diff options
3 files changed, 14 insertions, 8 deletions
diff --git a/ChangeLog b/ChangeLog
index 450cc62..b16bbce 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,13 @@
+2014-12-09: Matthias P. Braendli <matthias@mpb.li>
+ (v0.6.0):
+ * dabplus-enc:
+ Fix crashes with high bitrates
+ Improve wav input
+ Add option to quit on silence
+ Enable afterburner by default
+ Fix PAD insertion
+ * mot-encoder:
+ Add support for PNG slides
2014-09-26: Matthias P. Braendli <matthias@mpb.li>
* dabplus-enc:
diff --git a/README.md b/README.md
index ceebb3a..09f5d47 100644
--- a/README.md
+++ b/README.md
@@ -1,7 +1,7 @@
fdk-aac-dabplus Package
-This package contains several tools that use the standalone library
+This package contains an DAB+ encoder that uses the standalone library
of the Fraunhofer FDK AAC code from Android, patched for
960-transform to do DAB+ broadcast encoding.
@@ -69,7 +69,6 @@ input on port 9000.
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.
@@ -82,7 +81,6 @@ and higher are using AAC-LC.
Scenario 1
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
@@ -99,7 +97,6 @@ High occurrence of these will lead to audible artifacts.
Scenario 2
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.
@@ -127,7 +124,6 @@ Scenario 3
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
@@ -209,9 +205,9 @@ working with some receivers, 23 and 26 appear to be broken.
Known Limitations
*mot-encoder* encodes slides in a 10 second interval, which is not linked
-to the rate at which the encoder reads the PAD data.
+to the rate at which the encoder reads the PAD data. It also doesn't prioritise
+DLS transmission over Slides.
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
diff --git a/configure.ac b/configure.ac
index 2ec2760..1220cfe 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], [0.5.1], [http://opendigitalradio.org/])
+AC_INIT([fdk-aac-dabplus], [0.6.0], [http://opendigitalradio.org/])
AM_INIT_AUTOMAKE([tar-ustar foreign])