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+/* ------------------------------------------------------------------
+ * Copyright (C) 2011 Martin Storsjo
+ * Copyright (C) 2016 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.
+ * -------------------------------------------------------------------
+ */
+
+/*! \mainpage Introduction
+ * The ODR-mmbTools ODR-AudioEnc Audio encoder can encode audio for
+ * ODR-DabMux, both DAB and DAB+. The DAB encoder is based on toolame. The
+ * DAB+ encoder requires a the Fraunhofer FDK AAC library, with the
+ * necessary patches for 960-transform to do DAB+ broadcast encoding.
+ *
+ * This document describes some internals of the encoder, and is intended
+ * to help developers understand and improve the software package.
+ *
+ * User documentation is available in the README and in the ODR-mmbTools
+ * Guide, available on the www.opendigitalradio.org website.
+ *
+ * The readme for the whole package is \ref md_README
+ *
+ * Interesting starting points for the encoder
+ * - \ref odr-audioenc.cpp Main encoder file
+ * - \ref VLCInput.h VLC Input
+ * - \ref AlsaInput.h Alsa Input
+ * - \ref JackInput.h JACK Input
+ * - \ref SampleQueue.h
+ * - \ref charset.h Charset conversion
+ * - \ref toolame.h libtolame API
+ * - \ref AudioLevel
+ * - \ref DataInput
+ * - \ref SilenceDetection
+ *
+ * For the mot-encoder:
+ * - \ref mot-encoder.cpp
+ *
+ *
+ * \file odr-audioenc.cpp
+ * \brief The main file for the audio encoder
+ */
+
+#include "config.h"
+#include "AlsaInput.h"
+#include "FileInput.h"
+#include "JackInput.h"
+#include "VLCInput.h"
+#include "SampleQueue.h"
+#include "zmq.hpp"
+#include "common.h"
+
+extern "C" {
+#include "encryption.h"
+#include "utils.h"
+#include "wavreader.h"
+}
+
+#include <vector>
+#include <deque>
+#include <chrono>
+#include <thread>
+#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 <fcntl.h>
+
+#include "fdk-aac/aacenc_lib.h"
+
+extern "C" {
+#include <fec.h>
+#include "libtoolame-dab/toolame.h"
+}
+
+
+
+//! Enumeration of encoders we can use
+enum class encoder_selection_t {
+ fdk_dabplus,
+ toolame_dab
+};
+
+using namespace std;
+
+void usage(const char* name) {
+ fprintf(stderr,
+ "ODR-AudioEnc %s is an audio encoder for both DAB and DAB+.\n"
+ "The encoder can read from JACK, ALSA or\n"
+ "a file source and encode to a ZeroMQ output for ODR-DabMux.\n"
+ "(Experimental!)It can also use libvlc as an input.\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. It can be used for both ALSA and VLC inputs.\n"
+ "\n"
+ "When this option is enabled, you will see U and O printed in the\n"
+ "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"
+ "\nUsage:\n"
+ "%s [INPUT SELECTION] [OPTION...]\n",
+#if defined(GITVERSION)
+ GITVERSION
+#else
+ PACKAGE_VERSION
+#endif
+ , name);
+ fprintf(stderr,
+ " For the alsa input:\n"
+#if HAVE_ALSA
+ " -d, --device=alsa_device Set ALSA input device (default: \"default\").\n"
+#else
+ " The Alsa input was disabled at compile time\n"
+#endif
+ " For the file input:\n"
+ " -i, --input=FILENAME Input filename (default: stdin).\n"
+ " -f, --format={ wav, raw } Set input file format (default: wav).\n"
+ " --fifo-silence Input file is fifo and encoder generates silence when fifo is empty. Ignore EOF.\n"
+ " For the JACK input:\n"
+#if HAVE_JACK
+ " -j, --jack=name Enable JACK input, and define our name\n"
+#else
+ " The JACK input was disabled at compile-time\n"
+#endif
+ " For the VLC input:\n"
+#if HAVE_VLC
+ " -v, --vlc-uri=uri Enable VLC input and use the URI given as source\n"
+ " -C, --vlc-cache=ms Specify VLC network cache length.\n"
+ " -g, --vlc-gain=db Enable VLC audio compressor, with given compressor-makeup value.\n"
+ " Use this as a workaround to correct the gain for streams that are\n"
+ " much too loud.\n"
+ " -V Increase the VLC verbosity by one (can be given \n"
+ " multiple times)\n"
