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-rw-r--r--alsa-dabplus-zmq.c622
1 files changed, 622 insertions, 0 deletions
diff --git a/alsa-dabplus-zmq.c b/alsa-dabplus-zmq.c
new file mode 100644
index 0000000..14e6aa8
--- /dev/null
+++ b/alsa-dabplus-zmq.c
@@ -0,0 +1,622 @@
+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ * Copyright (C) 2013 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 <stdio.h>
+#include <stdint.h>
+#include <string.h>
+#include <alloca.h>
+#include <math.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <getopt.h>
+#include <zmq.h>
+#include <assert.h>
+#include "libAACenc/include/aacenc_lib.h"
+#include <error.h>
+#include <signal.h>
+#include <alsa/asoundlib.h>
+
+#include <fec.h>
+
+static struct {
+ snd_pcm_format_t format;
+ unsigned int channels;
+ unsigned int rate;
+} hwparams;
+
+
+void usage(const char* name) {
+ fprintf(stderr, "%s [OPTION...]\n", name);
+ fprintf(stderr,
+" -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+//" -d, --data=FILENAME Set data filename.\n"
+//" -g, --fs-bug Turn on FS bug mitigation.\n"
+//" -i, --input=FILENAME Input filename (default: stdin).\n"
+" -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n"
+" -a, --afterburner Turn on AAC encoder quality increaser.\n"
+//" -m, --message Turn on AAC frame messages.\n"
+//" -p, --pad=BYTES Set PAD size in bytes.\n"
+//" -f, --format={ wav, raw } Set input file format (default: wav).\n"
+" -d, --device=alsa_device Set ALSA input device (default: \"default\").\n"
+" -c, --channels={ 1, 2 } Nb of input channels for raw input (default: 2).\n"
+" -r, --rate={ 32000, 48000 } Sample rate for raw input (default: 48000).\n"
+//" -t, --type=TYPE Set data type (dls|pad|packet|dg).\n"
+//" -v, --verbose=LEVEL Set verbosity level.\n"
+//" -V, --version Print version and exit.\n"
+//" --mi=[ 0, ... ] Set AAC frame messages interval in milliseconds.\n"
+//" --ma=[ 0, ... ] Set AAC frame messages attack time in milliseconds.\n"
+//" -l, --lp Set frame size to 1024 instead of 960.\n"
+"\n"
+"Only the tcp:// zeromq transport has been tested until now.\n"
+
+);
+
+}
+
+static int in_aborting = 0;
+static snd_pcm_t *alsa_handle = NULL;
+
+static void prg_exit(int code)
+{
+ if (alsa_handle)
+ snd_pcm_close(alsa_handle);
+ exit(code);
+}
+
+static void signal_handler(int sig)
+{
+ if (in_aborting)
+ return;
+
+ in_aborting = 1;
+ if (alsa_handle)
+ snd_pcm_abort(alsa_handle);
+
+ if (sig == SIGABRT) {
+ /* do not call snd_pcm_close() and abort immediately */
+ alsa_handle = NULL;
+ exit(EXIT_FAILURE);
+ }
+ signal(sig, signal_handler);
+}
+
+const static int dump_hw_params = 0;
+
+// Set Alsa hardware parameters
+static void set_params(void)
+{
+ snd_pcm_hw_params_t *params;
+ snd_pcm_sw_params_t *swparams;
+ snd_pcm_uframes_t buffer_size;
+ int err;
+ size_t n;
+ unsigned int rate;
+ snd_pcm_uframes_t start_threshold, stop_threshold;
+ snd_pcm_hw_params_alloca(&params);
+ snd_pcm_sw_params_alloca(&swparams);
+ err = snd_pcm_hw_params_any(alsa_handle, params);
+ if (err < 0) {
+ fprintf(stderr, "Broken configuration for this PCM: no configurations available");
+ prg_exit(EXIT_FAILURE);
+ }
