Link against external fdk, rename to odr-audioencoder

1 files changed, 0 insertions, 1307 deletions

diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
deleted file mode 100644
index 35a7496..0000000
--- a/src/dabplus-enc.cpp
+++ /dev/null
@@ -1,1307 +0,0 @@
-/* ------------------------------------------------------------------
- * Copyright (C) 2011 Martin Storsjo
- * Copyright (C) 2016 Matthias P. Braendli
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *    http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
- * express or implied.
- * See the License for the specific language governing permissions
- * and limitations under the License.
- * -------------------------------------------------------------------
- */
-
-/*! \mainpage Introduction
- *  The ODR-mmbTools FDK-AAC-DABplus 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.
- *
- *  This document describes some internals of the encoder, and is intended
- *  to help developers understand and improve the software package.
- *
- *  User documentation is available in the README and in the ODR-mmbTools
- *  Guide, available on the www.opendigitalradio.org website.
- *
- *  The readme for the whole package is \ref md_README
- *
- *  Interesting starting points for the encoder
- *  - \ref dabplus-enc.cpp Main encoder file
- *  - \ref VLCInput.h VLC Input
- *  - \ref AlsaInput.h Alsa Input
- *  - \ref JackInput.h JACK Input
- *  - \ref SampleQueue.h
- *  - \ref charset.h Charset conversion
- *  - \ref toolame.h libtolame API
- *  - \ref AudioLevel
- *  - \ref DataInput
- *  - \ref SilenceDetection
- *
- *  For the mot-encoder:
- *  - \ref mot-encoder.cpp
- *
- *
- *  \file dabplus-enc.cpp
- *  \brief The main file for the audio encoder
- */
-
-#include "config.h"
-#include "AlsaInput.h"
-#include "FileInput.h"
-#include "JackInput.h"
-#include "VLCInput.h"
-#include "SampleQueue.h"
-#include "zmq.hpp"
-#include "common.h"
-
-extern "C" {
-#include "encryption.h"
-#include "utils.h"
-#include "wavreader.h"
-}
-
-#include <vector>
-#include <deque>
-#include <chrono>
-#include <thread>
-#include <string>
-#include <getopt.h>
-#include <cstdio>
-#include <stdint.h>
-#include <time.h>
-#include <unistd.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-
-#include "libAACenc/include/aacenc_lib.h"
-
-extern "C" {
-#include <fec.h>
-#include "libtoolame-dab/toolame.h"
-}
-
-
-
-//! Enumeration of encoders we can use
-enum class encoder_selection_t {
-    fdk_dabplus,
-    toolame_dab
-};
-
-using namespace std;
-
-void usage(const char* name) {
-    fprintf(stderr,
-    "dabplus-enc %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"
-    "a file source and encode to a ZeroMQ output for ODR-DabMux.\n"
-    "(Experimental!)It can also use libvlc as an input.\n"
-    "\n"
-    "The -D option enables experimental sound card clock drift compensation.\n"
-    "A consumer sound card has a clock that is always a bit imprecise, and\n"
-    "would drift off after some time. ODR-DabMux cannot handle such drift\n"
-    "because it would have to throw away or insert a full DAB+ superframe,\n"
-    "which would create audible artifacts. This drift compensation can\n"
-    "make sure that the encoding rate is correct by inserting or deleting\n"
-    "audio samples. It can be used for both ALSA and VLC inputs.\n"
-    "\n"
-    "When this option is enabled, you will see U and O printed in the\n"
-    "console. These correspond to audio underruns and overruns caused\n"
-    "by sound card clock drift. When sparse, they should not create audible\n"
-    "artifacts.\n"
-    "\n"
-    "This encoder includes PAD (DLS and MOT Slideshow) support by\n"
-    "http://rd.csp.it to be used with mot-encoder\n"
-    "\nUsage:\n"
-    "%s [INPUT SELECTION] [OPTION...]\n",
-#if defined(GITVERSION)
-    GITVERSION
-#else
-    PACKAGE_VERSION
-#endif
-    , name);
-    fprintf(stderr,
-    "   For the alsa input:\n"
-#if HAVE_ALSA
-    "     -d, --device=alsa_device             Set ALSA input device (default: \"default\").\n"
-#else
-    "     The Alsa input was disabled at compile time\n"
-#endif
-    "   For the file input:\n"
-    "     -i, --input=FILENAME                 Input filename (default: stdin).\n"
-    "     -f, --format={ wav, raw }            Set input file format (default: wav).\n"
-    "         --fifo-silence                   Input file is fifo and encoder generates silence when fifo is empty. Ignore EOF.\n"
-    "   For the JACK input:\n"
-#if HAVE_JACK
-    "     -j, --jack=name                      Enable JACK input, and define our name\n"
-#else
-    "     The JACK input was disabled at compile-time\n"
-#endif
-    "   For the VLC input:\n"
-#if HAVE_VLC
-    "     -v, --vlc-uri=uri                    Enable VLC input and use the URI given as source\n"
-    "     -C, --vlc-cache=ms                   Specify VLC network cache length.\n"
-    "     -g, --vlc-gain=db                    Enable VLC audio compressor, with given compressor-makeup value.\n"
-    "                                          Use this as a workaround to correct the gain for streams that are\n"
-    "                                          much too loud.\n"
-    "     -V                                   Increase the VLC verbosity by one (can be given \n"
-    "                                          multiple times)\n"
-    "     -L OPTION                            Give an additional options to VLC (can be given\n"
-    "                                          multiple times)\n"
-    "     -w, --write-icy-text=filename        Write the ICY Text into the file, so that mot-encoder can read it.\n"