+ " -L OPTION Give an additional options to VLC (can be given\n"
+ " multiple times)\n"
+ " -w, --write-icy-text=filename Write the ICY Text into the file, so that mot-encoder can read it.\n"
+ " -W, --write-icy-text-dl-plus When writing the ICY Text into the file, add DL Plus information.\n"
+#else
+ " The VLC input was disabled at compile-time\n"
+#endif
+ " Drift compensation\n"
+ " -D, --drift-comp Enable ALSA/VLC sound card drift compensation.\n"
+ " Encoder parameters:\n"
+ " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be a multiple of 8.\n"
+ " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n"
+ " -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n"
+ " DAB specific options\n"
+ " -a, --dab Encode in DAB and not in DAB+.\n"
+ " --dabmode=MODE Channel mode: s/d/j/m\n"
+ " (default: j if stereo, m if mono).\n"
+ " --dabpsy=PSY Psychoacoustic model 0/1/2/3\n"
+ " (default: 1).\n"
+ " DAB+ specific options\n"
+ " -A, --no-afterburner Disable AAC encoder quality increaser.\n"
+ " --aaclc Force the usage of AAC-LC (no SBR, no PS)\n"
+ " --sbr Force the usage of SBR\n"
+ " --ps Force the usage of PS\n"
+ " Output and pad parameters:\n"
+ " -o, --output=URI Output ZMQ uri. (e.g. 'tcp://localhost:9000')\n"
+ " -or- Output file uri. (e.g. 'file.dabp')\n"
+ " -or- a single dash '-' to denote stdout\n"
+ " If more than one ZMQ output is given, the socket\n"
+ " will be connected to all listed endpoints.\n"
+ " -k, --secret-key=FILE Enable ZMQ encryption with the given secret key.\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"
+ " -l, --level Show peak audio level indication.\n"
+ " -s, --silence=TIMEOUT Abort encoding after TIMEOUT seconds of silence.\n"
+ "\n"
+ "Only the tcp:// zeromq transport has been tested until now,\n"
+ " but epgm:// and pgm:// are also accepted\n"
+ );
+
+}
+
+/*! Setup the FDK AAC encoder
+ *
+ * \return 0 on success
+ */
+int prepare_aac_encoder(
+ HANDLE_AACENCODER *encoder,
+ int subchannel_index,
+ int channels,
+ int sample_rate,
+ int afterburner,
+ int *aot)
+{
+ 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(encoder, 0x01|0x02|0x04, channels) != AACENC_OK) {
+ fprintf(stderr, "Unable to open encoder\n");
+ return 1;
+ }
+
+ if (*aot == AOT_NONE) {
+
+ if(channels == 2 && subchannel_index <= 6) {
+ *aot = AOT_DABPLUS_PS;
+ }
+ else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) {
+ *aot = AOT_DABPLUS_SBR;
+ }
+ else {
+ *aot = AOT_DABPLUS_AAC_LC;
+ }
+ }
+
+ 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(*encoder, AACENC_AOT, *aot) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the AOT\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the sample rate\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_CHANNELMODE, mode) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the channel mode\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_CHANNELORDER, 1) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the wav channel order\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the granule length\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the RAW transmux\n");
+ return 1;
+ }
+
+ /*if (aacEncoder_SetParam(*encoder, AACENC_BITRATEMODE, 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(*encoder, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the bitrate\n");
+ return 1;
+ }
+ if (aacEncoder_SetParam(*encoder, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
+ fprintf(stderr, "Unable to set the afterburner mode\n");
+ return 1;
+ }
+ if (!afterburner) {
+ fprintf(stderr, "Warning: Afterburned disabled!\n");
+ }
+ if (aacEncEncode(*encoder, NULL, NULL, NULL, NULL) != AACENC_OK) {
+ fprintf(stderr, "Unable to initialize the encoder\n");
+ return 1;
+ }
+ return 0;
+}
+
+chrono::steady_clock::time_point timepoint_last_compensation;
+
+/*! Wait the proper amount of time to throttle down to nominal encoding
+ * rate, if drift compensation is enabled.
