rename AlsaDabplus.cpp to dabplus-enc-alsa-zmq.cpp

1 files changed, 0 insertions, 533 deletions

diff --git a/src/AlsaDabplus.cpp b/src/AlsaDabplus.cpp
deleted file mode 100644
index 99eec6e..0000000
--- a/src/AlsaDabplus.cpp
+++ /dev/null
@@ -1,533 +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"
-
-#include <string>
-#include <getopt.h>
-#include <cstdio>
-#include <stdint.h>
-#include <time.h>
-#include <unistd.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-
-#include "libAACenc/include/aacenc_lib.h"
-
-extern "C" {
-#include <fec.h>
-}
-
-using namespace std;
-
-void usage(const char* name) {
-    fprintf(stderr,
-    "dabplus-enc-alsa-zmq %s is a HE-AACv2 encoder for DAB+\n"
-    "based on fdk-aac-dabplus that can read from a ALSA source\n"
-    "and encode to a ZeroMQ output for ODR-DabMux.\n"
-    "\n"
-    "The -D option enables experimental sound card clock drift compensation.\n"
-    "A consumer sound card has a clock that is always a bit imprecise, and\n"
-    "would drift off after some time. ODR-DabMux cannot handle such drift\n"
-    "because it would have to throw away or insert a full DAB+ superframe,\n"
-    "which would create audible artifacts. This drift compensation can\n"
-    "make sure that the encoding rate is correct by inserting or deleting\n"
-    "audio samples.\n"
-    "\n"
-    "When this option is enabled, you will see U and O<number> printed in\n"
-    "the console. These correspond to audio underruns and overruns caused\n"
-    "by sound card clock drift. When sparse, they should not create audible\n"
-    "artifacts.\n"
-    "\n"
-    "This encoder includes PAD (DLS and MOT Slideshow) support by\n"
-    "http://rd.csp.it to be used with mot-encoder\n"
-    "\n"
-    "  http://opendigitalradio.org\n"
-    "\nUsage:\n"
-    "%s [OPTION...]\n",
-#if defined(GITVERSION)
-    GITVERSION
-#else
-    PACKAGE_VERSION
-#endif
-    , name);
-    fprintf(stderr,
-    "     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be 8 multiple.\n"
-    "     -D, --drift-comp                     Enable ALSA sound card drift compensation.\n"
-    //"   -i, --input=FILENAME                 Input filename (default: stdin).\n"
-    "     -o, --output=URI                     Output zmq uri. (e.g. 'tcp://*:9000')\n"
-    "     -a, --afterburner                    Turn on AAC encoder quality increaser.\n"
-    "     -p, --pad=BYTES                      Set PAD size in bytes.\n"
-    "     -P, --pad-fifo=FILENAME              Set PAD data input fifo name (default: /tmp/pad.fifo).\n"
-    "     -d, --device=alsa_device             Set ALSA input device (default: \"default\").\n"
-    "     -c, --channels={ 1, 2 }              Nb of input channels for raw input (default: 2).\n"
-    "     -r, --rate={ 32000, 48000 }          Sample rate for raw input (default: 48000).\n"
-    //"   -v, --verbose=LEVEL                  Set verbosity level.\n"
-    //"   -V, --version                        Print version and exit.\n"
-    "\n"
-    "Only the tcp:// zeromq transport has been tested until now.\n"
-    );
-
-}
-
-int prepare_aac_encoder(
-        HANDLE_AACENCODER *encoder,
-        int subchannel_index,
-        int channels,
-        int sample_rate,
-        int afterburner)
-{
-    HANDLE_AACENCODER handle = *encoder;
-
-    int aot = AOT_DABPLUS_AAC_LC;
-
-    CHANNEL_MODE mode;
-    switch (channels) {
-        case 1: mode = MODE_1; break;
-        case 2: mode = MODE_2; break;
-        default:
-                fprintf(stderr, "Unsupported channels number %d\n", channels);
-                return 1;
-    }
-
-
-    if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
-        fprintf(stderr, "Unable to open encoder\n");
-        return 1;
-    }
-
-    *encoder = handle;
-
-    if(channels == 2 && subchannel_index <= 6)
-        aot = AOT_DABPLUS_PS;
-    else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10)
-        aot = AOT_DABPLUS_SBR;
-
-    fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
