Remove some references to FDK

5 files changed, 27 insertions, 34 deletions

diff --git a/Doxyfile b/Doxyfile
index efae3a7..69047e8 100644
--- a/Doxyfile
+++ b/Doxyfile
@@ -1,7 +1,7 @@
 # Doxyfile 1.8.11
 
 DOXYFILE_ENCODING      = UTF-8
-PROJECT_NAME           = "FDK-AAC-DABplus"
+PROJECT_NAME           = "ODR-AudioEncoder"
 PROJECT_NUMBER         = $(PROJECT_NUMBER)
 PROJECT_BRIEF          = "ODR-mmbTools DAB and DAB+ Audio Encoder"
 
diff --git a/README.md b/README.md
index c666ccd..8fa15c0 100644
--- a/README.md
+++ b/README.md
@@ -1,14 +1,14 @@
-FDK-AAC-DABplus Package
-=======================
+ODR-AudioEncoder Package
+========================
 
 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.
+DAB+ broadcast encoding. FDK-AAC has to be supplied separately.
 
-The main tool is the *dabplus-enc* encoder, which can read audio from
+The main tool is the *odr-audioencoder* 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.
 
@@ -22,7 +22,7 @@ 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.
 
-*dabplus-enc* includes support for DAB MOT Slideshow and DLS, contributed by
+*odr-audioencoder* includes support for DAB MOT Slideshow and DLS, contributed by
 [CSP](http://rd.csp.it).
 
 To encode DLS and Slideshow data, the *mot-encoder* tool reads images
@@ -40,6 +40,7 @@ How to build
 Requirements:
 
 * A C++11 compiler
+* FDK-AAC with the DAB+ patches
 * ImageMagick magickwand (optional, for MOT slideshow)
 * Download and install libfec from https://github.com/Opendigitalradio/ka9q-fec
 * Install ZeroMQ 4.0.4 or more recent
@@ -51,8 +52,6 @@ Requirements:
 
 This package:
 
-    git clone https://github.com/Opendigitalradio/fdk-aac-dabplus.git
-    cd fdk-aac-dabplus
     ./bootstrap
     ./configure
     make
@@ -91,7 +90,7 @@ and higher are using AAC-LC.
 ZeroMQ output
 -------------
 
-The ZeroMQ output included in FDK-AAC-DABplus is able to connect to
+The ZeroMQ output included in ODR-AudioEncoder is able to connect to
 one or several instances of ODR-DabMux. The -o option can be used
 more than once to achieve this.
 
@@ -99,17 +98,17 @@ Scenario *wav file for offline processing*
 ------------------------------------------
 Wave file encoding, for non-realtime processing
 
-    dabplus-enc -b $BITRATE -i wave_file.wav -o station1.dabp
+    odr-audioencoder -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:
 
-    dabplus-enc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -l
+    odr-audioencoder -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -l
 
 To enable sound card drift compensation, add the option **-D**:
 
-    dabplus-enc -d $ALSASRC -c 2 -r 32000 -b $BITRATE -o $DST -D -l
+    odr-audioencoder -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
@@ -121,7 +120,7 @@ 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
+    odr-audioencoder -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 <filename>** option to write the ICY-Text into a file that can be read by
@@ -135,14 +134,14 @@ 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
+    odr-audioencoder -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
+playing a file is that *odr-audioencoder* 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
@@ -153,7 +152,7 @@ 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.
 
-    ./dabplus-enc -d hw:1 -c 2 -r 32000 -b 64 -o $DST -l
+    ./odr-audioencoder -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:
 
@@ -171,14 +170,14 @@ Live Stream resampling (to 32KHz) and encoding from FIFO and preparing for DAB m
 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> &
-    dabplus-enc -l -f raw --fifo-silence -i /tmp/aac.fifo -r 32000 -c 2 -b 72 -o /dev/stdout \
+    odr-audioencoder -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.
 