+ */
+void drift_compensation_delay(int sample_rate, int channels, size_t bytes)
+{
+ const size_t bytes_per_second = sample_rate * BYTES_PER_SAMPLE * channels;
+
+ size_t bytes_compensate = bytes;
+ const auto wait_time = std::chrono::milliseconds(1000ul * bytes_compensate / bytes_per_second);
+ assert(1000ul * bytes_compensate % bytes_per_second == 0);
+
+ const auto curTime = std::chrono::steady_clock::now();
+
+ const auto diff = curTime - timepoint_last_compensation;
+
+ if (diff < wait_time) {
+ auto waiting = wait_time - diff;
+ std::this_thread::sleep_for(waiting);
+ }
+
+ timepoint_last_compensation += wait_time;
+}
+
+#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 bitrate = 0; // 0 is default
+ int ch=0;
+
+ encoder_selection_t selected_encoder = encoder_selection_t::fdk_dabplus;
+
+ // For the ALSA input
+ const char *alsa_device = NULL;
+
+ // For the file input
+ const char *infile = NULL;
+ int raw_input = 0;
+
+ // For the VLC input
+ std::string vlc_uri = "";
+ std::string vlc_icytext_file = "";
+ bool vlc_icytext_dlplus = false;
+ std::string vlc_gain = "";
+ std::string vlc_cache = "";
+ std::vector<std::string> vlc_additional_opts;
+ unsigned verbosity = 0;
+
+ // For the file output
+ FILE *out_fh = NULL;
+
+ const char *jack_name = NULL;
+
+ std::vector<std::string> output_uris;
+
+ int sample_rate=48000, channels=2;
+ void *rs_handler = NULL;
+ bool afterburner = true;
+ bool inFifoSilence = false;
+ bool drift_compensation = false;
+ AACENC_InfoStruct info = { 0 };
+ int aot = AOT_NONE;
+
+ char dab_channel_mode = '\0';
+ int dab_psy_model = 1;
+ std::deque<uint8_t> toolame_output_buffer;
+
+ /* Keep track of peaks */
+ int peak_left = 0;
+ int peak_right = 0;
+
+ /* On silence, die after the silence_timeout expires */
+ bool die_on_silence = false;
+ int silence_timeout = 0;
+ int measured_silence_ms = 0;
+
+ /* For MOT Slideshow and DLS insertion */
+ const char* pad_fifo = "/tmp/pad.fifo";
+ int pad_fd;
+ int padlen = 0;
+
+ /* 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'},
+ {"dabmode", required_argument, 0, 4 },
+ {"dabpsy", required_argument, 0, 5 },
+ {"device", required_argument, 0, 'd'},
+ {"format", required_argument, 0, 'f'},
+ {"input", required_argument, 0, 'i'},
+ {"jack", required_argument, 0, 'j'},
+ {"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'},
+ {"silence", required_argument, 0, 's'},
+ {"vlc-cache", required_argument, 0, 'C'},
+ {"vlc-gain", required_argument, 0, 'g'},
+ {"vlc-uri", required_argument, 0, 'v'},
+ {"vlc-opt", required_argument, 0, 'L'},
+ {"write-icy-text", required_argument, 0, 'w'},
+ {"write-icy-text-dl-plus", no_argument, 0, 'W'},
+ {"aaclc", no_argument, 0, 0 },
+ {"dab", no_argument, 0, 'a'},
+ {"drift-comp", no_argument, 0, 'D'},
+ {"fifo-silence", no_argument, 0, 3 },
+ {"help", no_argument, 0, 'h'},
+ {"level", no_argument, 0, 'l'},
+ {"no-afterburner", no_argument, 0, 'A'},
+ {"ps", no_argument, 0, 2 },
+ {"sbr", no_argument, 0, 1 },
+ {"verbosity", no_argument, 0, 'V'},
+ {0, 0, 0, 0},
+ };
+
+ fprintf(stderr,
+ "Welcome to %s %s, compiled at %s, %s",
+ PACKAGE_NAME,
+#if defined(GITVERSION)
+ GITVERSION,
+#else
+ PACKAGE_VERSION,
+#endif
+ __DATE__, __TIME__);
+ fprintf(stderr, "\n");
+ fprintf(stderr, " http://opendigitalradio.org\n\n");
+
+
+ if (argc < 2) {
+ usage(argv[0]);
+ return 1;
+ }
+
+ int index;
+ while(ch != -1) {
+ ch = getopt_long(argc, argv, "aAhDlVb:c:f:i:j:k:L:o:r:d:p:P:s:v:w:Wg:C:", longopts, &index);
+ switch (ch) {
+ case 0: // AAC-LC
+ aot = AOT_DABPLUS_AAC_LC;
+ break;
+ case 1: // SBR