-    "     -W, --write-icy-text-dl-plus         When writing the ICY Text into the file, add DL Plus information.\n"
-#else
-    "     The VLC input was disabled at compile-time\n"
-#endif
-    "   Drift compensation\n"
-    "     -D, --drift-comp                     Enable ALSA/VLC sound card drift compensation.\n"
-    "   Encoder parameters:\n"
-    "     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be a multiple of 8.\n"
-    "     -c, --channels={ 1, 2 }              Nb of input channels (default: 2).\n"
-    "     -r, --rate={ 32000, 48000 }          Input sample rate (default: 48000).\n"
-    "   DAB specific options\n"
-    "     -a, --dab                            Encode in DAB and not in DAB+.\n"
-    "         --dabmode=MODE                   Channel mode: s/d/j/m\n"
-    "                                          (default: j if stereo, m if mono).\n"
-    "         --dabpsy=PSY                     Psychoacoustic model 0/1/2/3\n"
-    "                                          (default: 1).\n"
-    "   DAB+ specific options\n"
-    "     -A, --no-afterburner                 Disable AAC encoder quality increaser.\n"
-    "         --aaclc                          Force the usage of AAC-LC (no SBR, no PS)\n"
-    "         --sbr                            Force the usage of SBR\n"
-    "         --ps                             Force the usage of PS\n"
-    "   Output and pad parameters:\n"
-    "     -o, --output=URI                     Output ZMQ uri. (e.g. 'tcp://localhost:9000')\n"
-    "                                     -or- Output file uri. (e.g. 'file.dabp')\n"
-    "                                     -or- a single dash '-' to denote stdout\n"
-    "                                          If more than one ZMQ output is given, the socket\n"
-    "                                          will be connected to all listed endpoints.\n"
-    "     -k, --secret-key=FILE                Enable ZMQ encryption with the given secret key.\n"
-    "     -p, --pad=BYTES                      Set PAD size in bytes.\n"
-    "     -P, --pad-fifo=FILENAME              Set PAD data input fifo name"
-    "                                          (default: /tmp/pad.fifo).\n"
-    "     -l, --level                          Show peak audio level indication.\n"
-    "     -s, --silence=TIMEOUT                Abort encoding after TIMEOUT seconds of silence.\n"
-    "\n"
-    "Only the tcp:// zeromq transport has been tested until now,\n"
-    " but epgm:// and pgm:// are also accepted\n"
-    );
-
-}
-
-/*! Setup the FDK AAC encoder
- *
- * \return 0 on success
- */
-int prepare_aac_encoder(
-        HANDLE_AACENCODER *encoder,
-        int subchannel_index,
-        int channels,
-        int sample_rate,
-        int afterburner,
-        int *aot)
-{
-    CHANNEL_MODE mode;
-    switch (channels) {
-        case 1: mode = MODE_1; break;
-        case 2: mode = MODE_2; break;
-        default:
-                fprintf(stderr, "Unsupported channels number %d\n", channels);
-                return 1;
-    }
-
-
-    if (aacEncOpen(encoder, 0x01|0x02|0x04, channels) != AACENC_OK) {
-        fprintf(stderr, "Unable to open encoder\n");
-        return 1;
-    }
-
-    if (*aot == AOT_NONE) {
-
-        if(channels == 2 && subchannel_index <= 6) {
-            *aot = AOT_DABPLUS_PS;
-        }
-        else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) {
-            *aot = AOT_DABPLUS_SBR;
-        }
-        else {
-            *aot = AOT_DABPLUS_AAC_LC;
-        }
-    }
-
-    fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
-            subchannel_index,
-            *aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
-            *aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
-            *aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
-            channels, sample_rate);
-
-    if (aacEncoder_SetParam(*encoder, AACENC_AOT, *aot) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the AOT\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the sample rate\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_CHANNELMODE, mode) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the channel mode\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_CHANNELORDER, 1) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the wav channel order\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the granule length\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the RAW transmux\n");
-        return 1;
-    }
-
-    /*if (aacEncoder_SetParam(*encoder, AACENC_BITRATEMODE, AACENC_BR_MODE_SFR)
-     * != AACENC_OK) {
-        fprintf(stderr, "Unable to set the bitrate mode\n");
-        return 1;
-    }*/
-
-
-    fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
-    if (aacEncoder_SetParam(*encoder, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the bitrate\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(*encoder, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the afterburner mode\n");
-        return 1;
-    }
-    if (!afterburner) {
-        fprintf(stderr, "Warning: Afterburned disabled!\n");
-    }
-    if (aacEncEncode(*encoder, NULL, NULL, NULL, NULL) != AACENC_OK) {
-        fprintf(stderr, "Unable to initialize the encoder\n");
-        return 1;
-    }
-    return 0;
-}
-
-chrono::steady_clock::time_point timepoint_last_compensation;
-
-/*! Wait the proper amount of time to throttle down to nominal encoding
- * rate, if drift compensation is enabled.