 Return values
 -------------
-dabplus-enc returns:
+odr-audioencoder returns:
 
  * 0 if it encoded the whole input file
  * 1 if some options were not understood, or encoder initialisation failed
@@ -210,7 +209,7 @@ the generated files are smaller than 50kB and not larger than 320x240 pixels.
 
 Supported Encoders
 ------------------
-*dabplus-enc* can insert the PAD data from *mot-encoder* into the bitstream.
+*odr-audioencoder* can insert the PAD data from *mot-encoder* into the bitstream.
 The mp2 encoder [Toolame-DAB](https://github.com/Opendigitalradio/toolame-dab)
 can also read *mot-encoder* data.
 
@@ -254,17 +253,14 @@ ODR-DabMux v0.7.0 and later.
 LICENCE
 =======
 
-It's complicated. The FDK-AAC-DABplus project contains
+It's complicated. The ODR-AudioEncoder project contains
 
- - The Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for
-   Android, which is under its own licence. See NOTICE. This is built into a
-   shared library.
- - The code for dabplus-enc in src/ licensed under the Apache Licence v2.0. See
+ - The code for odr-audioencoder 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 dabplus-enc binary is linked agains the libtoolame-dab and fdk-aac-dabplus
+The odr-audioencoder binary is linked agains the libtoolame-dab and fdk-aac
 shared libraries.
 
 In addition to the audio encoder, there is also mot-encoder, containing code
diff --git a/libtoolame-dab/README.md b/libtoolame-dab/README.md
index f2e7d1a..0fa8cc6 100644
--- a/libtoolame-dab/README.md
+++ b/libtoolame-dab/README.md
@@ -78,7 +78,7 @@ Mike Cheng <mikecheng at NOT planckenergy.com> (remove the NOT)
 
 Matthias P. Braendli <matthias@mpb.li>
     PAD insertion for DLS and slides
-    Integration into FDK-AAC-DABplus
+    Integration into ODR-AudioEncoder
 
 REFERENCE PAPERS
 ================
diff --git a/src/VLCInput.h b/src/VLCInput.h
index da4b7ae..1d01d66 100644
--- a/src/VLCInput.h
+++ b/src/VLCInput.h
@@ -202,7 +202,7 @@ class VLCInput
 
         /* The thread running process takes samples from m_queue and writes
          * them into m_samplequeue. This decouples m_queue from m_samplequeue
-         * which is directly used by dabplus-enc.cpp
+         * which is directly used by odr-audioencoder.cpp
          */
         void process();
 
diff --git a/src/odr-audioencoder.cpp b/src/odr-audioencoder.cpp
index d98df64..80b7ae6 100644
--- a/src/odr-audioencoder.cpp
+++ b/src/odr-audioencoder.cpp
@@ -18,10 +18,10 @@
  */
 
 /*! \mainpage Introduction
- *  The ODR-mmbTools FDK-AAC-DABplus Audio encoder can encode audio for
+ *  The ODR-mmbTools ODR-AudioEncoder Audio encoder can encode audio for
  *  ODR-DabMux, both DAB and DAB+. 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.
+ *  DAB+ encoder requires a the Fraunhofer FDK AAC library, with the
+ *  necessary patches for 960-transform to do DAB+ broadcast encoding.
  *
  *  This document describes some internals of the encoder, and is intended
  *  to help developers understand and improve the software package.
@@ -101,10 +101,7 @@ using namespace std;
 void usage(const char* name) {
     fprintf(stderr,
     "ODR-AudioEncoder %s is an audio encoder for both DAB and DAB+.\n"
-    "The DAB+ HE-AACv2 encoder is based on a Thirt-Party Modified\n"
-    "Version of the Fraunhofer FDK AAC Codec Library for Android,\n"
-    "and the DAB encoder is using the tooLAME MPEG\n"
-    "encoder sources. The encoder can read from JACK, ALSA or\n"
+    "The encoder can read from JACK, ALSA or\n"
     "a file source and encode to a ZeroMQ output for ODR-DabMux.\n"
     "(Experimental!)It can also use libvlc as an input.\n"
     "\n"