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authorMatthias P. Braendli <matthias.braendli@mpb.li>2016-01-10 21:03:29 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2016-01-10 21:03:29 +0100
commite73462e974667783a6bc8dddd5204b0ab3febfc7 (patch)
treef9624ab51aa0e37345a75a40f94e9680031e85e9
parent1d7a8bc8ea37d90f19ecf0f595eaf85dfa891e93 (diff)
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Prepare v0.8.0v0.8.0
-rw-r--r--ChangeLog6
-rw-r--r--README.md45
-rw-r--r--configure.ac2
3 files changed, 26 insertions, 27 deletions
diff --git a/ChangeLog b/ChangeLog
index db60bf5..6494f2c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -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:
diff --git a/README.md b/README.md
index a290601..a962c2a 100644
--- a/README.md
+++ b/README.md
@@ -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])