ODR-AudioEnc Package
This package contains a DAB and DAB+ encoder that integrates into the ODR-mmbTools.
The DAB encoder is based on toolame. The DAB+ encoder uses a modified library of the Fraunhofer FDK AAC code from Android, patched for 960-transform to do DAB+ broadcast encoding. Both encoders are part of this repository.
The main tool is the odr-audioenc encoder, which can read audio from a file (raw or wav), from an ALSA source, from JACK or using libVLC or GStreamer, and encode to a file, a pipe, to an EDI or ZeroMQ output compatible with ODR-DabMux.
The libVLC input allows the encoder to use all inputs supported by VLC, and therefore also webstreams and other network sources.
The GStreamer input is an alternative to read from various sources.
The ALSA, libVLC and GStreamer inputs support sound card clock drift compensation, that can compensate for imprecise sound card clocks.
The JACK input does not automatically connect to anything. The encoder runs at the rate defined by the system clock, and therefore sound card clock drift compensation is also used.
odr-audioenc can insert Programme-Associated Data, that can be generated with ODR-PadEnc. ODR-AudioEnc v3 is compatible with ODR-PadEnc v3.
For detailed usage, see the usage screen of the tool with the -h option.
More information is available on the Opendigitalradio wiki
Installation
You have 3 ways to install odr-audioenc on your host:
Using your linux distribution packaging system
odr-audioenc is available on the official repositories of several debian-based distributions, such as Debian
(from Debian 12), Ubuntu (from 24.10), Opensuse and Arch.
If you are using Debian 12 (Bookworm), you will need to add the backports repository
Using installation scripts
If your linux distribution is debian-based, you can install odr-audioenc as well as the other main components of the mmbTools set with the Opendigitalradio dab-scripts
Compiling manually
Unlike the 2 previous options, this one allows you to compile odr-audioenc with the features you really need.
Requirements
For Debian Bullseye-based OS, run the following commands:
# Required packages
## C++11 compiler
sudo apt-get install --yes build-essential automake libtool
## ZeroMQ
sudo apt-get install --yes libzmq3-dev libzmq5
# optional packages
## alsa libraries
sudo apt-get install --yes libasound2-dev
## JACK audio connection kit
sudo apt-get install --yes libjack-jackd2-dev
## libvlc and vlc for the plugins
sudo apt-get install --yes libvlc-dev
## gstreamer-1.0
sudo apt-get install --yes libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev
## cURL to download the TAI-UTC bulletin, needed for timestamps in EDI output
sudo apt-get install --yes libcurl4-openssl-dev
Attention: on versions older than Debian Buster, you'll need vlc-nox instead of vlc-plugin-base
Compilation
- Clone this repository: ``` # stable version: git clone https://github.com/Opendigitalradio/ODR-AudioEnc.git
# or development version (at your own risk):
git clone https://github.com/Opendigitalradio/ODR-AudioEnc.git -b next
1. Configure the project
cd ODR-AudioEnc
./bootstrap
# Select the features you need:
./configure --enable-alsa --enable-jack --enable-vlc --enable-gst
1. Compile and install:
make
sudo make install
```
How to use
We assume that you have a ODR-DabMux configured for an EDI input on port 9000.
ALSASRC="default"
DST="tcp://yourserver:9000"
BITRATE=64
General remarks
Avoid using sources that are already encoded with a low bitrate, because encoder cascading will noticeably reduce audio quality. Best are sources encoded with a lossless codec (FLAC). Otherwise, try to get MP3 at 320kbps, AAC at 256kbps or higher bitrates.
Ideally use a source at the correct sampling rate (32kHz or 48kHz, according to
your encoder configuration). VLC can do resampling, but on some systems selects
the ugly resampler which creates artifacts. Try adding
-L --audio-resampler=samplerate -L --src-converter-type=0
to your command line, but enable verbose mode and read the VLC debug output to
check that it enables the libsamplerate resampler, and not the ugly resampler.