+ if (dump_hw_params) {
+ fprintf(stderr, "HW Params of device \"%s\":\n",
+ snd_pcm_name(alsa_handle));
+ fprintf(stderr, "--------------------\n");
+ // TODO log should be a snd_output_t *log;
+ snd_pcm_hw_params_dump(params, log);
+ fprintf(stderr, "--------------------\n");
+ }
+ err = snd_pcm_hw_params_set_access(alsa_handle, params,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (err < 0) {
+ fprintf(stderr, "Access type not available");
+ prg_exit(EXIT_FAILURE);
+ }
+ err = snd_pcm_hw_params_set_format(alsa_handle, params, hwparams.format);
+ if (err < 0) {
+ fprintf(stderr, "Sample format non available");
+ snd_pcm_format_t format;
+
+ fprintf(stderr, "Available formats:\n");
+ for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) {
+ if (snd_pcm_hw_params_test_format(alsa_handle, params, format) == 0)
+ fprintf(stderr, "- %s\n", snd_pcm_format_name(format));
+ }
+ prg_exit(EXIT_FAILURE);
+ }
+ err = snd_pcm_hw_params_set_channels(alsa_handle, params, hwparams.channels);
+ if (err < 0) {
+ fprintf(stderr, "Channels count non available");
+ prg_exit(EXIT_FAILURE);
+ }
+
+#if 0
+ err = snd_pcm_hw_params_set_periods_min(alsa_handle, params, 2);
+ assert(err >= 0);
+#endif
+ rate = hwparams.rate;
+ err = snd_pcm_hw_params_set_rate_near(alsa_handle, params, &hwparams.rate, 0);
+ assert(err >= 0);
+ if ((float)rate * 1.05 < hwparams.rate || (float)rate * 0.95 > hwparams.rate) {
+ char plugex[64];
+ const char *pcmname = snd_pcm_name(alsa_handle);
+ fprintf(stderr, "Warning: rate is not accurate (requested = %iHz, got = %iHz)\n", rate, hwparams.rate);
+ if (! pcmname || strchr(snd_pcm_name(alsa_handle), ':')) {
+ *plugex = 0;
+ }
+ else {
+ snprintf(plugex, sizeof(plugex), "(-Dplug:%s)",
+ snd_pcm_name(alsa_handle));
+ }
+ fprintf(stderr, " please, try the plug plugin %s\n",
+ plugex);
+ }
+ rate = hwparams.rate;
+ if (buffer_time == 0 && buffer_frames == 0) {
+ err = snd_pcm_hw_params_get_buffer_time_max(params,
+ &buffer_time, 0);
+ assert(err >= 0);
+ if (buffer_time > 500000)
+ buffer_time = 500000;
+ }
+ if (period_time == 0 && period_frames == 0) {
+ if (buffer_time > 0)
+ period_time = buffer_time / 4;
+ else
+ period_frames = buffer_frames / 4;
+ }
+ if (period_time > 0)
+ err = snd_pcm_hw_params_set_period_time_near(alsa_handle, params,
+ &period_time, 0);
+ else
+ err = snd_pcm_hw_params_set_period_size_near(alsa_handle, params,
+ &period_frames, 0);
+ assert(err >= 0);
+ if (buffer_time > 0) {
+ err = snd_pcm_hw_params_set_buffer_time_near(alsa_handle, params,
+ &buffer_time, 0);
+ } else {
+ err = snd_pcm_hw_params_set_buffer_size_near(alsa_handle, params,
+ &buffer_frames);
+ }
+ assert(err >= 0);
+ monotonic = snd_pcm_hw_params_is_monotonic(params);
+ can_pause = snd_pcm_hw_params_can_pause(params);
+ err = snd_pcm_hw_params(alsa_handle, params);
+ if (err < 0) {
+ fprintf(stderr, "Unable to install hw params:");
+ snd_pcm_hw_params_dump(params, log);
+ prg_exit(EXIT_FAILURE);
+ }
+ snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
+ snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
+ if (chunk_size == buffer_size) {
+ fprintf(stderr, "Can't use period equal to buffer size (%lu == %lu)",
