merge file and alsa encoders into dabplus-enc

There was a lot of redundant code between the two

1 files changed, 716 insertions, 0 deletions

diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp
new file mode 100644
index 0000000..82780d5
--- /dev/null
+++ b/src/dabplus-enc.cpp
@@ -0,0 +1,716 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ * Copyright (C) 2013,2014 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.
+ * -------------------------------------------------------------------
+ */
+
+#include "AlsaInput.h"
+#include "FileInput.h"
+#include "SampleQueue.h"
+#include "zmq.hpp"
+
+extern "C" {
+#include "encryption.h"
+#include "utils.h"
+#include "wavreader.h"
+}
+
+#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 "libAACenc/include/aacenc_lib.h"
+
+extern "C" {
+#include <fec.h>
+}
+
+using namespace std;
+
+void usage(const char* name) {
+    fprintf(stderr,
+    "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n"
+    "based on fdk-aac-dabplus that can read from a ALSA or file source\n"
+    "and encode to a ZeroMQ output for ODR-DabMux.\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.\n"
+    "\n"
+    "When this option is enabled, you will see U and O<number> printed in\n"
+    "the 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"
+    "\n"
+    "  http://opendigitalradio.org\n"
+    "\nUsage:\n"
+    "%s (-i file|-d alsa_device) [OPTION...]\n",
+#if defined(GITVERSION)
+    GITVERSION
+#else
+    PACKAGE_VERSION
+#endif
+    , name);
+    fprintf(stderr,
+    "   For the alsa input:\n"
+    "     -d, --device=alsa_device             Set ALSA input device (default: \"default\").\n"
+    "     -D, --drift-comp                     Enable ALSA sound card drift compensation.\n"
+    "   For the file input:\n"
+    "     -i, --input=FILENAME                 Input filename (default: stdin).\n"
+    "     -f, --format={ wav, raw }            Set input file format (default: wav).\n"
+    "   Encoder parameters:\n"
+    "     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be 8 multiple.\n"
+    "     -o, --output=URI                     Output zmq uri. (e.g. 'tcp://*:9000')\n"
+    "                                     -or- Output file uri. (e.g. 'file.dab')\n"
+    "                                     -or- a single dash '-' to denote stdout\n"
+    "     -a, --afterburner                    Turn on AAC encoder quality increaser.\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"
+    "     -c, --channels={ 1, 2 }              Nb of input channels (default: 2).\n"
+    "     -r, --rate={ 32000, 48000 }          Input sample rate (default: 48000).\n"
+    "     -k, --secret-key=FILE                Enable ZMQ encryption with the given secret key.\n"
+    "     -l, --level                          Show peak audio level indication.\n"
+    "\n"
+    "Only the tcp:// zeromq transport has been tested until now,\n"
+    " but epgm:// and pgm:// are also accepted\n"
+    );
+
+}
+
+int prepare_aac_encoder(
+        HANDLE_AACENCODER *encoder,
+        int subchannel_index,
+        int channels,
+        int sample_rate,
+        int afterburner)
+{
+    HANDLE_AACENCODER handle = *encoder;
+
+    int aot = AOT_DABPLUS_AAC_LC;
+
+    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(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
+        fprintf(stderr, "Unable to open encoder\n");
+        return 1;
+    }
+
+    *encoder = handle;
+
+    if(channels == 2 && subchannel_index <= 6)
+        aot = AOT_DABPLUS_PS;
+    else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
+        aot = AOT_DABPLUS_SBR;
+
+    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(handle, AACENC_AOT, aot) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the AOT\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the sample rate\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the channel mode\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the wav channel order\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the granule length\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the RAW transmux\n");
+        return 1;
+    }
+
+    /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *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(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the bitrate\n");
+        return 1;
+    }
+    if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+        fprintf(stderr, "Unable to set the afterburner mode\n");
+        return 1;
+    }
+    if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+        fprintf(stderr, "Unable to initialize the encoder\n");
+        return 1;
+    }
+    return 0;
+}
+
+#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 subchannel_index = 8; //64kbps subchannel
+    int ch=0;
+
+    // For the ALSA input
+    const char *alsa_device = NULL;
+
+    // For the file input
+    const char *infile = NULL;
+    int raw_input = 0;
+
+    // For the file output
+    FILE *out_fh;
+
+    const char *outuri = NULL;
+    int sample_rate=48000, channels=2;
+    const int bytes_per_sample = 2;
+    void *rs_handler = NULL;
+    bool afterburner = false;
+    bool drift_compensation = false;
+    AACENC_InfoStruct info = { 0 };
+
+    /* Keep track of peaks */
+    int peak_left  = 0;
+    int peak_right = 0;
+
+    /* For MOT Slideshow and DLS insertion */
+    const char* pad_fifo = "/tmp/pad.fifo";
+    int pad_fd;
+    unsigned char pad_buf[128];
+    int padlen;
+