The codecs do not behave well when your source material has peaks that go close
to saturation, especially when you have to resample. When you see little
exclamation marks with the -l option, it's too loud! Reduce the gain at the
source, or use the gain option if that's not possible.
DAB+ AAC encoder configuration
By default, when not overridden by the --aaclc, --sbr or --ps options,
the encoder is configured according to bitrate and number of channels.
If only one channel is used, SBR (Spectral-Band Replication, also called HE-AAC) is enabled up to 64kbps. AAC-LC is used for higher bitrates.
If two channels are used, PS (Parametric Stereo, also called HE-AAC v2) is enabled up to 48kbps. Between 56kbps and 80kbps, SBR is enabled. 88kbps and higher are using AAC-LC.
EDI output
The EDI output included in ODR-AudioEnc is able to connect to
one or several instances of ODR-DabMux. The -e option can be used
more than once to achieve this. The same goes for the ZeroMQ output (-o option).
Scenario wav file for offline processing
Wave file encoding, for non-realtime processing
odr-audioenc -b $BITRATE -i wave_file.wav -o station1.dabp
Scenario file that VLC supports
If you want to input a file through libvlc, you need to give an absolute path:
odr-audioenc -b $BITRATE -v file:///home/odr/audio/source.mp3 -o station1.dabp
Scenario ALSA
Live Stream from ALSA sound card at 32kHz, with EDI output for ODR-DabMux:
odr-audioenc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -e $DST -l
To enable sound card drift compensation, add the option -D:
odr-audioenc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -e $DST -D -l
You might see U and O appearing on the terminal. They correspond to audio underruns and overruns that happen due to the different speeds at which the audio is captured from the soundcard, and encoded into HE-AACv2.
High occurrence of these will lead to audible artifacts.
Scenario encode a webstream
You can use either GStreamer with the -G option or libVLC with -v.
Read a webstream and send it to ODR-DabMux over EDI:
odr-audioenc -G $URL -r 32000 -c 2 -e $DST -l -b $BITRATE
If you need to extract the ICY-Text information, e.g. for DLS, you can use the
-w <filename> option to write the ICY-Text into a file that can be read by
ODR-PadEnc. This does apparently not work for all ogg source streams when using
libVLC.
If the webstream bitrate is slightly wrong (bad clock at the source), you can
enable drift compensation with -D.
Scenario Custom GStreamer pipeline
The --gst-pipeline option lets you run custom pipelines, using the same
syntax as gst-launch, which can be necessary for sources that you cannot specify through a URI through the -G option.
For example, you may use udpsrc to receive an RTP stream:
odr-audioenc --gst-pipeline 'udpsrc port=5004 caps=application/x-rtp,media=(string)audio,payload=(int)10,clock-rate=44100 ! rtpL16depay ! audioconvert ! audioresample' \
-e $DST -l -b $BITRATE
Scenario JACK input
JACK input: Instead of -i (file input) or -d (ALSA input), use -j *name*, where name specifies the JACK
name for the encoder:
odr-audioenc -j myenc -l -b $BITRATE -e $DST
The JACK server must run at the samplerate of the encoder (32kHz or 48kHz). If that is not possible, one workaround is to access JACK through VLC, which will resample accordingly:
odr-audioenc -l -v jack://dab -b $BITRATE -e $DST
Scenario LiveWire or AES67
When audio data is available on the network as a multicast stream, it can be encoded using the following pipeline:
rtpdump -F payload 239.192.1.1/5004 | \
sox -t raw -e signed-integer -r 48000 -c 2 -b 24 -B /dev/stdin -t raw --no-dither -r 48000 -c 2 -b 16 -L /dev/stdout gain 4 | \
odr-audioenc -f raw -b $BITRATE -i /dev/stdin -e $DST
It is also possible to use the libvlc input, where you need to create an SDP file with the following contents:
v=0
o=Node 1 1 IN IP4 172.16.235.155
s=TestSine
t=0 0
a=type:multicast
c=IN IP4 239.192.0.1
m=audio 5004 RTP/AVP 97
a=rtpmap:97 L24/48000/2
Replace the IP address in the o= field by the one corresponding to your
source node IP address, and the IP in c= by the multicast IP of your stream.