- */
-void drift_compensation_delay(int sample_rate, int channels, size_t bytes)
-{
-    const size_t bytes_per_second = sample_rate * BYTES_PER_SAMPLE * channels;
-
-    size_t bytes_compensate = bytes;
-    const auto wait_time = std::chrono::milliseconds(1000ul * bytes_compensate / bytes_per_second);
-    assert(1000ul * bytes_compensate % bytes_per_second == 0);
-
-    const auto curTime = std::chrono::steady_clock::now();
-
-    const auto diff = curTime - timepoint_last_compensation;
-
-    if (diff < wait_time) {
-        auto waiting = wait_time - diff;
-        std::this_thread::sleep_for(waiting);
-    }
-
-    timepoint_last_compensation += wait_time;
-}
-
-#define no_argument 0
-#define required_argument 1
-#define optional_argument 2
-
-#define STATUS_PAD_INSERTED 0x1
-#define STATUS_OVERRUN 0x2
-#define STATUS_UNDERRUN 0x4
-
-int main(int argc, char *argv[])
-{
-    int bitrate = 0; // 0 is default
-    int ch=0;
-
-    encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus;
-
-    // For the ALSA input
-    const char *alsa_device = NULL;
-
-    // For the file input
-    const char *infile = NULL;
-    int raw_input = 0;
-
-    // For the VLC input
-    std::string vlc_uri = "";
-    std::string vlc_icytext_file = "";
-    bool vlc_icytext_dlplus = false;
-    std::string vlc_gain = "";
-    std::string vlc_cache = "";
-    std::vector<std::string> vlc_additional_opts;
-    unsigned verbosity = 0;
-
-    // For the file output
-    FILE *out_fh = NULL;
-
-    const char *jack_name = NULL;
-
-    std::vector<std::string> output_uris;
-
-    int sample_rate=48000, channels=2;
-    void *rs_handler = NULL;
-    bool afterburner = true;
-    bool inFifoSilence = false;
-    bool drift_compensation = false;
-    AACENC_InfoStruct info = { 0 };
-    int aot = AOT_NONE;
-
-    char dab_channel_mode = '\0';
-    int  dab_psy_model = 1;
-    std::deque<uint8_t> toolame_output_buffer;
-
-    /* Keep track of peaks */
-    int peak_left  = 0;
-    int peak_right = 0;
-
-    /* On silence, die after the silence_timeout expires */
-    bool die_on_silence = false;
-    int silence_timeout = 0;
-    int measured_silence_ms = 0;
-
-    /* For MOT Slideshow and DLS insertion */
-    const char* pad_fifo = "/tmp/pad.fifo";
-    int pad_fd;
-    int padlen = 0;
-
-    /* Encoder status, see the above STATUS macros */
-    int status = 0;
-
-    /* Whether to show the 'sox'-like measurement */
-    int show_level = 0;
-
-    /* Data for ZMQ CURVE authentication */
-    char* keyfile = NULL;
-    char secretkey[CURVE_KEYLEN+1];
-
-    const struct option longopts[] = {
-        {"bitrate",                required_argument,  0, 'b'},
-        {"channels",               required_argument,  0, 'c'},
-        {"dabmode",                required_argument,  0,  4 },
-        {"dabpsy",                 required_argument,  0,  5 },
-        {"device",                 required_argument,  0, 'd'},
-        {"format",                 required_argument,  0, 'f'},
-        {"input",                  required_argument,  0, 'i'},
-        {"jack",                   required_argument,  0, 'j'},
-        {"output",                 required_argument,  0, 'o'},
-        {"pad",                    required_argument,  0, 'p'},
-        {"pad-fifo",               required_argument,  0, 'P'},
-        {"rate",                   required_argument,  0, 'r'},
-        {"secret-key",             required_argument,  0, 'k'},
-        {"silence",                required_argument,  0, 's'},
-        {"vlc-cache",              required_argument,  0, 'C'},
-        {"vlc-gain",               required_argument,  0, 'g'},
-        {"vlc-uri",                required_argument,  0, 'v'},
-        {"vlc-opt",                required_argument,  0, 'L'},
-        {"write-icy-text",         required_argument,  0, 'w'},
-        {"write-icy-text-dl-plus", no_argument,        0, 'W'},
-        {"aaclc",                  no_argument,        0,  0 },
-        {"dab",                    no_argument,        0, 'a'},
-        {"drift-comp",             no_argument,        0, 'D'},
-        {"fifo-silence",           no_argument,        0,  3 },
-        {"help",                   no_argument,        0, 'h'},
-        {"level",                  no_argument,        0, 'l'},
-        {"no-afterburner",         no_argument,        0, 'A'},
-        {"ps",                     no_argument,        0,  2 },
-        {"sbr",                    no_argument,        0,  1 },
-        {"verbosity",              no_argument,        0, 'V'},
-        {0, 0, 0, 0},
-    };
-
-    fprintf(stderr,
-            "Welcome to %s %s, compiled at %s, %s",
-            PACKAGE_NAME,
-#if defined(GITVERSION)
-            GITVERSION,
-#else
-            PACKAGE_VERSION,
-#endif
-            __DATE__, __TIME__);
-    fprintf(stderr, "\n");
-    fprintf(stderr, "  http://opendigitalradio.org\n\n");
-
-
-    if (argc < 2) {
-        usage(argv[0]);
-        return 1;
-    }
-
-    int index;
-    while(ch != -1) {
-        ch = getopt_long(argc, argv, "aAhDlVb:c:f:i:j:k:L:o:r:d:p:P:s:v:w:Wg:C:", longopts, &index);
-        switch (ch) {
-        case 0: // AAC-LC
-            aot = AOT_DABPLUS_AAC_LC;
-            break;
-        case 1: // SBR
-            aot = AOT_DABPLUS_SBR;
-            break;
