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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-01-10 21:03:29 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2016-01-10 21:03:29 +0100 |
commit | e73462e974667783a6bc8dddd5204b0ab3febfc7 (patch) | |
tree | f9624ab51aa0e37345a75a40f94e9680031e85e9 | |
parent | 1d7a8bc8ea37d90f19ecf0f595eaf85dfa891e93 (diff) | |
download | fdk-aac-dabplus-0.8.0.tar.gz fdk-aac-dabplus-0.8.0.tar.bz2 fdk-aac-dabplus-0.8.0.zip |
Prepare v0.8.0v0.8.0
-rw-r--r-- | ChangeLog | 6 | ||||
-rw-r--r-- | README.md | 45 | ||||
-rw-r--r-- | configure.ac | 2 |
3 files changed, 26 insertions, 27 deletions
@@ -1,3 +1,9 @@ +2016-01-10: Matthias P. Braendli <matthias@mpb.li> + (v0.8.0): + * dabplus-enc: + Fix bug in v0.7.4 where the VLC compressor was + active even if not enabled with the gain option. + Add drift compensation to VLC input. 2016-01-03: Matthias P. Braendli <matthias@mpb.li> (v0.7.4): * dabplus-enc: @@ -9,9 +9,6 @@ The main tool is the *dabplus-enc* 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 ALSA input supports experimental 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. @@ -19,6 +16,9 @@ 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. +The ALSA and libVLC inputs support experimental sound card clock drift +compensation, that can compensate for imprecise sound card clocks. + *dabplus-enc* includes support for DAB MOT Slideshow and DLS, contributed by [CSP](http://rd.csp.it). @@ -92,6 +92,12 @@ The ZeroMQ output included in FDK-AAC-DABplus 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 + + dabplus-enc -b $BITRATE -i wave_file.wav -o station1.dabp + Scenario *ALSA* --------------- Live Stream from ALSA sound card at 32kHz, with ZMQ output for ODR-DabMux: @@ -114,7 +120,12 @@ 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. +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 +*mot-encoder*. + +If the webstream bitrate is slightly wrong (bad clock at the source), you can +enable drift compensation with **-D**. Scenario *JACK input* --------------------- @@ -149,21 +160,10 @@ 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 *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*. - - sox -t alsa $ALSASRC -b 16 -t raw - rate 32k channels 2 | \ - dabplus-enc -r 32000 -l \ - -i - -b $BITRATE -f raw -o $DST -p 53 - -The -p 53 sets the padlen, compatible with the default *mot-encoder* setting. *mot-encoder* needs -to be given the same value for this option. - - 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. @@ -171,14 +171,7 @@ using mplayer. If there are no data in FIFO, encoder generates silence. dabplus-enc -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 oputput in mplayer. Mplayer log messages on stdout. - -Scenario *wav file for offline processing* ------------------------------------------- -Wave file encoding, for non-realtime processing - - dabplus-enc -b $BITRATE -i wave_file.wav -o station1.dabp - +*Note*: Do not use /dev/stdout for pcm output in mplayer. Mplayer log messages on stdout. Return values ------------- @@ -189,7 +182,7 @@ dabplus-enc returns: * 2 if the silence timeout was reached * 3 if the AAC encoder failed * 4 it the ZeroMQ send failed - * 5 if the ALSA input had a fault + * 5 if the input had a fault Usage of MOT Slideshow and DLS ============================== diff --git a/configure.ac b/configure.ac index 83811e5..8aa842f 100644 --- a/configure.ac +++ b/configure.ac @@ -1,7 +1,7 @@ dnl -*- Autoconf -*- dnl Process this file with autoconf to produce a configure script. -AC_INIT([fdk-aac-dabplus], [0.7.4], [http://opendigitalradio.org/]) +AC_INIT([fdk-aac-dabplus], [0.8.0], [http://opendigitalradio.org/]) AC_CONFIG_AUX_DIR(.) AC_CONFIG_MACRO_DIR([m4]) AM_INIT_AUTOMAKE([tar-ustar foreign]) |