust the settings and parameters of Dsd according to the signal type and modulation mode. You can do this by using different options and arguments in the command line or by changing them in the GUI. Some of the important settings and parameters are:



  • -fa, -fd, -fr, -fx: These options tell Dsd what modulation mode to use for decoding. -fa is for auto mode, -fd is for FDMA mode, -fr is for TDMA mode, and -fx is for X2-TDMA mode.



  • -mp, -md, -mn, -ms, -my, -mv, -mp: These options tell Dsd what format to use for decoding. -mp is for P25 Phase 1, -md is for DMR/MOTOTRBO, -mn is for NXDN48/96, -ms is for D-STAR, -my is for Yaesu System Fusion, -mv is for ProVoice, and -mp is for dPMR.



  • -e: This option tells Dsd to enable error correction for decoding.



  • -g: This option tells Dsd to enable inverted signal decoding.



  • -u: This option tells Dsd to enable unvoiced speech synthesis.



  • -v: This option tells Dsd to enable verbose output.



  • -w: This option tells Dsd to write the decoded audio to a file.



  • -x: This option tells Dsd to write the decoded information to a file.



  • -q: This argument tells Dsd what level of squelch to use for decoding. It can be a value from 0 (no squelch) to 100 (maximum squelch).



  • -ma, -mb, -mc: These arguments tell Dsd what level of audio gain to use for decoding. They can be values from 0 (no gain) to 64 (maximum gain). -ma is for AMBE gain, -mb is for IMBE gain, and -mc is for C4FM gain.



  • /dev/dsp1, /dev/dsp2, etc.: These arguments tell Dsd what audio device to use for input or output. You can use different numbers or names depending on your system configuration.



You can experiment with different settings and parameters until you find the best combination for decoding your desired signal.


How to set up and run Dsd on Linux?




You need to compile Dsd and m belib from their source code or use a package manager to install them




To set up and run Dsd on Linux, you need to compile Dsd and mbelib from their source code or use a package manager to install them. If you want to compile them from their source code, you need to have the following dependencies installed on your system:



  • git: A version control system that allows you to clone the source code repositories of Dsd and mbelib.



  • build-essential: A package that contains the essential tools for compiling C programs, such as gcc, make, and ld.



  • libsndfile1-dev: A package that contains the development files for libsndfile, a library that allows Dsd to read and write audio files.



You can install these dependencies using your package manager, such as apt, yum, or pacman. For example, if you are using Ubuntu, you can use this command to install them:



sudo apt-get install git build-essential libsndfile1-dev


Then, you need to clone the source code repositories of Dsd and mbelib using git. You can use these commands to do so:



git clone https://github.com/szechyjs/mbelib.git git clone https://github.com/szechyjs/dsd.git


Next, you need to compile and install mbelib using make. You can use these commands to do so:



cd mbelib make sudo make install


Finally, you need to compile and install Dsd using make. You can use these commands to do so:



cd dsd make sudo make install


If you prefer to use a package manager to install Dsd and mbelib, you can check if they are available in your distribution's repositories or in third-party repositories. For example, if you are using Ubuntu, you can use this command to install them from a third-party repository:



sudo add-apt-repository ppa:bladerf/bladerf sudo apt-get update sudo apt-get install dsd mbelib


You need to use a tool such as PulseAudio or ALSA to route the audio from the SDR device to the Dsd input




To run Dsd on Linux, you also need to use a tool such as PulseAudio or ALSA to route the audio from the SDR device to the Dsd input. PulseAudio and ALSA are sound systems that allow you to manage the audio devices and streams on your system. You can use either one of them depending on your preference and configuration.


If you are using PulseAudio, you need to create a null sink and a loopback module that will connect the audio output from the SDR device to the audio input of Dsd. You can use these commands to do so:



pactl load-module module-null-sink sink_name=dsd_sink pactl load-module module-loopback source=dsd_sink.monitor sink=alsa_input.pci-0000_00_1b.0.analog-stereo


These commands will create a null sink named dsd_sink and a loopback module that will route the audio from dsd_sink.monitor (the monitor source of the null sink) to alsa_input.pci-0000_00_1b.0.analog-stereo (the default audio input device). You can change the names of the devices according to your system configuration.


Then, you need to launch your SDR software (such as gqrx) and tune it to the frequency of the digital voice signal that you want to decode. You need to set the modulation mode to NFM (narrowband FM) and adjust the filter bandwidth to match the channel width of the signal (usually 12.5 kHz or 6.25 kHz). You need to set the audio output device of your SDR software to dsd_sink (the null sink that you created earlier).


If you are using ALSA, you need to create an .asoundrc file in your home directory that will define a virtual audio device that will connect the audio output from the SDR device to the audio input of Dsd. You can use this template for your .asoundrc file:



pcm.dsd type plug slave pcm "hw:1,0" route_policy "duplicate" pcm.!default type asym playback.pcm type plug slave.pcm "hw:0,0" capture.pcm type plug slave.pcm "dsd"


This file will create a virtual audio device named dsd that will duplicate the audio output from the SDR device (hw:1,0) to the audio input of Dsd (hw:0,0). You can change the names and numbers of the devices according to your system configuration.


Then, you need to launch your SDR software (such as gqrx) and tune it to the frequency of the digital voice signal that you want to decode. You need to set the modulation mode to NFM (narrowband FM) and adjust the filter bandwidth to match the channel width of the signal (usually 12.5 kHz or 6.25 kHz). You need to set the audio output device of your SDR software to default (the virtual audio device that you defined in your .asoundrc file).


You need to launch Dsd from the terminal and specify the options and arguments for decoding




To run Dsd on Linux, you need to launch it from the terminal and specify the options and arguments for decoding. You can use the same options and arguments as on Windows, but you need to use different names for the audio devices. For example, you can use this command to run Dsd:



dsd -i /dev/dsp -o /dev/dsp -fd


This command tells Dsd to use the default audio input and output devices and to decode P25 signals with FDMA modulation. You can change the options and arguments of Dsd according to the signal type and modulation mode that you want to decode. You can type dsd -h to see the help message and the list of available options and arguments.


How to decode different digital speech formats with Dsd?




You need to tune your SDR software to the frequency of the digital voice signal you want to decode




To decode different digital speech formats with Dsd, you need to tune your SDR software to the frequency of the digital voice signal that you want to decode. You can use any SDR software that works with your SDR device, such as SDR#, gqrx, CubicSDR, or HDSDR. You need to find out the frequency of the digital voice signal that you are interested in by using online resources such as https://www.radioreference.com/, https://www.sigidwiki.com/wiki/Signal_Identification_Guide, or https://www.sdr.hu/, or by scanning the spectrum with your SDR software.


You need to identify the format and modulation type of the signal by looking at its spectrum or using online resources




To decode different digital speech formats with Dsd, you also need to identify the format and modulation type of the signal by looking at its spectrum or using online resources. Different digital speech formats have different characteristics in their spectrum, s


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