From 114f153e6c1bc71e9f0ffc47abbce2e2c3803eb3 Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Fri, 9 Sep 2016 15:07:50 +0200 Subject: Remove some references to FDK --- Doxyfile | 2 +- README.md | 44 ++++++++++++++++++++------------------------ libtoolame-dab/README.md | 2 +- src/VLCInput.h | 2 +- src/odr-audioencoder.cpp | 11 ++++------- 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 ** 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 & - 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 (remove the NOT) Matthias P. Braendli 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" -- cgit v1.2.3