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. FDK-AAC has to be supplied separately, and is available in the 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, and encode to a file, a pipe, or to a 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 ALSA and libVLC 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.
For detailed usage, see the usage screen of the tool with the -h option.
More information is available on the Opendigitalradio wiki
How to build
- A C++11 compiler
- FDK-AAC with the DAB+ patches
- ZeroMQ 4.0.4 or more recent
- JACK audio connection kit (optional)
- The alsa libraries (libasound2, optional)
- libvlc and vlc for the plugins (optional)
./bootstrap ./configure make sudo make install
If you want to use ALSA, JACK and libVLC inputs, please use
./configure --enable-alsa --enable-jack --enable-vlc
- See the possible scenarios below on how to use the tools
How to use
We assume that you have a ODR-DabMux configured for an ZeroMQ input on port 9000.
ALSASRC="default" DST="tcp://yourserver:9000" BITRATE=64
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.
The ZeroMQ output included in ODR-AudioEnc is able to connect to one or several instances of ODR-DabMux. The -o option can be used more than once to achieve this.
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
Live Stream from ALSA sound card at 32kHz, with ZMQ output for ODR-DabMux:
odr-audioenc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -l
To enable sound card drift compensation, add the option -D:
odr-audioenc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $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 libVLC input for a webstream
Read a webstream and send it to ODR-DabMux over ZMQ:
odr-audioenc -v $URL -r 32000 -c 2 -o $DST -l -b $BITRATE
If you need to extract the ICY-Text information, e.g. for DLS, you can use the -w option to write the ICY-Text into a file that can be read by ODR-PadEnc.
If the webstream bitrate is slightly wrong (bad clock at the source), you can enable drift compensation with -D.
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 -f raw -o $DST
The samplerate of the JACK server should be 32kHz or 48kHz.
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 22.214.171.124/5004 | \ 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 -o $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 126.96.36.199 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.
Scenario local file through snd-aloop
Play some local audio source from a file, with 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:
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 -o $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 ZeroMQ.
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 log messages on stdout.
- 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 the ZeroMQ send 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.
The gain option for libVLC enables the VLC audio compressor with default settings. This has more impact than just changing the volume of the audio.
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.
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.
The odr-audioenc binary is linked against the libtoolame-dab and fdk-aac shared 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.