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Diffstat (limited to 'src/dabplus-enc.cpp')
| -rw-r--r-- | src/dabplus-enc.cpp | 716 |
1 files changed, 716 insertions, 0 deletions
diff --git a/src/dabplus-enc.cpp b/src/dabplus-enc.cpp new file mode 100644 index 0000000..82780d5 --- /dev/null +++ b/src/dabplus-enc.cpp @@ -0,0 +1,716 @@ +/* ------------------------------------------------------------------ + * 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 "FileInput.h" +#include "SampleQueue.h" +#include "zmq.hpp" + +extern "C" { +#include "encryption.h" +#include "utils.h" +#include "wavreader.h" +} + +#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 "libAACenc/include/aacenc_lib.h" + +extern "C" { +#include <fec.h> +} + +using namespace std; + +void usage(const char* name) { + fprintf(stderr, + "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n" + "based on fdk-aac-dabplus that can read from a ALSA or file 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 (-i file|-d alsa_device) [OPTION...]\n", +#if defined(GITVERSION) + GITVERSION +#else + PACKAGE_VERSION +#endif + , name); + fprintf(stderr, + " For the alsa input:\n" + " -d, --device=alsa_device Set ALSA input device (default: \"default\").\n" + " -D, --drift-comp Enable ALSA sound card drift compensation.\n" + " For the file input:\n" + " -i, --input=FILENAME Input filename (default: stdin).\n" + " -f, --format={ wav, raw } Set input file format (default: wav).\n" + " Encoder parameters:\n" + " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" + " -o, --output=URI Output zmq uri. (e.g. 'tcp://*:9000')\n" + " -or- Output file uri. (e.g. 'file.dab')\n" + " -or- a single dash '-' to denote stdout\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" + " -c, --channels={ 1, 2 } Nb of input channels (default: 2).\n" + " -r, --rate={ 32000, 48000 } Input sample rate (default: 48000).\n" + " -k, --secret-key=FILE Enable ZMQ encryption with the given secret key.\n" + " -l, --level Show peak audio level indication.\n" + "\n" + "Only the tcp:// zeromq transport has been tested until now,\n" + " but epgm:// and pgm:// are also accepted\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 + +#define STATUS_PAD_INSERTED 0x1 +#define STATUS_OVERRUN 0x2 +#define STATUS_UNDERRUN 0x4 + +int main(int argc, char *argv[]) +{ + int subchannel_index = 8; //64kbps subchannel + int ch=0; + + // For the ALSA input + const char *alsa_device = NULL; + + // For the file input + const char *infile = NULL; + int raw_input = 0; + + // For the file output + FILE *out_fh; + + 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 }; + + /* Keep track of peaks */ + int peak_left = 0; + int peak_right = 0; + + /* For MOT Slideshow and DLS insertion */ + const char* pad_fifo = "/tmp/pad.fifo"; + int pad_fd; + unsigned char pad_buf[128]; + int padlen; + + /* 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'}, + {"device", required_argument, 0, 'd'}, + {"format", required_argument, 0, 'f'}, + {"input", required_argument, 0, 'i'}, + {"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'}, + {"afterburner", no_argument, 0, 'a'}, + {"drift-comp", no_argument, 0, 'D'}, + {"help", no_argument, 0, 'h'}, + {"level", no_argument, 0, 'l'}, + {0,0,0,0}, + }; + + if (argc < 2) { + usage(argv[0]); + return 1; + } + + int index; + while(ch != -1) { + ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index); + switch (ch) { + case 'a': + afterburner = true; + break; + case 'b': + subchannel_index = atoi(optarg) / 8; + 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 'k': + keyfile = optarg; + break; + case 'l': + show_level = 1; + break; + case 'o': + outuri = optarg; + break; + case 'p': + padlen = atoi(optarg); + break; + case 'P': + pad_fifo = optarg; + break; + case 'r': + sample_rate = atoi(optarg); + break; + case '?': + case 'h': + usage(argv[0]); + return 1; + } + } + + if (alsa_device && infile) { + fprintf(stderr, "You must define either alsa or file input, not both\n"); + return 1; + } + + if (subchannel_index < 1 || subchannel_index > 24) { + fprintf(stderr, "Bad