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-rw-r--r--src/dabplus-enc-alsa-zmq.cpp635
1 files changed, 0 insertions, 635 deletions
diff --git a/src/dabplus-enc-alsa-zmq.cpp b/src/dabplus-enc-alsa-zmq.cpp
deleted file mode 100644
index 699a256..0000000
--- a/src/dabplus-enc-alsa-zmq.cpp
+++ /dev/null
@@ -1,635 +0,0 @@
-/* ------------------------------------------------------------------
- * 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"
-
-extern "C" {
-#include "encryption.h"
-#include "utils.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-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"
- " -k, --secret-key=FILE Set the secret key for encryption.\n"
- " -l, --level Show level indication.\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
-
-#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;
- 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 };
-
- /* Keep track of peaks */
- int peak_left = 0;
- int peak_right = 0;
-
-
- const char* pad_fifo = "/tmp/pad.fifo";
- int pad_fd;
- unsigned char pad_buf[128];
- int padlen;
-
- int status = 0;
- 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'},
- {"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'},
- {"secret-key", required_argument, 0, 'k'},
- {"drift-comp", no_argument, 0, 'D'},
- {"afterburner", no_argument, 0, 'a'},
- {"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:k:o:r:d:p: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 'k':
- keyfile = optarg;
- break;
- case 'D':
- drift_compensation = true;
- break;
- case 'p':
- padlen = atoi(optarg);
- break;
- case 'P':
- pad_fifo = optarg;
- break;
- case 'l':
- show_level = 1;
- break;
- case '?':
- case 'h':
- usage(argv[0]);
- return 1;
- }
- }
-
- if(subchannel_index < 1 || subchannel_index > 24) {
- fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n",
- subchannel_index);
- return 1;
- }
-
- if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
- fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
- return 1;
- }
-
- const int enc_calls_per_output = sample_rate / 16000;
-
- 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);
-
- 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);
-
- 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 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);
-
- ssize_t read;
- 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);
-}
-