+    /* 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'},
+        {"device",        required_argument,  0, 'd'},
+        {"format",        required_argument,  0, 'f'},
+        {"input",         required_argument,  0, 'i'},
+        {"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'},
+        {"afterburner",   no_argument,        0, 'a'},
+        {"drift-comp",    no_argument,        0, 'D'},
+        {"help",          no_argument,        0, 'h'},
+        {"level",         no_argument,        0, 'l'},
+        {0,0,0,0},
+    };
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    int index;
+    while(ch != -1) {
+        ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index);
+        switch (ch) {
+        case 'a':
+            afterburner = true;
+            break;
+        case 'b':
+            subchannel_index = atoi(optarg) / 8;
+            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 'k':
+            keyfile = optarg;
+            break;
+        case 'l':
+            show_level = 1;
+            break;
+        case 'o':
+            outuri = optarg;
+            break;
+        case 'p':
+            padlen = atoi(optarg);
+            break;
+        case 'P':
+            pad_fifo = optarg;
+            break;
+        case 'r':
+            sample_rate = atoi(optarg);
+            break;
+        case '?':
+        case 'h':
+            usage(argv[0]);
+            return 1;
+        }
+    }
+
+    if (alsa_device && infile) {
+        fprintf(stderr, "You must define either alsa or file input, not both\n");
+        return 1;
+    }
+
+    if (subchannel_index < 1 || subchannel_index > 24) {
+        fprintf(stderr, "Bad subchannels number: %d, 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;
+    }
+
+    /* 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
+     */
+    const int enc_calls_per_output = sample_rate / 16000;
+
+    zmq::context_t zmq_ctx;
+    zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
+
+    if (outuri) {
+        if (strcmp(outuri, "-") == 0) {
+            out_fh = stdout;
+        }
+        else if ((strncmp(outuri, "tcp://", 6) == 0) ||
+                (strncmp(outuri, "pgm://", 6) == 0) ||
+                (strncmp(outuri, "epgm://", 7) == 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(outuri);
+        }
+        else { // We assume it's a file name
+            out_fh = fopen(outuri, "wb");
+
+            if (!out_fh) {
+                fprintf(stderr, "Can't open output file!\n");
+                return 1;
+            }
+        }
+    }
+    else {
+        fprintf(stderr, "Output URI not 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;
+        }
+    }
+
+
+    HANDLE_AACENCODER encoder;
+
+    if (prepare_aac_encoder(&encoder, subchannel_index, channels,
+                sample_rate, afterburner) != 0) {
+        fprintf(stderr, "Encoder preparation failed\n");
+        return 2;
+    }
+
+    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);
+
+    uint8_t input_buf[input_size];
+
+    int max_size = 2*input_size + NUM_SAMPLES_PER_CALL;
+
+    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
+    AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
+    AlsaInputDirect   alsa_in_direct(alsa_device, channels, sample_rate);
+    FileInput         file_in(infile, raw_input, sample_rate);
+
+    if (infile) {
+        if (file_in.prepare() != 0) {
+            fprintf(stderr, "File input preparation failed\n");
+            return 1;
+        }
+    }
+    else if (drift_compensation) {
+        if (alsa_in_threaded.prepare() != 0) {
+            fprintf(stderr, "Alsa 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;
+        }
+    }
+
+    int outbuf_size = subchannel_index*120;
+    uint8_t zmqframebuf[2048];
+    zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf;
+
+    uint8_t outbuf[2048];
+
+    if(outbuf_size % 5 != 0) {
+        fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
+    }
+
+    fprintf(stderr, "Starting encoding\n");
+
+    int send_error_count = 0;
+    struct timespec tp_next;
+    clock_gettime(CLOCK_MONOTONIC, &tp_next);
+
+    int calls = 0; // for checking
+    while (1) {
+        int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
+        int out_identifier = OUT_BITSTREAM_DATA;
+
+        AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+        AACENC_InArgs in_args = { 0 };
+        AACENC_OutArgs out_args = { 0 };
+        void *in_ptr[2], *out_ptr;
+        int in_size[2], in_elem_size[2];
+        int out_size, out_elem_size;
+
+
+        // -------------- wait the right amount of time
+        if (drift_compensation) {
+            struct timespec tp_now;
+            clock_gettime(CLOCK_MONOTONIC, &tp_now);
+
+            unsigned long time_now  = (1000000000ul * tp_now.tv_sec) +
+                tp_now.tv_nsec;
+            unsigned long time_next = (1000000000ul * tp_next.tv_sec) +
+                tp_next.tv_nsec;
+
+            const unsigned long dabplus_superframe_nsec = 120000000ul;
+
+            const unsigned long wait_time =
+                dabplus_superframe_nsec / enc_calls_per_output;
+
+            unsigned long waiting = wait_time - (time_now - time_next);
+            if ((time_now - time_next) < wait_time) {
+                //printf("Sleep %zuus\n", waiting / 1000);
+                usleep(waiting / 1000);
+            }
+
+            // Move our time_counter 60ms into the future.