+ chunk_size, buffer_size);
+ prg_exit(EXIT_FAILURE);
+ }
+ snd_pcm_sw_params_current(alsa_handle, swparams);
+ if (avail_min < 0)
+ n = chunk_size;
+ else
+ n = (double) rate * avail_min / 1000000;
+ err = snd_pcm_sw_params_set_avail_min(alsa_handle, swparams, n);
+
+ /* round up to closest transfer boundary */
+ n = buffer_size;
+ if (start_delay <= 0) {
+ start_threshold = n + (double) rate * start_delay / 1000000;
+ } else
+ start_threshold = (double) rate * start_delay / 1000000;
+ if (start_threshold < 1)
+ start_threshold = 1;
+ if (start_threshold > n)
+ start_threshold = n;
+ err = snd_pcm_sw_params_set_start_threshold(alsa_handle, swparams, start_threshold);
+ assert(err >= 0);
+ if (stop_delay <= 0)
+ stop_threshold = buffer_size + (double) rate * stop_delay / 1000000;
+ else
+ stop_threshold = (double) rate * stop_delay / 1000000;
+ err = snd_pcm_sw_params_set_stop_threshold(alsa_handle, swparams, stop_threshold);
+ assert(err >= 0);
+
+ if (snd_pcm_sw_params(alsa_handle, swparams) < 0) {
+ fprintf(stderr, "unable to install sw params:");
+ snd_pcm_sw_params_dump(swparams, log);
+ prg_exit(EXIT_FAILURE);
+ }
+
+ if (setup_chmap())
+ prg_exit(EXIT_FAILURE);
+
+ if (verbose)
+ snd_pcm_dump(alsa_handle, log);
+
+ bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
+ bits_per_frame = bits_per_sample * hwparams.channels;
+ chunk_bytes = chunk_size * bits_per_frame / 8;
+ audiobuf = realloc(audiobuf, chunk_bytes);
+ if (audiobuf == NULL) {
+ fprintf(stderr, "not enough memory");
+ prg_exit(EXIT_FAILURE);
+ }
+ // fprintf(stderr, "real chunk_size = %i, frags = %i, total = %i\n", chunk_size, setup.buf.block.frags, setup.buf.block.frags * chunk_size);
+
+ /* stereo VU-meter isn't always available... */
+ if (vumeter == VUMETER_STEREO) {
+ if (hwparams.channels != 2 || !interleaved || verbose > 2)
+ vumeter = VUMETER_MONO;
+ }
+
+ /* show mmap buffer arragment */
+ if (mmap_flag && verbose) {
+ const snd_pcm_channel_area_t *areas;
+ snd_pcm_uframes_t offset, size = chunk_size;
+ int i;
+ err = snd_pcm_mmap_begin(alsa_handle, &areas, &offset, &size);
+ if (err < 0) {
+ fprintf(stderr, "snd_pcm_mmap_begin problem: %s", snd_strerror(err));
+ prg_exit(EXIT_FAILURE);
+ }
+ for (i = 0; i < hwparams.channels; i++)
+ fprintf(stderr, "mmap_area[%i] = %p,%u,%u (%u)\n", i, areas[i].addr, areas[i].first, areas[i].step, snd_pcm_format_physical_width(hwparams.format));
+ /* not required, but for sure */
+ snd_pcm_mmap_commit(alsa_handle, offset, 0);
+ }
+
+ buffer_frames = buffer_size; /* for position test */
+}
+
+#define no_argument 0
+#define required_argument 1
+#define optional_argument 2
+
+int main(int argc, char *argv[]) {
+ int subchannel_index = 8; //64kbps subchannel
+ int ch=0;
+ int err;
+ const char *alsa_device = "default";
+ const char *outuri = NULL;
+ int sample_rate=48000, channels=2;
+ const int bits_per_sample = 16;
+ uint8_t* input_buf;
+ int16_t* convert_buf;
+ void *rs_handler = NULL;
+ int aot = AOT_DABPLUS_AAC_LC;
+ int afterburner = 0;
+ HANDLE_AACENCODER handle;
+ CHANNEL_MODE mode;
+ AACENC_InfoStruct info = { 0 };
+
+ void *zmq_context = zmq_ctx_new();
+ void *zmq_sock = NULL;
+