+ aot = AOT_DABPLUS_SBR;
+ break;
+ case 2: // PS
+ aot = AOT_DABPLUS_PS;
+ break;
+ case 3: // FIFO SILENCE
+ case 4: // DAB channel mode
+ dab_channel_mode = optarg[0];
+ break;
+ case 5: // DAB psy model
+ dab_psy_model = atoi(optarg);
+ break;
+ case 'a':
+ selected_encoder = encoder_selection_t::toolame_dab;
+ break;
+ case 'A':
+ afterburner = false;
+ break;
+ case 'b':
+ bitrate = atoi(optarg);
+ 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 'j':
+#if HAVE_JACK
+ jack_name = optarg;
+#else
+ fprintf(stderr, "JACK disabled at compile time!\n");
+ return 1;
+#endif
+ break;
+ case 'k':
+ keyfile = optarg;
+ break;
+ case 'l':
+ show_level = 1;
+ break;
+ case 'o':
+ output_uris.push_back(optarg);
+ break;
+ case 'p':
+ padlen = atoi(optarg);
+ break;
+ case 'P':
+ pad_fifo = optarg;
+ break;
+ case 'r':
+ sample_rate = atoi(optarg);
+ break;
+ case 's':
+ silence_timeout = atoi(optarg);
+ if (silence_timeout > 0 && silence_timeout < 3600*24*30) {
+ die_on_silence = true;
+ }
+ else {
+ fprintf(stderr, "Invalid silence timeout (%d) given!\n", silence_timeout);
+ return 1;
+ }
+
+ break;
+#ifdef HAVE_VLC
+ case 'v':
+ vlc_uri = optarg;
+ break;
+ case 'w':
+ vlc_icytext_file = optarg;
+ break;
+ case 'W':
+ vlc_icytext_dlplus = true;
+ break;
+ case 'g':
+ vlc_gain = optarg;
+ break;
+ case 'C':
+ vlc_cache = optarg;
+ break;
+ case 'L':
+ vlc_additional_opts.push_back(optarg);
+ break;
+#else
+ case 'v':
+ case 'w':
+ fprintf(stderr, "VLC input not enabled at compile time!\n");
+ return 1;
+#endif
+ case 'V':
+ verbosity++;
+ break;
+ case '?':
+ case 'h':
+ usage(argv[0]);
+ return 1;
+ }
+ }
+
+ int num_inputs = 0;
+ if (alsa_device) num_inputs++;
+ if (infile) num_inputs++;
+ if (jack_name) num_inputs++;
+ if (vlc_uri != "") num_inputs++;
+
+ if (num_inputs > 1) {
+ fprintf(stderr, "You must define only one possible input, not several!\n");
+ return 1;
+ }
+
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ if (bitrate == 0) {
+ bitrate = 64;
+ }
+
+ int subchannel_index = bitrate / 8;
+
+ if (subchannel_index < 1 || subchannel_index > 24) {
+ fprintf(stderr, "Bad subchannel index: %d, must be between 1 and 24. 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;
+ }
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ if (bitrate == 0) {
+ bitrate = 192;
+ }
+
+ if ( ! (sample_rate == 24000 || sample_rate == 48000)) {
+ fprintf(stderr, "Invalid sample rate. Possible values are: 24000, 48000.\n");
+ return 1;
+ }
+ }
+
+ if (padlen < 0) {
+ fprintf(stderr, "Invalid PAD length specified\n");
+ return 1;
+ }
+
+ zmq::context_t zmq_ctx;
+ zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
+
+ if (not output_uris.empty()) {
+ for (auto uri : output_uris) {
+ if (uri == "-") {
+ if (out_fh != NULL) {
+ fprintf(stderr, "You can't write to more than one file!\n");
+ return 1;
+ }
+ out_fh = stdout;
+ }
+ else if ((uri.compare(0, 6, "tcp://") == 0) ||
+ (uri.compare(0, 6, "pgm://") == 0) ||
+ (uri.compare(0, 7, "epgm://") == 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(uri.c_str());
+ }
+ else { // We assume it's a file name
+ if (out_fh != NULL) {
+ fprintf(stderr, "You can't write to more than one file!\n");
+ return 1;
+ }
+
+ out_fh = fopen(uri.c_str(), "wb");
+
+ if (!out_fh) {
+ fprintf(stderr, "Can't open output file!\n");
+ return 1;
+ }
+ }
+ }
+ }
+ else {
+ fprintf(stderr, "No output URI 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;
+ }
+ }
+
+
+ std::vector<uint8_t> input_buf;
+
+ HANDLE_AACENCODER encoder;
+