-        case 2: // PS
-            aot = AOT_DABPLUS_PS;
-            break;
-        case 3: // FIFO SILENCE
-        case 4: // DAB channel mode
-            dab_channel_mode = optarg[0];
-            break;
-        case 5: // DAB psy model
-            dab_psy_model = atoi(optarg);
-            break;
-        case 'a':
-            selected_encoder = encoder_selection_t::toolame_dab;
-            break;
-        case 'A':
-            afterburner = false;
-            break;
-        case 'b':
-            bitrate = atoi(optarg);
-            break;
-        case 'c':
-            channels = atoi(optarg);
-            break;
-        case 'd':
-            alsa_device = optarg;
-            break;
-        case 'D':
-            drift_compensation = true;
-            break;
-        case 'f':
-            if(strcmp(optarg, "raw")==0) {
-                raw_input = 1;
-            } else if(strcmp(optarg, "wav")!=0)
-                usage(argv[0]);
-            break;
-        case 'i':
-            infile = optarg;
-            break;
-        case 'j':
-#if HAVE_JACK
-            jack_name = optarg;
-#else
-            fprintf(stderr, "JACK disabled at compile time!\n");
-            return 1;
-#endif
-            break;
-        case 'k':
-            keyfile = optarg;
-            break;
-        case 'l':
-            show_level = 1;
-            break;
-        case 'o':
-            output_uris.push_back(optarg);
-            break;
-        case 'p':
-            padlen = atoi(optarg);
-            break;
-        case 'P':
-            pad_fifo = optarg;
-            break;
-        case 'r':
-            sample_rate = atoi(optarg);
-            break;
-        case 's':
-            silence_timeout = atoi(optarg);
-            if (silence_timeout > 0 && silence_timeout < 3600*24*30) {
-                die_on_silence = true;
-            }
-            else {
-                fprintf(stderr, "Invalid silence timeout (%d) given!\n", silence_timeout);
-                return 1;
-            }
-
-            break;
-#ifdef HAVE_VLC
-        case 'v':
-            vlc_uri = optarg;
-            break;
-        case 'w':
-            vlc_icytext_file = optarg;
-            break;
-        case 'W':
-            vlc_icytext_dlplus = true;
-            break;
-        case 'g':
-            vlc_gain = optarg;
-            break;
-        case 'C':
-            vlc_cache = optarg;
-            break;
-        case 'L':
-            vlc_additional_opts.push_back(optarg);
-            break;
-#else
-        case 'v':
-        case 'w':
-            fprintf(stderr, "VLC input not enabled at compile time!\n");
-            return 1;
-#endif
-        case 'V':
-            verbosity++;
-            break;
-        case '?':
-        case 'h':
-            usage(argv[0]);
-            return 1;
-        }
-    }
-
-    int num_inputs = 0;
-    if (alsa_device) num_inputs++;
-    if (infile) num_inputs++;
-    if (jack_name) num_inputs++;
-    if (vlc_uri != "") num_inputs++;
-
-    if (num_inputs > 1) {
-        fprintf(stderr, "You must define only one possible input, not several!\n");
-        return 1;
-    }
-
-    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
-        if (bitrate == 0) {
-            bitrate = 64;
-        }
-
-        int subchannel_index = bitrate / 8;
-
-        if (subchannel_index < 1 || subchannel_index > 24) {
-            fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. Try other bitrate.\n",
-                    subchannel_index);
-            return 1;
-        }
-
-        if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
-            fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
-            return 1;
-        }
-    }
-    else if (selected_encoder == encoder_selection_t::toolame_dab) {
-        if (bitrate == 0) {
-            bitrate = 192;
-        }
-
-        if ( ! (sample_rate == 24000 || sample_rate == 48000)) {
-            fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n");
-            return 1;
-        }
-    }
-
-    if (padlen < 0) {
-        fprintf(stderr, "Invalid PAD length specified\n");
-        return 1;
-    }
-
-    zmq::context_t zmq_ctx;
-    zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
-
-    if (not output_uris.empty()) {
-        for (auto uri : output_uris) {
-            if (uri == "-") {
-                if (out_fh != NULL) {
-                    fprintf(stderr, "You can't write to more than one file!\n");
-                    return 1;
-                }
-                out_fh = stdout;
-            }
-            else if ((uri.compare(0, 6, "tcp://") == 0) ||
-                    (uri.compare(0, 6, "pgm://") == 0) ||
-                    (uri.compare(0, 7, "epgm://") == 0)) {
-                if (keyfile) {
-                    fprintf(stderr, "Enabling encryption\n");
-
-                    int rc = readkey(keyfile, secretkey);
-                    if (rc) {
-                        fprintf(stderr, "Error reading secret key\n");
-                        return 2;
-                    }
-
-                    const int yes = 1;
-                    zmq_sock.setsockopt(ZMQ_CURVE_SERVER,
-                            &yes, sizeof(yes));
-
-                    zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY,