subchannels number: %d, 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; + } + + /* 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 + */ + const int enc_calls_per_output = sample_rate / 16000; + + zmq::context_t zmq_ctx; + zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB); + + if (outuri) { + if (strcmp(outuri, "-") == 0) { + out_fh = stdout; + } + else if ((strncmp(outuri, "tcp://", 6) == 0) || + (strncmp(outuri, "pgm://", 6) == 0) || + (strncmp(outuri, "epgm://", 7) == 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(outuri); + } + else { // We assume it's a file name + out_fh = fopen(outuri, "wb"); + + if (!out_fh) { + fprintf(stderr, "Can't open output file!\n"); + return 1; + } + } + } + else { + fprintf(stderr, "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; + } + } + + + 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; + } + + /* No input defined ? default to alsa "default" */ + if (!alsa_device) { + alsa_device = "default"; + } + + // We'll use one of the tree possible inputs + AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue); + AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate); + FileInput file_in(infile, raw_input, sample_rate); + + if (infile) { + if (file_in.prepare() != 0) { + fprintf(stderr, "File input preparation failed\n"); + return 1; + } + } + else 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 zmqframebuf[2048]; + zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf; + + uint8_t outbuf[2048]; + + 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 dabplus_superframe_nsec = 120000000ul; + + const unsigned long wait_time = + dabplus_superframe_nsec / enc_calls_per_output; + + 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) + status |= STATUS_PAD_INSERTED; + } + else { + // Some other error occurred during read. + fprintf(stderr, "Unable to read from PAD!\n"); + break; + } + + // -------------- Read Data + memset(outbuf, 0x00, outbuf_size); + memset(input_buf, 0x00, input_size); + + ssize_t read; + if (infile) { + read = file_in.read(input_buf, input_size); + if (read < 0) { + break; + } + else if (read != input_size) { + fprintf(stderr, "Short file read !\n"); + break; + } + } + else if (drift_compensation) { + if (alsa_in_threaded.fault_detected()) { + fprintf(stderr, "Detected fault in alsa input!\n"); + break; + } + + size_t overruns; + read = queue.pop(input_buf, input_size, &overruns); // returns bytes + + if (read != input_size) { + status |= STATUS_UNDERRUN; + } + + if (overruns) { + status |= STATUS_OVERRUN; + } + } + else { + read = alsa_in_direct.read(input_buf, input_size); + if (read < 0) { + break; + } + else if (read != input_size) { + fprintf(stderr, "Short alsa read !\n"); + } + } + + for (int i = 0; i < read; 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); + } + + // -------------- 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) { + fprintf(stderr, "encoder error: EOF reached\n"); + break; + } + fprintf(stderr, "Encoding failed (%d)\n", err); + break; + } + calls++; + + /* Check if the encoder has generated output data */ + if (out_args.numOutBytes != 0) + { + // Our timing code depends on this + if (! ((sample_rate == 32000 && calls == 2) || + (sample_rate == 48000 && calls == 3)) ) { + fprintf(stderr, "INTERNAL ERROR! sample rate %d, calls %d\n", + sample_rate, calls); + } + 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_frame_header->version = 1; + zmq_frame_header->encoder = ZMQ_ENCODER_FDK; + zmq_frame_header->datasize = outbuf_size; + zmq_frame_header->audiolevel_left = peak_left; + zmq_frame_header->audiolevel_right = peak_right; + + memcpy(ZMQ_FRAME_DATA(zmq_frame_header), + outbuf, outbuf_size); + + zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header), + 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 (show_level && out_args.numOutBytes + row*10 == outbuf_size) { + 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" : " "); + } + + status = 0; + } + + fflush(stdout); + } + fprintf(stderr, "\n"); + + zmq_sock.close(); + free_rs_char(rs_handler); + + aacEncClose(&encoder); +} + |