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authorMatthias P. Braendli <matthias.braendli@mpb.li>2014-03-09 17:00:08 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2014-03-09 17:00:08 +0100
commite7ff854765251735cf8d3631af51e6b47809bc98 (patch)
treed597fd224f7d450d7d109b86357056417e101179 /src/dabplus-enc-file-zmq.c
parent1a0fc3ef6348f0c85a2bdbc97ebf5422217dabdd (diff)
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tidy dabplus-enc-file-zmq
Diffstat (limited to 'src/dabplus-enc-file-zmq.c')
-rw-r--r--src/dabplus-enc-file-zmq.c500
1 files changed, 500 insertions, 0 deletions
diff --git a/src/dabplus-enc-file-zmq.c b/src/dabplus-enc-file-zmq.c
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+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ * Copyright (C) 2013,2014 Matthias P. Braendli
+ * Copyright (C) 2014 CSP Innovazione nelle ICT s.c.a r.l.
+ * http://rd.csp.it/
+ *
+ * http://opendigitalradio.org
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation, either version 3 of the
+ * License, or (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program If not, see <http://www.gnu.org/licenses/>.
+ * -------------------------------------------------------------------
+ */
+
+#include <stdio.h>
+#include <stdint.h>
+#include <string.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <getopt.h>
+#include <zmq.h>
+#include <assert.h>
+#include "libAACenc/include/aacenc_lib.h"
+#include "wavreader.h"
+
+#include <fec.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <errno.h>
+
+#include "contrib/lib_crc.h"
+
+#define MIN(a,b) (((a)<(b))?(a):(b))
+#define MAX(a,b) (((a)>(b))?(a):(b))
+
+void usage(const char* name) {
+ fprintf(stderr,
+ "%s is a HE-AACv2 encoder for DAB+ based on fdk-aac-dabplus\n"
+ "that can encode from a file or pipe source, and encode\n"
+ "to a ZeroMQ output for ODR-DabMux.\n"
+ "\n"
+ "It includes PAD (DLS and MOT Slideshow) support by http://rd.csp.it\n"
+ "to be used with mot-encoder\n"
+ "\n"
+ " http://opendigitalradio.org\n"
+ "\nUsage:\n"
+ "%s [OPTION...]\n", name, name);
+
+ fprintf(stderr,
+ " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+ //" -d, --data=FILENAME Set data filename.\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"
+ " -p, --pad=BYTES Set PAD size in bytes.\n"
+ " -f, --format={ wav, raw } Set input file format (default: wav).\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"
+ "\n"
+ "Only the tcp:// zeromq transport has been tested until now.\n"
+
+ );
+
+}
+
+
+#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;
+ const char *infile = NULL;
+ const char *outuri = NULL;
+ FILE *in_fh;
+ void *wav;
+ int wav_format, bits_per_sample, sample_rate=48000, channels=2;
+ uint8_t* input_buf;
+ int16_t* convert_buf;
+ void *rs_handler = NULL;
+ int aot = AOT_DABPLUS_AAC_LC;
+ int afterburner = 0, raw_input=0;
+ HANDLE_AACENCODER handle;
+ CHANNEL_MODE mode;
+ AACENC_InfoStruct info = { 0 };
+
+ int pad_fd;
+ unsigned char pad_buf[128];
+ int padlen;
+
+ void *zmq_context = zmq_ctx_new();
+ void *zmq_sock = NULL;
+
+ const struct option longopts[] = {
+ {"bitrate", required_argument, 0, 'b'},
+ {"input", required_argument, 0, 'i'},
+ {"output", required_argument, 0, 'o'},
+ {"format", required_argument, 0, 'f'},
+ {"rate", required_argument, 0, 'r'},
+ {"channels", required_argument, 0, 'c'},
+ {"pad", required_argument, 0, 'p'},
+ {"afterburner", no_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0,0,0,0},
+ };
+
+ if (argc == 1) {
+ usage(argv[0]);
+ return 0;
+ }
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "tlhab:c:i:o:r:f:p:", longopts, &index);