-                            secretkey, CURVE_KEYLEN);
-                }
-                zmq_sock.connect(uri.c_str());
-            }
-            else { // We assume it's a file name
-                if (out_fh != NULL) {
-                    fprintf(stderr, "You can't write to more than one file!\n");
-                    return 1;
-                }
-
-                out_fh = fopen(uri.c_str(), "wb");
-
-                if (!out_fh) {
-                    fprintf(stderr, "Can't open output file!\n");
-                    return 1;
-                }
-            }
-        }
-    }
-    else {
-        fprintf(stderr, "No output URI defined\n");
-        return 1;
-    }
-
-    if (padlen != 0) {
-        int flags;
-        if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) {
-            if (errno != EEXIST) {
-                fprintf(stderr, "Can't create pad file: %d!\n", errno);
-                return 1;
-            }
-        }
-        pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK);
-        if (pad_fd == -1) {
-            fprintf(stderr, "Can't open pad file!\n");
-            return 1;
-        }
-        flags = fcntl(pad_fd, F_GETFL, 0);
-        if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) {
-            fprintf(stderr, "Can't set non-blocking mode in pad file!\n");
-            return 1;
-        }
-    }
-
-
-    std::vector<uint8_t> input_buf;
-
-    HANDLE_AACENCODER encoder;
-
-    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
-        int subchannel_index = bitrate / 8;
-        if (prepare_aac_encoder(&encoder, subchannel_index, channels,
-                    sample_rate, afterburner, &aot) != 0) {
-            fprintf(stderr, "Encoder preparation failed\n");
-            return 1;
-        }
-
-        if (aacEncInfo(encoder, &info) != AACENC_OK) {
-            fprintf(stderr, "Unable to get the encoder info\n");
-            return 1;
-        }
-
-        // Each DAB+ frame will need input_size audio bytes
-        const int input_size = channels * BYTES_PER_SAMPLE * info.frameLength;
-        fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n",
-                info.frameLength,
-                input_size);
-
-        input_buf.resize(input_size);
-    }
-    else if (selected_encoder == encoder_selection_t::toolame_dab) {
-        int err = toolame_init();
-
-        if (err == 0) {
-            err = toolame_set_samplerate(sample_rate);
-        }
-
-        if (err == 0) {
-            err = toolame_set_bitrate(bitrate);
-        }
-
-        if (err == 0) {
-            err = toolame_set_psy_model(dab_psy_model);
-        }
-
-        if (dab_channel_mode == '\0') {
-            if (channels == 2) {
-                dab_channel_mode = 'j'; // Default to joint-stereo
-            }
-            else if (channels == 1) {
-                dab_channel_mode = 'm'; // Default to mono
-            }
-            else {
-                fprintf(stderr, "Unsupported channels number %d\n", channels);
-                return 1;
-            }
-        }
-
-        if (err == 0) {
-            err = toolame_set_channel_mode(dab_channel_mode);
-        }
-
-        if (err == 0) {
-            err = toolame_set_pad(padlen);
-        }
-
-        if (err) {
-            fprintf(stderr, "libtoolame-dab init failed: %d\n", err);
-            return err;
-        }
-
-        input_buf.resize(channels * 1152 * BYTES_PER_SAMPLE);
-    }
-
-    /* We assume that we need to call the encoder
-     * enc_calls_per_output before it gives us one encoded audio
-     * frame. This information is used when the alsa drift compensation
-     * is active. This is only valid for FDK-AAC.
-     */
-    const int enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ?
-        sample_rate / 8000 :
-        sample_rate / 16000;
-
-    int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL;
-
-    /*! The SampleQueue \c queue is given to the inputs, so that they
-     * can fill it.
-     */
-    SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
-
-    /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
-    rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
-    if (rs_handler == NULL) {
-        perror("init_rs_char failed");
-        return 1;
-    }
-
-    /* No input defined ? default to alsa "default" */
-    if (!alsa_device) {
-        alsa_device = "default";
-    }
-
-    // We'll use one of the tree possible inputs
-#if HAVE_ALSA
-    AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
-    AlsaInputDirect   alsa_in_direct(alsa_device, channels, sample_rate);
-#endif
-    FileInput         file_in(infile, raw_input, sample_rate);
-#if HAVE_JACK
-    JackInput         jack_in(jack_name, channels, sample_rate, queue);
-#endif
-#if HAVE_VLC
-    VLCInput          vlc_input(vlc_uri, sample_rate, channels, verbosity, vlc_gain, vlc_cache, vlc_additional_opts, queue);
-#endif
-
-    if (infile) {
-        if (file_in.prepare() != 0) {
-            fprintf(stderr, "File input preparation failed\n");
-            return 1;
-        }
-    }
-#if HAVE_JACK
-    else if (jack_name) {
-        if (jack_in.prepare() != 0) {
-            fprintf(stderr, "JACK preparation failed\n");
-            return 1;
-        }
-    }
-#endif
-#if HAVE_VLC
-    else if (vlc_uri != "") {
