diff options
Diffstat (limited to 'src/odr-audioencoder.cpp')
| -rw-r--r-- | src/odr-audioencoder.cpp | 1304 |
1 files changed, 0 insertions, 1304 deletions
diff --git a/src/odr-audioencoder.cpp b/src/odr-audioencoder.cpp deleted file mode 100644 index 80b7ae6..0000000 --- a/src/odr-audioencoder.cpp +++ /dev/null @@ -1,1304 +0,0 @@ -/* ------------------------------------------------------------------ - * 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-AudioEncoder 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-audioencoder.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-audioencoder.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-AudioEncoder %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; -} - |