Then use this SDP file as input for the VLC input.
This could maybe also work with GStreamer, but needs more testing. Help would be appreciated in improving the GStreamer input code to also support more advanced features, some pointers are in TODO.md
Scenario local file through snd-aloop
Play some local audio source from a file, with EDI or ZMQ output for ODR-DabMux. The problem with playing a file is that odr-audioenc cannot directly be used, because ODR-DabMux does not back-pressure the encoder, which will therefore encode much faster than realtime.
While this issue is sorted out, the following trick is a very flexible solution: use the alsa virtual loop soundcard snd-aloop in the following way:
modprobe snd-aloop
This creates a new audio card (usually 'hw:1' but have a look at /proc/asound/card to be sure) that
can then be used for the alsa encoder.
./odr-audioenc -d hw:1 -c 2 -r 32000 -b 64 -e $DST -l
Then, you can use any media player that has an alsa output to play whatever source it supports:
cd your/preferred/music
mplayer -ao alsa:device=hw=1.1 -srate 32000 -format=s16le -shuffle *
Important: you must specify the correct sample rate and sample format on both "sides" of the virtual sound card.
Scenario mplayer and fifo
Warning: Connection through pipes to ODR-DabMux are deprecated in favour of EDI.
Live Stream resampling (to 32KHz) and encoding from FIFO and preparing for DAB muxer, with FIFO to odr-dabmux using mplayer. If there are no data in FIFO, encoder generates silence.
mplayer -quiet -loop 0 -af resample=32000:nowaveheader,format=s16le,channels=2 -ao pcm:file=/tmp/aac.fifo:fast <FILE/URL> &
odr-audioenc -l -f raw --fifo-silence -i /tmp/aac.fifo -r 32000 -c 2 -b 72 -o /dev/stdout \
mbuffer -q -m 10k -P 100 -s 1080 > station1.fifo
Note: Do not use /dev/stdout for PCM output in mplayer. Mplayer logs messages to stdout.
Return values
odr-audioenc returns:
- 0 if it encoded the whole input file
- 1 if some options were not understood, or encoder initialisation failed
- 2 if the silence timeout was reached
- 3 if the AAC encoder failed
- 4 it sending data over the network failed
- 5 if the input had a fault
The -R option to get ODR-AudioEnc to restart the input
automatically has been deprecated. As this feature does not guarantee that
the odr-audioenc process will never die, running it under a process supervisor
is recommended regardless of this feature being enabled or not. It will be removed
in a future version.
Known Limitations
Some receivers did not decode audio anymore between v0.3.0 and v0.5.0, because of a change implemented to get PAD to work. The change was subsequently reverted in v0.5.1 because it was deemed essential that audio decoding works on all receivers. v0.7.0 fixes most issues, and PAD now works much more reliably.
Version 0.4.0 of the encoder changed the ZeroMQ framing. It will only work with ODR-DabMux v0.7.0 and later.
LICENCE
The ODR-AudioEnc project contains
- The code for odr-audioenc in src/ licensed under the Apache Licence v2.0. See http://www.apache.org/licenses/LICENSE-2.0
- libtoolame-dab, derived from TooLAME, licensed under LGPL v2.1 or later. See
libtoolame-dab/LGPL.txt. This is built into a shared library. - EDI output (files in src/edi) are GPLv3+
- The FDK-AAC encoder, patched for DAB+ support, licensed under the terms in
fdk-aac/NOTICE, built into a shared library.
The odr-audioenc binary is statically linked against the libtoolame-dab and fdk-aac libraries.
Whether it is legal or not to distribute compiled binaries from these sources is unclear to me. Please seek legal advice to answer this question.