add non-working alsa-dabplus-zmq.c

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);
+}
+