-        if (vlc_input.prepare() != 0) {
-            fprintf(stderr, "VLC with drift compensation: preparation failed\n");
-            return 1;
-        }
-
-        fprintf(stderr, "Start VLC thread\n");
-        vlc_input.start();
-    }
-#endif
-#if HAVE_ALSA
-    else if (drift_compensation) {
-        if (alsa_in_threaded.prepare() != 0) {
-            fprintf(stderr, "Alsa with drift compensation: preparation failed\n");
-            return 1;
-        }
-
-        fprintf(stderr, "Start ALSA capture thread\n");
-        alsa_in_threaded.start();
-    }
-    else {
-        if (alsa_in_direct.prepare() != 0) {
-            fprintf(stderr, "Alsa preparation failed\n");
-            return 1;
-        }
-    }
-#else
-    else {
-        fprintf(stderr, "No input defined\n");
-        return 1;
-    }
-#endif
-
-    int outbuf_size;
-    std::vector<uint8_t> zmqframebuf;
-    std::vector<uint8_t> outbuf;
-    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
-        outbuf_size = bitrate/8*120;
-        outbuf.resize(24*120);
-        zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120);
-
-        if(outbuf_size % 5 != 0) {
-            fprintf(stderr, "Warning: (outbuf_size mod 5) = %d\n", outbuf_size % 5);
-        }
-    }
-    else if (selected_encoder == encoder_selection_t::toolame_dab) {
-        outbuf_size = 4092;
-        outbuf.resize(outbuf_size);
-        fprintf(stderr, "Setting outbuf size to %zu\n", outbuf.size());
-
-        // ODR-DabMux expects frames of length 3*bitrate
-        zmqframebuf.resize(ZMQ_HEADER_SIZE + 3 * bitrate);
-    }
-
-    zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0];
-
-    unsigned char pad_buf[padlen + 1];
-
-    fprintf(stderr, "Starting encoding\n");
-
-    int retval = 0;
-    int send_error_count = 0;
-    timepoint_last_compensation = chrono::steady_clock::now();
-
-    int calls = 0; // for checking
-    ssize_t read_bytes = 0;
-    do {
-        AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
-
-        // --------------- Read data from the PAD fifo
-        int ret;
-        if (padlen != 0) {
-            ret = read(pad_fd, pad_buf, padlen + 1);
-        }
-        else {
-            ret = 0;
-        }
-
-
-        if(ret < 0 && errno == EAGAIN) {
-            // If this condition passes, there is no data to be read
-            in_buf.numBufs = 1;    // Samples;
-        }
-        else if(ret >= 0) {
-            // Otherwise, you're good to go and buffer should contain "count" bytes.
-            in_buf.numBufs = 2;    // Samples + Data;
-            if (ret > 0)
-                status |= STATUS_PAD_INSERTED;
-        }
-        else {
-            // Some other error occurred during read.
-            fprintf(stderr, "Unable to read from PAD!\n");
-                        break;
-        }
-
-        // -------------- Read Data
-        memset(&outbuf[0], 0x00, outbuf_size);
-        memset(&input_buf[0], 0x00, input_buf.size());
-
-        /*! \section DataInput
-         * We read data input either in a blocking way (file input, VLC or ALSA
-         * without drift compensation) or in a non-blocking way (VLC or ALSA
-         * with drift compensation, JACK).
-         *
-         * The file input doesn't need the queue at all. But the other inputs
-         * do, and either use \c pop() or \c pop_wait() depending on if it's blocking or not
-         *
-         * In non-blocking, the \c queue makes the data available without delay, and the
-         * \c drift_compensation_delay() function handles rate throttling.
-         */
-
-        if (infile) {
-            read_bytes = file_in.read(&input_buf[0], input_buf.size());
-            if (read_bytes < 0) {
-                break;
-            }
-            else if (read_bytes != input_buf.size()) {
-                if (inFifoSilence && file_in.eof()) {
-                    memset(&input_buf[0], 0, input_buf.size());
-                    read_bytes = input_buf.size();
-                    usleep((long)input_buf.size() * 1000000 /
-                            (BYTES_PER_SAMPLE * channels * sample_rate));
-                }
-                else {
-                    fprintf(stderr, "Short file read !\n");
-                    read_bytes = 0;
-                }
-            }
-        }
-#if HAVE_VLC
-        else if (not vlc_uri.empty()) {
-
-            if (drift_compensation && vlc_input.fault_detected()) {
-                fprintf(stderr, "Detected fault in VLC input!\n");
-                retval = 5;
-                break;
-            }
-
-            if (drift_compensation) {
-                size_t overruns;
-                size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
-                read_bytes = input_buf.size();
-                drift_compensation_delay(sample_rate, channels, read_bytes);
-
-                if (bytes_from_queue != input_buf.size()) {
-                    status |= STATUS_UNDERRUN;
-                }
-
-                if (overruns) {
-                    status |= STATUS_OVERRUN;
-                }
-            }
-            else {
-                const int timeout_ms = 1000;
-                read_bytes = input_buf.size();
-
-                /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do
-                 * its job.