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ int subchannel_index = bitrate / 8;
+ if (prepare_aac_encoder(&encoder, subchannel_index, channels,
+ sample_rate, afterburner, &aot) != 0) {
+ fprintf(stderr, "Encoder preparation failed\n");
+ return 1;
+ }
+
+ 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);
+
+ input_buf.resize(input_size);
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ int err = toolame_init();
+
+ if (err == 0) {
+ err = toolame_set_samplerate(sample_rate);
+ }
+
+ if (err == 0) {
+ err = toolame_set_bitrate(bitrate);
+ }
+
+ if (err == 0) {
+ err = toolame_set_psy_model(dab_psy_model);
+ }
+
+ if (dab_channel_mode == '\0') {
+ if (channels == 2) {
+ dab_channel_mode = 'j'; // Default to joint-stereo
+ }
+ else if (channels == 1) {
+ dab_channel_mode = 'm'; // Default to mono
+ }
+ else {
+ fprintf(stderr, "Unsupported channels number %d\n", channels);
+ return 1;
+ }
+ }
+
+ if (err == 0) {
+ err = toolame_set_channel_mode(dab_channel_mode);
+ }
+
+ if (err == 0) {
+ err = toolame_set_pad(padlen);
+ }
+
+ if (err) {
+ fprintf(stderr, "libtoolame-dab init failed: %d\n", err);
+ return err;
+ }
+
+ input_buf.resize(channels * 1152 * BYTES_PER_SAMPLE);
+ }
+
+ /* 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. This is only valid for FDK-AAC.
+ */
+ const int enc_calls_per_output = (aot == AOT_DABPLUS_AAC_LC) ?
+ sample_rate / 8000 :
+ sample_rate / 16000;
+
+ int max_size = 8*input_buf.size() + NUM_SAMPLES_PER_CALL;
+
+ /*! The SampleQueue \c queue is given to the inputs, so that they
+ * can fill it.
+ */
+ 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
+#if HAVE_ALSA
+ AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
+ AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
+#endif
+ FileInput file_in(infile, raw_input, sample_rate);
+#if HAVE_JACK
+ JackInput jack_in(jack_name, channels, sample_rate, queue);
+#endif
+#if HAVE_VLC
+ VLCInput vlc_input(vlc_uri, sample_rate, channels, verbosity, vlc_gain, vlc_cache, vlc_additional_opts, queue);
+#endif
+
+ if (infile) {
+ if (file_in.prepare() != 0) {
+ fprintf(stderr, "File input preparation failed\n");
+ return 1;
+ }
+ }
+#if HAVE_JACK
+ else if (jack_name) {
+ if (jack_in.prepare() != 0) {
+ fprintf(stderr, "JACK preparation failed\n");
+ return 1;
+ }
+ }
+#endif
+#if HAVE_VLC
+ else if (vlc_uri != "") {
+ if (vlc_input.prepare() != 0) {
+ fprintf(stderr, "VLC with drift compensation: preparation failed\n");
+ return 1;
+ }
+
+ fprintf(stderr, "Start VLC thread\n");
+ vlc_input.start();
+ }
+#endif
+#if HAVE_ALSA
+ else if (drift_compensation) {
+ if (alsa_in_threaded.prepare() != 0) {
+ fprintf(stderr, "Alsa with drift compensation: 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;
+ }
+ }
+#else
+ else {
+ fprintf(stderr, "No input defined\n");
+ return 1;
+ }
+#endif
+
+ int outbuf_size;
+ std::vector<uint8_t> zmqframebuf;
+ std::vector<uint8_t> outbuf;
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ outbuf_size = bitrate/8*120;
+ outbuf.resize(24*120);
+ zmqframebuf.resize(ZMQ_HEADER_SIZE + 24*120);
+
+ if(outbuf_size % 5 != 0) {
+ fprintf(stderr, "Warning: (outbuf_size mod 5) = %d\n", outbuf_size % 5);
+ }
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ outbuf_size = 4092;
+ outbuf.resize(outbuf_size);
+ fprintf(stderr, "Setting outbuf size to %zu\n", outbuf.size());
+
+ // ODR-DabMux expects frames of length 3*bitrate
+ zmqframebuf.resize(ZMQ_HEADER_SIZE + 3 * bitrate);
+ }
+
+ zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)&zmqframebuf[0];
+
+ unsigned char pad_buf[padlen + 1];
+
+ fprintf(stderr, "Starting encoding\n");
