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| -rw-r--r-- | README.md | 71 |
1 files changed, 36 insertions, 35 deletions
@@ -6,8 +6,8 @@ of the Fraunhofer FDK AAC code from Android, patched for 960-transform to do DAB+ broadcast encoding. The main tool is the *dabplus-enc* encoder, which can read audio from -a file (raw or wav), from an ALSA source or from JACK, and encode -to a file, a pipe, or to a ZeroMQ output compatible with ODR-DabMux. +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 ALSA input supports experimental sound card clock drift compensation, that can compensate for imprecise sound card clocks. @@ -16,6 +16,9 @@ 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. +The libVLC input allows the encoder to use all inputs supported by VLC, and +therefore also webstreams, and other network sources. + *dabplus-enc* includes support for DAB MOT Slideshow and DLS, contributed by [CSP](http://rd.csp.it). @@ -37,8 +40,9 @@ Requirements: * ImageMagick magickwand (for MOT slideshow) * The alsa libraries (libasound2) * Download and install libfec from https://github.com/Opendigitalradio/ka9q-fec -* Download and install ZeroMQ from http://download.zeromq.org/zeromq-4.0.3.tar.gz +* Download and install ZeroMQ from http://download.zeromq.org/zeromq-4.0.4.tar.gz * JACK audio connection kit (optional) +* libvlc and vlc for the plugins (optional) This package: @@ -49,9 +53,9 @@ This package: make sudo make install -If you want to use the JACK input, please use +If you want to use the JACK and libVLC input, please use - ./configure --enable-jack + ./configure --enable-jack --enable-vlc * See the possible scenarios below on how to use the tools * use mot-encoder to encode images into MOT Slideshow @@ -79,8 +83,8 @@ 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. -Scenario 1 ----------- +Scenario *ALSA* +--------------- Live Stream from ALSA sound card at 32kHz, with ZMQ output for ODR-DabMux: dabplus-enc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -l @@ -95,8 +99,25 @@ the audio is captured from the soundcard, and encoded into HE-AACv2. High occurrence of these will lead to audible artifacts. -Scenario 2 ----------- +Scenario *libVLC input for a webstream* +--------------------------------------- +Read a webstream and send it to ODR-DabMux over ZMQ: + + dabplus-enc -v $URL -r 32000 -c 2 -o $DST -l -b $BITRATE + +This scenario does not yet support ICY-text extraction for DLS. + +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: + + dabplus-enc -j myenc -l -b $BITRATE -f raw -o $DST + +The samplerate of the JACK server should be 32kHz or 48kHz. + +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 *dabplus-enc* cannot directly be used, because ODR-DabMux does not back-pressure the encoder, which will therefore encode much faster than realtime. @@ -119,8 +140,8 @@ Then, you can use any media player that has an alsa output to play whatever sour Important: you must specify the correct sample rate on both "sides" of the virtual sound card. -Scenario 3 ----------- +Scenario *sox and pipes* +------------------------ Live Stream encoding and preparing for DAB muxer, with ZMQ output, at 32kHz, using sox. This illustrates the fifo input over standard input of *dabplus-enc*. @@ -132,20 +153,8 @@ The -p 53 sets the padlen, compatible with the default mot-encoder setting. mot- to be given the same value for this option. -Scenario 4 ----------- -Live Stream encoding and preparing for DAB muxer, with FIFO to odr-dabmux, 48kHz, using -arecord. - - arecord -t raw -f S16_LE -c 2 -r 48000 -D plughw:CARD=Loopback,DEV=0,SUBDEV=0 | \ - dabplus-enc -l -b $BITRATE -f raw -c 2 -r 48000 -i /dev/stdin -o - | \ - mbuffer -q -m 10k -P 100 -s 360 > station1.fifo - -Here we are using the ALSA plughw feature. - - -Scenario 5 ----------- +Scenario *mplayer and fifo* +--------------------------- 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. @@ -155,20 +164,12 @@ using mplayer. If there are no data in FIFO, encoder generates silence. *Note*: Do not use /dev/stdout for pcm oputput in mplayer. Mplayer log messages on stdout. -Scenario 6 ----------- +Scenario *wav file for offline processing* +------------------------------------------ Wave file encoding, for non-realtime processing dabplus-enc -b $BITRATE -i wave_file.wav -o station1.dabp -Scenario 7 ----------- -JACK input: Instead of -i (file input) or -d (ALSA input), use -j *name*, where *name* specifies the JACK -name for the encoder: - - dabplus-enc -j myenc -l -b $BITRATE -f raw -o $DST - -The samplerate of the JACK server should be 32kHz or 48kHz. Return values ------------- |