-            subchannel_index,
-            aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
-            aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
-            aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
-            channels, sample_rate);
-
-    if (aacEncoder_SetParam(handle, AACENC_AOT, aot) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the AOT\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the sample rate\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the channel mode\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the wav channel order\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the granule length\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the RAW transmux\n");
-        return 1;
-    }
-
-    /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*)
-     * != AACENC_OK) {
-        fprintf(stderr, "Unable to set the bitrate mode\n");
-        return 1;
-    }*/
-
-
-    fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
-    if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the bitrate\n");
-        return 1;
-    }
-    if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
-        fprintf(stderr, "Unable to set the afterburner mode\n");
-        return 1;
-    }
-    if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
-        fprintf(stderr, "Unable to initialize the encoder\n");
-        return 1;
-    }
-    return 0;
-}
-
-
-#define no_argument 0
-#define required_argument 1
-#define optional_argument 2
-
-int main(int argc, char *argv[]) {
-    int subchannel_index = 8; //64kbps subchannel
-    int ch=0;
-    const char *alsa_device = "default";
-    const char *outuri = NULL;
-    int sample_rate=48000, channels=2;
-    const int bytes_per_sample = 2;
-    void *rs_handler = NULL;
-    bool afterburner = false;
-    bool drift_compensation = false;
-    AACENC_InfoStruct info = { 0 };
-
-    char* pad_fifo = "/tmp/pad.fifo";
-    int pad_fd;
-    unsigned char pad_buf[128];
-    int padlen;
-
-    const struct option longopts[] = {
-        {"bitrate",     required_argument,  0, 'b'},
-        {"output",      required_argument,  0, 'o'},
-        {"device",      required_argument,  0, 'd'},
-        {"rate",        required_argument,  0, 'r'},
-        {"channels",    required_argument,  0, 'c'},
-        {"pad",         required_argument,  0, 'p'},
-        {"pad-fifo",    required_argument,  0, 'P'},
-        {"drift-comp",  no_argument,        0, 'D'},
-        {"afterburner", no_argument,        0, 'a'},
-        {"help",        no_argument,        0, 'h'},
-        {0,0,0,0},
-    };
-
-    if (argc < 2) {
-        usage(argv[0]);
-        return 1;
-    }
-
-    int index;
-    while(ch != -1) {
-        ch = getopt_long(argc, argv, "hab:c:o:r:d:Dp:P:", longopts, &index);
-        switch (ch) {
-        case 'd':
-            alsa_device = optarg;
-            break;
-        case 'a':
-            afterburner = true;
-            break;
-        case 'b':
-            subchannel_index = atoi(optarg) / 8;
-            break;
-        case 'c':
-            channels = atoi(optarg);
-            break;
-        case 'r':
-            sample_rate = atoi(optarg);
-            break;
-        case 'o':
-            outuri = optarg;
-            break;
-        case 'D':
-            drift_compensation = true;
-            break;
-        case 'p':
-            padlen = atoi(optarg);
-            break;
-        case 'P':
-            pad_fifo = optarg;
-            break;
-        case '?':
-        case 'h':
-            usage(argv[0]);
-            return 1;
-        }
-    }
-
-    if(subchannel_index < 1 || subchannel_index > 24) {
-        fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n",
-                subchannel_index);
-        return 1;
-    }
-
-    fprintf(stderr, "Setting up ZeroMQ socket\n");
-    if (!outuri) {
-        fprintf(stderr, "ZeroMQ output URI not defined\n");
-        return 1;
-    }