-                 */
-                size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes
-
-                if (bytes_from_queue < read_bytes) {
-                    // queue timeout occurred
-                    fprintf(stderr, "Detected fault in VLC input! No data in time.\n");
-                    retval = 5;
-                    break;
-                }
-            }
-
-            if (not vlc_icytext_file.empty()) {
-                vlc_input.write_icy_text(vlc_icytext_file, vlc_icytext_dlplus);
-            }
-        }
-#endif
-        else if (drift_compensation || jack_name) {
-#if HAVE_ALSA
-            if (drift_compensation && alsa_in_threaded.fault_detected()) {
-                fprintf(stderr, "Detected fault in alsa input!\n");
-                retval = 5;
-                break;
-            }
-#endif
-
-            size_t overruns;
-            size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
-            read_bytes = input_buf.size();
-            drift_compensation_delay(sample_rate, channels, read_bytes);
-
-            if (bytes_from_queue != input_buf.size()) {
-                status |= STATUS_UNDERRUN;
-            }
-
-            if (overruns) {
-                status |= STATUS_OVERRUN;
-            }
-        }
-        else {
-#if HAVE_ALSA
-            read_bytes = alsa_in_direct.read(&input_buf[0], input_buf.size());
-            if (read_bytes < 0) {
-                break;
-            }
-            else if (read_bytes != input_buf.size()) {
-                fprintf(stderr, "Short alsa read !\n");
-            }
-#endif
-        }
-
-        /*! \section AudioLevel
-         * Audio level measurement is always done assuming we have two
-         * channels, and is formally wrong in mono, but still gives
-         * numbers one can use.
-         *
-         * \todo fix level measurement in mono
-         */
-        for (int i = 0; i < read_bytes; i+=4) {
-            int16_t l = input_buf[i] | (input_buf[i+1] << 8);
-            int16_t r = input_buf[i+2] | (input_buf[i+3] << 8);
-            peak_left  = MAX(peak_left,  l);
-            peak_right = MAX(peak_right, r);
-        }
-
-        /*! \section SilenceDetection
-         * Silence detection looks at the audio level and is
-         * only useful if the connection dropped, or if no data is available. It is not
-         * useful if the source is nearly silent (some noise present), because the
-         * threshold is 0, and not configurable. The rationale is that we want to
-         * guard against connection issues, not source level issues
-         */
-        if (die_on_silence && MAX(peak_left, peak_right) == 0) {
-            const unsigned int frame_time_msec = 1000ul *
-                read_bytes / (BYTES_PER_SAMPLE * channels * sample_rate);
-
-            measured_silence_ms += frame_time_msec;
-
-            if (measured_silence_ms > 1000*silence_timeout) {
-                fprintf(stderr, "Silence detected for %d seconds, aborting.\n",
-                        silence_timeout);
-                retval = 2;
-                break;
-            }
-        }
-        else {
-            measured_silence_ms = 0;
-        }
-
-        int numOutBytes = 0;
-        if (read_bytes and
-                selected_encoder == encoder_selection_t::fdk_dabplus) {
-            AACENC_InArgs in_args = { 0 };
-            AACENC_OutArgs out_args = { 0 };
-            // -------------- AAC Encoding
-            //
-            int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
-            int out_identifier = OUT_BITSTREAM_DATA;
-            const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
-            const int subchannel_index = bitrate / 8;
-
-            void *in_ptr[2], *out_ptr;
-            int in_size[2], in_elem_size[2];
-            int out_size, out_elem_size;
-
-            in_ptr[0] = &input_buf[0];
-            in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
-            in_size[0] = read_bytes;
-            in_size[1] = calculated_padlen;
-            in_elem_size[0] = BYTES_PER_SAMPLE;
-            in_elem_size[1] = sizeof(uint8_t);
-            in_args.numInSamples = input_buf.size()/BYTES_PER_SAMPLE;
-            in_args.numAncBytes = calculated_padlen;
-
-            in_buf.bufs = (void**)&in_ptr;
-            in_buf.bufferIdentifiers = in_identifier;
-            in_buf.bufSizes = in_size;
-            in_buf.bufElSizes = in_elem_size;
-
-            out_ptr = &outbuf[0];
-            out_size = outbuf.size();
-            out_elem_size = 1;
-            out_buf.numBufs = 1;
-            out_buf.bufs = &out_ptr;
-            out_buf.bufferIdentifiers = &out_identifier;
-            out_buf.bufSizes = &out_size;
-            out_buf.bufElSizes = &out_elem_size;
-
-            AACENC_ERROR err;
-            if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args))
-                    != AACENC_OK) {
-                if (err == AACENC_ENCODE_EOF) {
-                    fprintf(stderr, "encoder error: EOF reached\n");
-                    break;
-                }
-                fprintf(stderr, "Encoding failed (%d)\n", err);
-                retval = 3;
-                break;
-            }
-            calls++;
-
-            numOutBytes = out_args.numOutBytes;
-        }
-        else if (selected_encoder == encoder_selection_t::toolame_dab) {
-            int calculated_padlen = 0;
-            if (ret == padlen + 1) {
-                calculated_padlen = pad_buf[padlen];
-                if (calculated_padlen <= 2) {
-                    stringstream ss;
-                    ss << "Invalid XPAD Length " << calculated_padlen;
-                    throw runtime_error(ss.str());
-                }
-            }
-
-            /*! \note toolame expects the audio to be in another shape as
-             * we have in input_buf, and we need to convert first
-             */
-            short input_buffers[2][1152];
-
-            if (channels == 1) {
-                memcpy(input_buffers[0], &input_buf[0], 1152 * BYTES_PER_SAMPLE);
-            }
-            else if (channels == 2) {
-                for (int i = 0; i < 1152; i++) {
-                    int16_t l = input_buf[4*i]   | (input_buf[4*i+1] << 8);
-                    int16_t r = input_buf[4*i+2] | (input_buf[4*i+3] << 8);
-
-                    input_buffers[0][i] = l;
-                    input_buffers[1][i] = r;
-                }
-            }
-            else {
-                fprintf(stderr, "INTERNAL ERROR! invalid number of channels\n");
-            }
-
-            if (read_bytes) {
-                numOutBytes = toolame_encode_frame(input_buffers, pad_buf, calculated_padlen, &outbuf[0], outbuf.size());
-            }
-            else {
-                numOutBytes = toolame_finish(&outbuf[0], outbuf.size());
-            }
-        }
-
-        /* Check if the encoder has generated output data.