+ switch (ch) {
+ case 'f':
+ if(strcmp(optarg, "raw")==0) {
+ raw_input = 1;
+ } else if(strcmp(optarg, "wav")!=0)
+ usage(argv[0]);
+ 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 'i':
+ infile = optarg;
+ break;
+ case 'o':
+ outuri = optarg;
+ break;
+ case 'p':
+ padlen = atoi(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(padlen != 0) {
+ int flags;
+ if (mkfifo("/tmp/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("/tmp/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;
+ }
+ }
+
+ if(raw_input) {
+ if(infile && strcmp(infile, "-")) {
+ in_fh = fopen(infile, "rb");
+ if(!in_fh) {
+ fprintf(stderr, "Can't open input file!\n");
+ return 1;
+ }
+ } else {
+ in_fh = stdin;
+ }
+ } else {
+ wav = wav_read_open(infile);
+ if (!wav) {
+ fprintf(stderr, "Unable to open wav file %s\n", infile);
+ return 1;
+ }
+ if (!wav_get_header(wav, &wav_format, &channels, &sample_rate, &bits_per_sample, NULL)) {
+ fprintf(stderr, "Bad wav file %s\n", infile);
+ return 1;
+ }
+ if (wav_format != 1) {
+ fprintf(stderr, "Unsupported WAV format %d\n", wav_format);
+ return 1;
+ }
+ if (bits_per_sample != 16) {
+ fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
+ return 1;
+ }
+ if (channels > 2) {
+ fprintf(stderr, "Unsupported WAV channels %d\n", channels);
+ 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;
+ }
+
+
+ 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;
+ }
+
+
+ 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 samplerate\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 (aacEncoder_SetParam(handle, AACENC_ANCILLARY_BITRATE, 0) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the ancillary bitrate\n");
+ return 1;
+ }
+ if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
+ fprintf(stderr, "Unable to initialize the encoder\n");
+ return 1;
+ }
+ if (aacEncInfo(handle, &info) != AACENC_OK) {
+ fprintf(stderr, "Unable to get the encoder info\n");
+ return 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");
+ return 0;
+ }
+
+ 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, IN_ANCILLRY_DATA};
+ int in_size[2], in_elem_size[2];
+ int out_identifier = OUT_BITSTREAM_DATA;
+ int out_size, out_elem_size;
+ int pcmread=0, i, ret;
+ int send_error;
+ void *in_ptr[2], *out_ptr;
+ AACENC_ERROR err;
+
+ // Read data from the PAD fifo
+ 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;
+ }
+ else {
+ // Some other error occurred during read.
+ fprintf(stderr, "Unable to read from PAD!\n");
+ break;
+ }
+
+ if(raw_input) {
+ if(fread(input_buf, input_size, 1, in_fh) == 1) {
+ pcmread = input_size;
+ } else {
+ fprintf(stderr, "Unable to read from input!\n");
+ break;
+ }
+ } else {
+ pcmread = wav_read_data(wav, input_buf, input_size);
+ }
+
+ for (i = 0; i < pcmread/2; i++) {
+ const uint8_t* in = &input_buf[2*i];
+ convert_buf[i] = in[0] | (in[1] << 8);
+ }
+
+ if (pcmread <= 0) {
+ in_args.numInSamples = -1;
+ } else {
+ in_ptr[0] = convert_buf;
+ in_ptr[1] = pad_buf;
+ in_size[0] = pcmread;
+ in_size[1] = padlen;
+
+ in_elem_size[0] = 2;
+ in_elem_size[1] = sizeof(UCHAR);
+
+ in_args.numInSamples = pcmread/2;
+ in_args.numAncBytes = padlen;
+
+ //in_buf.numBufs = 2; // Samples + Data
+
+ 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;
+
+ 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");
+ return 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);
+ if(raw_input) {
+ fclose(in_fh);
+ } else {
+ wav_read_close(wav);
+ }
+ zmq_close(zmq_sock);
+ free_rs_char(rs_handler);
+
+ aacEncClose(&handle);
+
+ zmq_ctx_term(zmq_context);
+ return 0;
+}
+