+
+ int retval = 0;
+ int send_error_count = 0;
+ timepoint_last_compensation = chrono::steady_clock::now();
+
+ int calls = 0; // for checking
+ ssize_t read_bytes = 0;
+ do {
+ AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
+
+ // --------------- Read data from the PAD fifo
+ int ret;
+ if (padlen != 0) {
+ ret = read(pad_fd, pad_buf, padlen + 1);
+ }
+ 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[0], 0x00, outbuf_size);
+ memset(&input_buf[0], 0x00, input_buf.size());
+
+ /*! \section DataInput
+ * We read data input either in a blocking way (file input, VLC or ALSA
+ * without drift compensation) or in a non-blocking way (VLC or ALSA
+ * with drift compensation, JACK).
+ *
+ * The file input doesn't need the queue at all. But the other inputs
+ * do, and either use \c pop() or \c pop_wait() depending on if it's blocking or not
+ *
+ * In non-blocking, the \c queue makes the data available without delay, and the
+ * \c drift_compensation_delay() function handles rate throttling.
+ */
+
+ if (infile) {
+ read_bytes = file_in.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
+ break;
+ }
+ else if (read_bytes != input_buf.size()) {
+ if (inFifoSilence && file_in.eof()) {
+ memset(&input_buf[0], 0, input_buf.size());
+ read_bytes = input_buf.size();
+ usleep((long)input_buf.size() * 1000000 /
+ (BYTES_PER_SAMPLE * channels * sample_rate));
+ }
+ else {
+ fprintf(stderr, "Short file read !\n");
+ read_bytes = 0;
+ }
+ }
+ }
+#if HAVE_VLC
+ else if (not vlc_uri.empty()) {
+
+ if (drift_compensation && vlc_input.fault_detected()) {
+ fprintf(stderr, "Detected fault in VLC input!\n");
+ retval = 5;
+ break;
+ }
+
+ if (drift_compensation) {
+ size_t overruns;
+ size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
+ read_bytes = input_buf.size();
+ drift_compensation_delay(sample_rate, channels, read_bytes);
+
+ if (bytes_from_queue != input_buf.size()) {
+ status |= STATUS_UNDERRUN;
+ }
+
+ if (overruns) {
+ status |= STATUS_OVERRUN;
+ }
+ }
+ else {
+ const int timeout_ms = 1000;
+ read_bytes = input_buf.size();
+
+ /*! pop_wait() must return after a timeout, otherwise the silence detector cannot do
+ * its job.
+ */
+ size_t bytes_from_queue = queue.pop_wait(&input_buf[0], read_bytes, timeout_ms); // returns bytes
+
+ if (bytes_from_queue < read_bytes) {
+ // queue timeout occurred
+ fprintf(stderr, "Detected fault in VLC input! No data in time.\n");
+ retval = 5;
+ break;
+ }
+ }
+
+ if (not vlc_icytext_file.empty()) {
+ vlc_input.write_icy_text(vlc_icytext_file, vlc_icytext_dlplus);
+ }
+ }
+#endif
+ else if (drift_compensation || jack_name) {
+#if HAVE_ALSA
+ if (drift_compensation && alsa_in_threaded.fault_detected()) {
+ fprintf(stderr, "Detected fault in alsa input!\n");
+ retval = 5;
+ break;
+ }
+#endif
+
+ size_t overruns;
+ size_t bytes_from_queue = queue.pop(&input_buf[0], input_buf.size(), &overruns); // returns bytes
+ read_bytes = input_buf.size();
+ drift_compensation_delay(sample_rate, channels, read_bytes);
+
+ if (bytes_from_queue != input_buf.size()) {
+ status |= STATUS_UNDERRUN;
+ }
+
+ if (overruns) {
+ status |= STATUS_OVERRUN;
+ }
+ }
+ else {
+#if HAVE_ALSA
+ read_bytes = alsa_in_direct.read(&input_buf[0], input_buf.size());
+ if (read_bytes < 0) {
+ break;
+ }
+ else if (read_bytes != input_buf.size()) {
+ fprintf(stderr, "Short alsa read !\n");
+ }
+#endif
+ }
+
+ /*! \section AudioLevel
+ * Audio level measurement is always done assuming we have two
+ * channels, and is formally wrong in mono, but still gives
+ * numbers one can use.
