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diff --git a/src/dabplus-enc-alsa-zmq.cpp b/src/dabplus-enc-alsa-zmq.cpp
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+/* ------------------------------------------------------------------
+ * 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 "SampleQueue.h"
+#include "zmq.hpp"
+
+#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 "libAACenc/include/aacenc_lib.h"
+
+extern "C" {
+#include <fec.h>
+}
+
+using namespace std;
+
+void usage(const char* name) {
+ fprintf(stderr,
+ "dabplus-enc-alsa-zmq %s is a HE-AACv2 encoder for DAB+\n"
+ "based on fdk-aac-dabplus that can read from a ALSA 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 [OPTION...]\n",
+#if defined(GITVERSION)
+ GITVERSION
+#else
+ PACKAGE_VERSION
+#endif
+ , name);
+ fprintf(stderr,
+ " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n"
+ " -D, --drift-comp Enable ALSA sound card drift compensation.\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"
+ " -P, --pad-fifo=FILENAME Set PAD data input fifo name (default: /tmp/pad.fifo).\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"
+ //" -v, --verbose=LEVEL Set verbosity level.\n"
+ //" -V, --version Print version and exit.\n"
+ "\n"
+ "Only the tcp:// zeromq transport has been tested until now.\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
+
+int main(int argc, char *argv[]) {
+ int subchannel_index = 8; //64kbps subchannel
+ int ch=0;
+ const char *alsa_device = "default";
+ 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 };
+
+ char* pad_fifo = "/tmp/pad.fifo";
+ int pad_fd;
+ unsigned char pad_buf[128];
+ int padlen;
+
+ 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'},
+ {"pad", required_argument, 0, 'p'},
+ {"pad-fifo", required_argument, 0, 'P'},
+ {"drift-comp", no_argument, 0, 'D'},
+ {"afterburner", no_argument, 0, 'a'},
+ {"help", no_argument, 0, 'h'},
+ {0,0,0,0},
+ };
+
+ if (argc < 2) {
+ usage(argv[0]);
+ return 1;
+ }
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "hab:c:o:r:d:Dp:P:", longopts, &index);
+ switch (ch) {
+ case 'd':
+ alsa_device = optarg;
+ break;
+ case 'a':
+ afterburner = true;
+ 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 'D':
+ drift_compensation = true;
+ break;
+ case 'p':
+ padlen = atoi(optarg);
+ break;
+ case 'P':
+ pad_fifo = 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;
+ }
+
+ fprintf(stderr, "Setting up ZeroMQ socket\n");
+ if (!outuri) {
+ fprintf(stderr, "ZeroMQ 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;
+ }
+ }
+
+ zmq::context_t zmq_ctx;
+ zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
+ zmq_sock.connect(outuri);
+
+ 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;
+ }
+
+ // We'll use either of the two possible alsa inputs.
+ AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
+ AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+
+ 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 outbuf[20480];
+
+ 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 wait_time = 120000000ul / 2;
+
+ 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)
+ fprintf(stderr, "p");
+ }
+ else {
+ // Some other error occurred during read.
+ fprintf(stderr, "Unable to read from PAD!\n");
+ break;
+ }
+
+ // -------------- Read Data
+ memset(outbuf, 0x00, outbuf_size);
+
+ size_t read;
+ if (drift_compensation) {
+ size_t overruns;
+ read = queue.pop(input_buf, input_size, &overruns); // returns bytes
+
+ if (read != input_size) {
+ fprintf(stderr, "U");
+ }
+
+ if (overruns) {
+ fprintf(stderr, "O%zu", overruns);
+ }
+ }
+ else {
+ read = alsa_in_direct.read(input_buf, input_size);
+ if (read != input_size) {
+ fprintf(stderr, "Short alsa read !\n");
+ }
+ }
+
+ // -------------- 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)
+ break;
+ fprintf(stderr, "Encoding failed\n");
+ break;
+ }
+ calls++;
+
+ /* Check if the encoder has generated output data */
+ if (out_args.numOutBytes != 0)
+ {
+ // Our timing code depends on this
+ assert (calls == 2);
+ 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_sock.send(outbuf, outbuf_size, 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 (out_args.numOutBytes + row*10 == outbuf_size)
+ fprintf(stderr, ".");
+ }
+ }
+
+ zmq_sock.close();
+ free_rs_char(rs_handler);
+
+ aacEncClose(&encoder);
+
+}
+
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