+ const struct option longopts[] = {
+ {"bitrate", required_argument, 0, 'b'},
+ {"output", required_argument, 0, 'o'},
+ {"device", required_argument, 0, 'd'},
+ {"rate", required_argument, 0, 'r'},
+ {"channels", required_argument, 0, 'c'},
+ //{"lp", no_argument, 0, 'l'},
+ {"afterburner", no_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0,0,0,0},
+ };
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "lhab:c:o:r:d:", longopts, &index);
+ switch (ch) {
+ case 'd':
+ alsa_device = optarg;
+ break;
+ case 'a':
+ afterburner = 1;
+ break;
+ case 'b':
+ subchannel_index = atoi(optarg) / 8;
+ break;
+ case 'c':
+ channels = atoi(optarg);
+ break;
+ case 'r':
+ sample_rate = atoi(optarg);
+ break;
+ case 'o':
+ outuri = optarg;
+ break;
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ return 1;
+ }
+ }
+
+ if(subchannel_index < 1 || subchannel_index > 24) {
+ fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n", subchannel_index);
+ return 1;
+ }
+
+ if (outuri) {
+ zmq_sock = zmq_socket(zmq_context, ZMQ_PUB);
+ if (zmq_sock == NULL) {
+ fprintf(stderr, "Error occurred during zmq_socket: %s\n", zmq_strerror(errno));
+ return 2;
+ }
+ if (zmq_connect(zmq_sock, outuri) != 0) {
+ fprintf(stderr, "Error occurred during zmq_connect: %s\n", zmq_strerror(errno));
+ return 2;
+ }
+ } else {
+ fprintf(stderr, "Output URI not defined\n");
+ return 1;
+ }
+
+
+ const int open_mode = 0; //|= SND_PCM_NONBLOCK;
+ const snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE;
+ const int nonblock = 0;
+ snd_pcm_info_t *alsa_info;
+
+ err = snd_pcm_open(&alsa_handle, alsa_device, stream, open_mode);
+ if (err < 0) {
+ fprintf(stderr, "audio open error: %s", snd_strerror(err));
+ return 1;
+ }
+
+ if ((err = snd_pcm_info(alsa_handle, alsa_info)) < 0) {
+ fprintf(stderr, "info error: %s", snd_strerror(err));
+ prg_exit(1);
+ }
+
+ if (nonblock) {
+ err = snd_pcm_nonblock(alsa_handle, 1);
+ if (err < 0) {
+ fprintf(stderr, "nonblock setting error: %s", snd_strerror(err));
+ prg_exit(1);
+ }
+ }
+
+ signal(SIGINT, signal_handler);
+ signal(SIGTERM, signal_handler);
+ signal(SIGABRT, signal_handler);
+
+ switch (channels) {
+ case 1: mode = MODE_1; break;
+ case 2: mode = MODE_2; break;
+ default:
+ fprintf(stderr, "Unsupported channels number %d\n", channels);
+ prg_exit(1);
+ }
+
+
+ if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
+ fprintf(stderr, "Unable to open encoder\n");
+ prg_exit(1);
+ }
+
+
+ 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");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\n");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the channel mode\n");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the wav channel order\n");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\n");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the RAW transmux\n");
+ prg_exit(1);
+ }
+
+ /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate mode\n");
+ prg_exit(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");
+ prg_exit(1);
+ }
+ if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the afterburner mode\n");
+ prg_exit(1);
+ }
+ if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+ fprintf(stderr, "Unable to initialize the encoder\n");