+ *
+ * \todo fix level measurement in mono
+ */
+ for (int i = 0; i < read_bytes; 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);
+ }
+
+ /*! \section SilenceDetection
+ * Silence detection looks at the audio level and is
+ * only useful if the connection dropped, or if no data is available. It is not
+ * useful if the source is nearly silent (some noise present), because the
+ * threshold is 0, and not configurable. The rationale is that we want to
+ * guard against connection issues, not source level issues
+ */
+ if (die_on_silence && MAX(peak_left, peak_right) == 0) {
+ const unsigned int frame_time_msec = 1000ul *
+ read_bytes / (BYTES_PER_SAMPLE * channels * sample_rate);
+
+ measured_silence_ms += frame_time_msec;
+
+ if (measured_silence_ms > 1000*silence_timeout) {
+ fprintf(stderr, "Silence detected for %d seconds, aborting.\n",
+ silence_timeout);
+ retval = 2;
+ break;
+ }
+ }
+ else {
+ measured_silence_ms = 0;
+ }
+
+ int numOutBytes = 0;
+ if (read_bytes and
+ selected_encoder == encoder_selection_t::fdk_dabplus) {
+ AACENC_InArgs in_args = { 0 };
+ AACENC_OutArgs out_args = { 0 };
+ // -------------- AAC Encoding
+ //
+ int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
+ int out_identifier = OUT_BITSTREAM_DATA;
+ const int calculated_padlen = ret > 0 ? pad_buf[padlen] : 0;
+ const int subchannel_index = bitrate / 8;
+
+ void *in_ptr[2], *out_ptr;
+ int in_size[2], in_elem_size[2];
+ int out_size, out_elem_size;
+
+ in_ptr[0] = &input_buf[0];
+ in_ptr[1] = pad_buf + (padlen - calculated_padlen); // offset due to unused PAD bytes
+ in_size[0] = read_bytes;
+ in_size[1] = calculated_padlen;
+ in_elem_size[0] = BYTES_PER_SAMPLE;
+ in_elem_size[1] = sizeof(uint8_t);
+ in_args.numInSamples = input_buf.size()/BYTES_PER_SAMPLE;
+ in_args.numAncBytes = calculated_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[0];
+ out_size = outbuf.size();
+ 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);
+ retval = 3;
+ break;
+ }
+ calls++;
+
+ numOutBytes = out_args.numOutBytes;
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ int calculated_padlen = 0;
+ if (ret == padlen + 1) {
+ calculated_padlen = pad_buf[padlen];
+ if (calculated_padlen <= 2) {
+ stringstream ss;
+ ss << "Invalid XPAD Length " << calculated_padlen;
+ throw runtime_error(ss.str());
+ }
+ }
+
+ /*! \note toolame expects the audio to be in another shape as
+ * we have in input_buf, and we need to convert first
+ */
+ short input_buffers[2][1152];
+
+ if (channels == 1) {
+ memcpy(input_buffers[0], &input_buf[0], 1152 * BYTES_PER_SAMPLE);
+ }
+ else if (channels == 2) {
+ for (int i = 0; i < 1152; i++) {
+ int16_t l = input_buf[4*i] | (input_buf[4*i+1] << 8);
+ int16_t r = input_buf[4*i+2] | (input_buf[4*i+3] << 8);
+
+ input_buffers[0][i] = l;
+ input_buffers[1][i] = r;
+ }
+ }
+ else {
+ fprintf(stderr, "INTERNAL ERROR! invalid number of channels\n");
+ }
+
+ if (read_bytes) {
+ numOutBytes = toolame_encode_frame(input_buffers, pad_buf, calculated_padlen, &outbuf[0], outbuf.size());
+ }
+ else {
+ numOutBytes = toolame_finish(&outbuf[0], outbuf.size());
+ }
+ }
+
+ /* Check if the encoder has generated output data.