+            // The encoder needs two calls for one frame
+            tp_next.tv_nsec += wait_time;
+            if (tp_next.tv_nsec >  1000000000L) {
+                tp_next.tv_nsec -= 1000000000L;
+                tp_next.tv_sec  += 1;
+            }
+        }
+
+        // --------------- Read data from the PAD fifo
+        int ret;
+        if (padlen != 0) {
+            ret = read(pad_fd, pad_buf, padlen);
+        }
+        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, 0x00, outbuf_size);
+        memset(input_buf, 0x00, input_size);
+
+        ssize_t read;
+        if (infile) {
+            read = file_in.read(input_buf, input_size);
+            if (read < 0) {
+                break;
+            }
+            else if (read != input_size) {
+                fprintf(stderr, "Short file read !\n");
+                break;
+            }
+        }
+        else if (drift_compensation) {
+            if (alsa_in_threaded.fault_detected()) {
+                fprintf(stderr, "Detected fault in alsa input!\n");
+                break;
+            }
+
+            size_t overruns;
+            read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+
+            if (read != input_size) {
+                status |= STATUS_UNDERRUN;
+            }
+
+            if (overruns) {
+                status |= STATUS_OVERRUN;
+            }
+        }
+        else {
+            read = alsa_in_direct.read(input_buf, input_size);
+            if (read < 0) {
+                break;
+            }
+            else if (read != input_size) {
+                fprintf(stderr, "Short alsa read !\n");
+            }
+        }
+
+        for (int i = 0; i < read; 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);
+        }
+
+        // -------------- AAC Encoding
+
+        in_ptr[0] = input_buf;
+        in_ptr[1] = pad_buf;
+        in_size[0] = read;
+        in_size[1] = padlen;
+        in_elem_size[0] = BYTES_PER_SAMPLE;
+        in_elem_size[1] = sizeof(uint8_t);
+        in_args.numInSamples = input_size/BYTES_PER_SAMPLE;
+        in_args.numAncBytes = 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;
+        out_size = sizeof(outbuf);
+        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);
+            break;
+        }
+        calls++;
+
+        /* Check if the encoder has generated output data */
+        if (out_args.numOutBytes != 0)
+        {
+            // Our timing code depends on this
+            if (! ((sample_rate == 32000 && calls == 2) ||
+                   (sample_rate == 48000 && calls == 3)) ) {
+                fprintf(stderr, "INTERNAL ERROR! sample rate %d, calls %d\n",
+                    sample_rate, calls);
+                }
+            calls = 0;
+
+            // ----------- RS encoding
+            int row, col;
+            unsigned char buf_to_rs_enc[110];
+            unsigned char rs_enc[10];
+            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);
+                }
+            }
+
+            // ------------ ZeroMQ transmit
+            try {
+                zmq_frame_header->version = 1;
+                zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
+                zmq_frame_header->datasize = outbuf_size;
+                zmq_frame_header->audiolevel_left = peak_left;
+                zmq_frame_header->audiolevel_right = peak_right;
+
+                memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
+                        outbuf, outbuf_size);
+
+                zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header),
+                        ZMQ_DONTWAIT);
+            }
+            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");
+                break;
+            }
+
+            if (show_level && out_args.numOutBytes + row*10 == outbuf_size) {
+                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" : " ");
+            }
+
+            status = 0;
+        }
+
+        fflush(stdout);
+    }
+    fprintf(stderr, "\n");
+
+    zmq_sock.close();
+    free_rs_char(rs_handler);
+
+    aacEncClose(&encoder);
+}
+