+ prg_exit(1);
+ }
+ if (aacEncInfo(handle, &info) != AACENC_OK) {
+ fprintf(stderr, "Unable to get the encoder info\n");
+ prg_exit(1);
+ }
+
+ fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength);
+
+ int input_size = channels*2*info.frameLength;
+ input_buf = (uint8_t*) malloc(input_size);
+ convert_buf = (int16_t*) malloc(input_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");
+ prg_exit(1);
+ }
+
+ int loops = 0;
+ int outbuf_size = subchannel_index*120;
+ uint8_t outbuf[20480];
+
+ if(outbuf_size % 5 != 0) {
+ fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
+ }
+
+ fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+ //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5;
+ fprintf(stderr, "outbuf_size: %d\n", outbuf_size);
+
+ int frame=0;
+ int send_error_count = 0;
+ while (1) {
+ memset(outbuf, 0x00, outbuf_size);
+
+ AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+ AACENC_InArgs in_args = { 0 };
+ AACENC_OutArgs out_args = { 0 };
+ int in_identifier = IN_AUDIO_DATA;
+ int in_size, in_elem_size;
+ int out_identifier = OUT_BITSTREAM_DATA;
+ int out_size, out_elem_size;
+ int read=0, i;
+ int send_error;
+ void *in_ptr, *out_ptr;
+ AACENC_ERROR err;
+
+ // raw input
+ if(fread(input_buf, input_size, 1, in_fh) == 1) {
+ read = input_size;
+ } else {
+ fprintf(stderr, "Unable to read from input!\n");
+ break;
+ }
+
+ for (i = 0; i < read/2; i++) {
+ const uint8_t* in = &input_buf[2*i];
+ convert_buf[i] = in[0] | (in[1] << 8);
+ }
+
+ if (read <= 0) {
+ in_args.numInSamples = -1;
+ } else {
+ in_ptr = convert_buf;
+ in_size = read;
+ in_elem_size = 2;
+
+ in_args.numInSamples = read/2;
+ in_buf.numBufs = 1;
+ in_buf.bufs = &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;
+
+ if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
+ if (err == AACENC_ENCODE_EOF)
+ break;
+ fprintf(stderr, "Encoding failed\n");
+ prg_exit(1);
+ }
+ if (out_args.numOutBytes == 0)
+ continue;
+#if 0
+ unsigned char au_start[6];
+ unsigned char* sfbuf = outbuf;
+ au_start[0] = 6;
+ au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4);
+ au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5);
+ fprintf (stderr, "au_start[0] = %d\n", au_start[0]);
+ fprintf (stderr, "au_start[1] = %d\n", au_start[1]);
+ fprintf (stderr, "au_start[2] = %d\n", au_start[2]);
+#endif
+
+ 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);
+ }
+ }
+
+ send_error = zmq_send(zmq_sock, outbuf, outbuf_size, ZMQ_DONTWAIT);
+ if (send_error < 0) {
+ fprintf(stderr, "ZeroMQ send failed! %s\n", zmq_strerror(errno));
+ send_error_count ++;
+ }
+
+ if (send_error_count > 10)
+ {
+ fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n");
+ break;
+ }
+ //fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh);
+ //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size);
+ if(out_args.numOutBytes + row*10 == outbuf_size)
+ fprintf(stderr, ".");
+
+// if(frame > 10)
+// break;
+ frame++;
+ }
+ free(input_buf);
+ free(convert_buf);
+
+ zmq_close(zmq_sock);
+ free_rs_char(rs_handler);
+
+ aacEncClose(&handle);
+
+ zmq_ctx_term(zmq_context);
+ prg_exit(0);
+}
+