+ * DAB+ requires RS encoding, which is not done in ODR-DabMux and not necessary
+ * for DAB.
+ */
+ if (numOutBytes != 0 and
+ selected_encoder == encoder_selection_t::fdk_dabplus) {
+
+ // Our timing code depends on this
+ if (calls != enc_calls_per_output) {
+ fprintf(stderr, "INTERNAL ERROR! calls=%d"
+ ", expected %d\n",
+ calls, enc_calls_per_output);
+ }
+ calls = 0;
+
+ int row, col;
+ unsigned char buf_to_rs_enc[110];
+ unsigned char rs_enc[10];
+ const int subchannel_index = bitrate / 8;
+ 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);
+ }
+ }
+
+ numOutBytes = outbuf_size;
+ }
+
+ if (numOutBytes != 0) {
+ if (out_fh) {
+ fwrite(&outbuf[0], 1, numOutBytes, out_fh);
+ }
+ else {
+ // ------------ ZeroMQ transmit
+ try {
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
+ zmq_frame_header->version = 1;
+ zmq_frame_header->datasize = numOutBytes;
+ zmq_frame_header->audiolevel_left = peak_left;
+ zmq_frame_header->audiolevel_right = peak_right;
+
+ assert(ZMQ_FRAME_SIZE(zmq_frame_header) <= zmqframebuf.size());
+
+ memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
+ &outbuf[0], numOutBytes);
+
+ zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
+ ZMQ_DONTWAIT);
+
+ }
+ else if (selected_encoder == encoder_selection_t::toolame_dab) {
+ toolame_output_buffer.insert(toolame_output_buffer.end(),
+ outbuf.begin(), outbuf.begin() + numOutBytes);
+
+ while (toolame_output_buffer.size() > 3 * bitrate) {
+ zmq_frame_header->encoder = ZMQ_ENCODER_TOOLAME;
+ zmq_frame_header->version = 1;
+ zmq_frame_header->datasize = 3 * bitrate;
+ zmq_frame_header->audiolevel_left = peak_left;
+ zmq_frame_header->audiolevel_right = peak_right;
+
+ uint8_t *encoded_frame = ZMQ_FRAME_DATA(zmq_frame_header);
+
+ // no memcpy for std::deque
+ for (size_t i = 0; i < 3*bitrate; i++) {
+ encoded_frame[i] = toolame_output_buffer[i];
+ }
+
+ zmq_sock.send(&zmqframebuf[0], ZMQ_FRAME_SIZE(zmq_frame_header),
+ ZMQ_DONTWAIT);
+
+ toolame_output_buffer.erase(toolame_output_buffer.begin(),
+ toolame_output_buffer.begin() + 3 * bitrate);
+ }
+ }
+ }
+ 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");
+ retval = 4;
+ break;
+ }
+ }
+ }
+
+ if (numOutBytes != 0)
+ {
+ if (show_level) {
+ if (channels == 1) {
+ fprintf(stderr, "\rIn: [%-6s] %1s %1s %1s",
+ level(1, MAX(peak_right, peak_left)),
+ status & STATUS_PAD_INSERTED ? "P" : " ",
+ status & STATUS_UNDERRUN ? "U" : " ",
+ status & STATUS_OVERRUN ? "O" : " ");
+ }
+ else if (channels == 2) {
+ 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" : " ");
+ }
+ }
+ else {
+ if (status & STATUS_OVERRUN) {
+ fprintf(stderr, "O");
+ }
+
+ if (status & STATUS_UNDERRUN) {
+ fprintf(stderr, "U");
+ }
+
+ }
+
+ peak_right = 0;
+ peak_left = 0;
+
+ status = 0;
+ }
+
+ fflush(stdout);
+ } while (read_bytes > 0);
+
+ fprintf(stderr, "\n");
+
+ if (out_fh) {
+ fclose(out_fh);
+ }
+
+ zmq_sock.close();
+ free_rs_char(rs_handler);
+
+ if (selected_encoder == encoder_selection_t::fdk_dabplus) {
+ aacEncClose(&encoder);
+ }
+
+ return retval;
+}
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-11 05:52:08 +0000