-         * DAB+ requires RS encoding, which is not done in ODR-DabMux and not necessary
-         * for DAB.
-         */
-        if (numOutBytes != 0 and
-            selected_encoder == encoder_selection_t::fdk_dabplus) {
-
-            // Our timing code depends on this
-            if (calls != enc_calls_per_output) {
-                fprintf(stderr, "INTERNAL ERROR! calls=%d"
-                        ", expected %d\n",
-                        calls, enc_calls_per_output);
-            }
-            calls = 0;
-
-            int row, col;
-            unsigned char buf_to_rs_enc[110];
-            unsigned char rs_enc[10];
-            const int subchannel_index = bitrate / 8;
-            for(row=0; row < subchannel_index; row++) {
-                for(col=0;col < 110; col++) {
-                    buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
-                }
-
-                encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
-
-                for(col=110; col<120; col++) {
-                    outbuf[subchannel_index * col + row] = rs_enc[col-110];
-                    assert(subchannel_index * col + row < outbuf_size);
-                }
-            }
-
-            numOutBytes = outbuf_size;
-        }
-
-        if (numOutBytes != 0) {
-            if (out_fh) {
-                fwrite(&outbuf[0], 1, numOutBytes, out_fh);
-            }
-            else {
-                // ------------ ZeroMQ transmit
-                try {
-                    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
-                        zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
-                        zmq_frame_header->version = 1;
-                        zmq_frame_header->datasize = numOutBytes;
-                        zmq_frame_header->audiolevel_left = peak_left;
-                        zmq_frame_header->audiolevel_right = peak_right;
-
-                        assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= zmqframebuf.size());
-
-                        memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
-                                &outbuf[0], numOutBytes);
-
-                        zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
-                                ZMQ_DONTWAIT);
-
-                    }
-                    else if (selected_encoder == encoder_selection_t::toolame_dab) {
-                        toolame_output_buffer.insert(toolame_output_buffer.end(),
-                                outbuf.begin(), outbuf.begin() + numOutBytes);
-
-                        while (toolame_output_buffer.size() > 3 * bitrate) {
-                            zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME;
-                            zmq_frame_header->version = 1;
-                            zmq_frame_header->datasize = 3 * bitrate;
-                            zmq_frame_header->audiolevel_left = peak_left;
-                            zmq_frame_header->audiolevel_right = peak_right;
-
-                            uint8_t *encoded_frame = ZMQ_FRAME_DATA(zmq_frame_header);
-
-                            // no memcpy for std::deque
-                            for (size_t i = 0; i < 3*bitrate; i++) {
-                                encoded_frame[i] = toolame_output_buffer[i];
-                            }
-
-                            zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
-                                    ZMQ_DONTWAIT);
-
-                            toolame_output_buffer.erase(toolame_output_buffer.begin(),
-                                                        toolame_output_buffer.begin() + 3 * bitrate);
-                        }
-                    }
-                }
-                catch (zmq::error_t& e) {
-                    fprintf(stderr, "ZeroMQ send error !\n");
-                    send_error_count ++;
-                }
-
-                if (send_error_count > 10)
-                {
-                    fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n");
-                    retval = 4;
-                    break;
-                }
-            }
-        }
-
-        if (numOutBytes != 0)
-        {
-            if (show_level) {
-                if (channels == 1) {
-                    fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s",
-                            level(1, MAX(peak_right, peak_left)),
-                            status & STATUS_PAD_INSERTED ? "P" : " ",
-                            status & STATUS_UNDERRUN ? "U" : " ",
-                            status & STATUS_OVERRUN ? "O" : " ");
-                }
-                else if (channels == 2) {
-                    fprintf(stderr, "\rIn: [%6s|%-6s] %1s %1s %1s",
-                            level(0, peak_left),
-                            level(1, peak_right),
-                            status & STATUS_PAD_INSERTED ? "P" : " ",
-                            status & STATUS_UNDERRUN ? "U" : " ",
-                            status & STATUS_OVERRUN ? "O" : " ");
-                }
-            }
-            else {
-                if (status & STATUS_OVERRUN) {
-                    fprintf(stderr, "O");
-                }
-
-                if (status & STATUS_UNDERRUN) {
-                    fprintf(stderr, "U");
-                }
-
-            }
-
-            peak_right = 0;
-            peak_left = 0;
-
-            status = 0;
-        }
-
-        fflush(stdout);
-    } while (read_bytes > 0);
-
-    fprintf(stderr, "\n");
-
-    if (out_fh) {
-        fclose(out_fh);
-    }
-
-    zmq_sock.close();
-    free_rs_char(rs_handler);
-
-    if (selected_encoder == encoder_selection_t::fdk_dabplus) {
-        aacEncClose(&encoder);
-    }
-
-    return retval;
-}
-