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Diffstat (limited to 'src/AlsaDabplus.cpp')
| -rw-r--r-- | src/AlsaDabplus.cpp | 533 |
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); - -} - |