-
-    if (padlen != 0) {
-        int flags;
-        if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) {
-            if (errno != EEXIST) {
-                fprintf(stderr, "Can't create pad file: %d!\n", errno);
-                return 1;
-            }
-        }
-        pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK);
-        if (pad_fd == -1) {
-            fprintf(stderr, "Can't open pad file!\n");
-            return 1;
-        }
-        flags = fcntl(pad_fd, F_GETFL, 0);
-        if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) {
-            fprintf(stderr, "Can't set non-blocking mode in pad file!\n");
-            return 1;
-        }
-    }
-
-    zmq::context_t zmq_ctx;
-    zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);
-    zmq_sock.connect(outuri);
-
-    HANDLE_AACENCODER encoder;
-
-    if (prepare_aac_encoder(&encoder, subchannel_index, channels,
-                sample_rate, afterburner) != 0) {
-        fprintf(stderr, "Encoder preparation failed\n");
-        return 2;
-    }
-
-    if (aacEncInfo(encoder, &info) != AACENC_OK) {
-        fprintf(stderr, "Unable to get the encoder info\n");
-        return 1;
-    }
-
-    // Each DAB+ frame will need input_size audio bytes
-    const int input_size = channels * bytes_per_sample * info.frameLength;
-    fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n",
-            info.frameLength,
-            input_size);
-
-    uint8_t input_buf[input_size];
-
-    int max_size = 2*input_size + NUM_SAMPLES_PER_CALL;
-
-    SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);
-
-    /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
-    rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
-    if (rs_handler == NULL) {
-        perror("init_rs_char failed");
-        return 1;
-    }
-
-    // We'll use either of the two possible alsa inputs.
-    AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
-    AlsaInputDirect alsa_in_direct(alsa_device, channels, sample_rate);
-
-    if (drift_compensation) {
-        if (alsa_in_threaded.prepare() != 0) {
-            fprintf(stderr, "Alsa preparation failed\n");
-            return 1;
-        }
-
-        fprintf(stderr, "Start ALSA capture thread\n");
-        alsa_in_threaded.start();
-    }
-    else {
-        if (alsa_in_direct.prepare() != 0) {
-            fprintf(stderr, "Alsa preparation failed\n");
-            return 1;
-        }
-    }
-
-    int outbuf_size = subchannel_index*120;
-    uint8_t outbuf[20480];
-
-    if(outbuf_size % 5 != 0) {
-        fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
-    }
-
-    fprintf(stderr, "Starting encoding\n");
-
-    int send_error_count = 0;
-    struct timespec tp_next;
-    clock_gettime(CLOCK_MONOTONIC, &tp_next);
-
-    int calls = 0; // for checking
-    while (1) {
-        int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
-        int out_identifier = OUT_BITSTREAM_DATA;
-
-        AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
-        AACENC_InArgs in_args = { 0 };
-        AACENC_OutArgs out_args = { 0 };
-        void *in_ptr[2], *out_ptr;
-        int in_size[2], in_elem_size[2];
-        int out_size, out_elem_size;
-
-
-        // -------------- wait the right amount of time
-        if (drift_compensation) {
-            struct timespec tp_now;
-            clock_gettime(CLOCK_MONOTONIC, &tp_now);
-
-            unsigned long time_now  = (1000000000ul * tp_now.tv_sec) +
-                tp_now.tv_nsec;
-            unsigned long time_next = (1000000000ul * tp_next.tv_sec) +
-                tp_next.tv_nsec;
-
-            const unsigned long wait_time = 120000000ul / 2;
-
-            unsigned long waiting = wait_time - (time_now - time_next);
-            if ((time_now - time_next) < wait_time) {
-                //printf("Sleep %zuus\n", waiting / 1000);
-                usleep(waiting / 1000);
-            }
-
-            // Move our time_counter 60ms into the future.
-            // The encoder needs two calls for one frame
-            tp_next.tv_nsec += wait_time;
-            if (tp_next.tv_nsec >  1000000000L) {
-                tp_next.tv_nsec -= 1000000000L;
-                tp_next.tv_sec  += 1;
-            }
-        }
-
-        // --------------- Read data from the PAD fifo
-        int ret;
-        if (padlen != 0) {
-            ret = read(pad_fd, pad_buf, padlen);
-        }
-        else {
-            ret = 0;
-        }
-
-
-        if(ret < 0 && errno == EAGAIN) {
-            // If this condition passes, there is no data to be read
-            in_buf.numBufs = 1;    // Samples;
-        }
-        else if(ret >= 0) {
-            // Otherwise, you're good to go and buffer should contain "count" bytes.
-            in_buf.numBufs = 2;    // Samples + Data;
-            if (ret > 0)
-                fprintf(stderr, "p");
-        }
-        else {
-            // Some other error occurred during read.
-            fprintf(stderr, "Unable to read from PAD!\n");
-                        break;
-        }
-
-        // -------------- Read Data
-        memset(outbuf, 0x00, outbuf_size);
-
-        size_t read;
-        if (drift_compensation) {
-            size_t overruns;
-            read = queue.pop(input_buf, input_size, &overruns); // returns bytes
-
-            if (read != input_size) {
-                fprintf(stderr, "U");
-            }
-
-            if (overruns) {
-                fprintf(stderr, "O%zu", overruns);
-            }
-        }
-        else {
-            read = alsa_in_direct.read(input_buf, input_size);
-            if (read != input_size) {
-                fprintf(stderr, "Short alsa read !\n");
-            }
-        }
-
-        // -------------- AAC Encoding
-
-        in_ptr[0] = input_buf;
-        in_ptr[1] = pad_buf;
-        in_size[0] = read;
-        in_size[1] = padlen;
-        in_elem_size[0] = BYTES_PER_SAMPLE;
-        in_elem_size[1] = sizeof(uint8_t);
-        in_args.numInSamples = input_size/BYTES_PER_SAMPLE;
-        in_args.numAncBytes = padlen;
-
-        in_buf.bufs = (void**)&in_ptr;
-        in_buf.bufferIdentifiers = in_identifier;
-        in_buf.bufSizes = in_size;
-        in_buf.bufElSizes = in_elem_size;
-
-        out_ptr = outbuf;
-        out_size = sizeof(outbuf);
-        out_elem_size = 1;
-        out_buf.numBufs = 1;
-        out_buf.bufs = &out_ptr;
-        out_buf.bufferIdentifiers = &out_identifier;
-        out_buf.bufSizes = &out_size;
-        out_buf.bufElSizes = &out_elem_size;
-
-        AACENC_ERROR err;
-        if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args))
-                != AACENC_OK) {
-            if (err == AACENC_ENCODE_EOF)
-                break;
-            fprintf(stderr, "Encoding failed\n");
-            break;
-        }
-        calls++;
-
-        /* Check if the encoder has generated output data */
-        if (out_args.numOutBytes != 0)
-        {
-            // Our timing code depends on this
-            assert (calls == 2);
-            calls = 0;
-
-            // ----------- RS encoding
-            int row, col;
-            unsigned char buf_to_rs_enc[110];
-            unsigned char rs_enc[10];
-            for(row=0; row < subchannel_index; row++) {
-                for(col=0;col < 110; col++) {
-                    buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
-                }
-
-                encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);
-
-                for(col=110; col<120; col++) {
-                    outbuf[subchannel_index * col + row] = rs_enc[col-110];
-                    assert(subchannel_index * col + row < outbuf_size);
-                }
-            }
-
-            // ------------ ZeroMQ transmit
-            try {
-                zmq_sock.send(outbuf, outbuf_size, ZMQ_DONTWAIT);
-            }
-            catch (zmq::error_t& e) {
-                fprintf(stderr, "ZeroMQ send error !\n");
-                send_error_count ++;
-            }
-
-            if (send_error_count > 10)
-            {
-                fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n");
-                break;
-            }
-
-            if (out_args.numOutBytes + row*10 == outbuf_size)
-                fprintf(stderr, ".");
-        }
-    }
-
-    zmq_sock.close();
-    free_rs_char(rs_handler);
-
-    aacEncClose(&encoder);
-
-}
-