/* ------------------------------------------------------------------ * 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 #include #include #include #include #include #include #include #include #include #include "libAACenc/include/aacenc_lib.h" #include #include #include #include static struct { snd_pcm_format_t format; unsigned int channels; unsigned int rate; } hwparams; void usage(const char* name) { fprintf(stderr, "%s [OPTION...]\n", name); fprintf(stderr, " -b, --bitrate={ 8, 16, ..., 192 } Output bitrate in kbps. Must be 8 multiple.\n" //" -d, --data=FILENAME Set data filename.\n" //" -g, --fs-bug Turn on FS bug mitigation.\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" //" -m, --message Turn on AAC frame messages.\n" //" -p, --pad=BYTES Set PAD size in bytes.\n" //" -f, --format={ wav, raw } Set input file format (default: wav).\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" //" -t, --type=TYPE Set data type (dls|pad|packet|dg).\n" //" -v, --verbose=LEVEL Set verbosity level.\n" //" -V, --version Print version and exit.\n" //" --mi=[ 0, ... ] Set AAC frame messages interval in milliseconds.\n" //" --ma=[ 0, ... ] Set AAC frame messages attack time in milliseconds.\n" //" -l, --lp Set frame size to 1024 instead of 960.\n" "\n" "Only the tcp:// zeromq transport has been tested until now.\n" ); } static snd_pcm_t *alsa_handle = NULL; static void prg_exit(int code) { if (alsa_handle) { snd_pcm_close(alsa_handle); } exit(code); } static void alsa_prepare(const char* alsa_dev, unsigned int rate, unsigned int channels) { int err; snd_pcm_hw_params_t *hw_params; fprintf(stderr, "Initialising ALSA...\n"); const int open_mode = 0; //|= SND_PCM_NONBLOCK; if ((err = snd_pcm_open(&alsa_handle, alsa_dev, SND_PCM_STREAM_CAPTURE, open_mode)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", alsa_dev, snd_strerror(err)); prg_exit(1); } const int nonblock = 0; //TODO remove dead code if (nonblock) { err = snd_pcm_nonblock(alsa_handle, 1); if (err < 0) { fprintf(stderr, "nonblock setting error: %s", snd_strerror(err)); prg_exit(1); } } if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params_any(alsa_handle, hw_params)) < 0) { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params_set_access(alsa_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params_set_format(alsa_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params_set_rate_near(alsa_handle, hw_params, &rate, 0)) < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params_set_channels(alsa_handle, hw_params, channels)) < 0) { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror(err)); prg_exit(1); } if ((err = snd_pcm_hw_params(alsa_handle, hw_params)) < 0) { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror(err)); prg_exit(1); } snd_pcm_hw_params_free (hw_params); if ((err = snd_pcm_prepare(alsa_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror(err)); prg_exit(1); } fprintf(stderr, "ALSA init done.\n"); } static size_t alsa_read(uint8_t* buf, snd_pcm_uframes_t length) { int i; int err; err = snd_pcm_readi(alsa_handle, buf, length); if (err != length) { if (err < 0) { fprintf (stderr, "read from audio interface failed (%s)\n", snd_strerror(err)); } else { fprintf(stderr, "short alsa read: %d\n", err); } } return err; } static void signal_handler(int sig) { fprintf(stderr, "Caught signal %d\n", sig); if (alsa_handle) { snd_pcm_abort(alsa_handle); alsa_handle = NULL; } if (sig == SIGABRT) { /* do not call snd_pcm_close() and abort immediately */ alsa_handle = NULL; exit(EXIT_FAILURE); } signal(sig, signal_handler); } #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; int err; const char *alsa_device = "default"; const char *outuri = NULL; int sample_rate=48000, channels=2; const int bytes_per_sample = 2; uint8_t* input_buf; int16_t* convert_buf; void *rs_handler = NULL; int aot = AOT_DABPLUS_AAC_LC; int afterburner = 0; HANDLE_AACENCODER handle; CHANNEL_MODE mode; AACENC_InfoStruct info = { 0 }; void *zmq_context = zmq_ctx_new(); void *zmq_sock = NULL; 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'}, //{"lp", no_argument, 0, 'l'}, {"afterburner", no_argument, 0, 'a'}, {"help", no_argument, 0, 'h'}, {0,0,0,0}, }; int index; while(ch != -1) { ch = getopt_long(argc, argv, "lhab:c:o:r:d:", longopts, &index); switch (ch) { case 'd': alsa_device = optarg; break; case 'a': afterburner = 1; 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 '?': 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) { zmq_sock = zmq_socket(zmq_context, ZMQ_PUB); if (zmq_sock == NULL) { fprintf(stderr, "Error occurred during zmq_socket: %s\n", zmq_strerror(errno)); return 2; } if (zmq_connect(zmq_sock, outuri) != 0) { fprintf(stderr, "Error occurred during zmq_connect: %s\n", zmq_strerror(errno)); return 2; } } else { fprintf(stderr, "Output URI not defined\n"); return 1; } alsa_prepare(alsa_device, sample_rate, channels); signal(SIGINT, signal_handler); signal(SIGTERM, signal_handler); signal(SIGABRT, signal_handler); switch (channels) { case 1: mode = MODE_1; break; case 2: mode = MODE_2; break; default: fprintf(stderr, "Unsupported channels number %d\n", channels); prg_exit(1); } if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) { fprintf(stderr, "Unable to open encoder\n"); prg_exit(1); } 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"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) { fprintf(stderr, "Unable to set the AOT\n"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) { fprintf(stderr, "Unable to set the channel mode\n"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) { fprintf(stderr, "Unable to set the wav channel order\n"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) { fprintf(stderr, "Unable to set the AOT\n"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) { fprintf(stderr, "Unable to set the RAW transmux\n"); prg_exit(1); } /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*) != AACENC_OK) { fprintf(stderr, "Unable to set the bitrate mode\n"); prg_exit(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"); prg_exit(1); } if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) { fprintf(stderr, "Unable to set the afterburner mode\n"); prg_exit(1); } if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) { fprintf(stderr, "Unable to initialize the encoder\n"); prg_exit(1); } if (aacEncInfo(handle, &info) != AACENC_OK) { fprintf(stderr, "Unable to get the encoder info\n"); prg_exit(1); } fprintf(stderr, "DAB+ Encoding: framelen=%d\n", info.frameLength); int input_size = channels * bytes_per_sample * info.frameLength; input_buf = (uint8_t*) malloc(input_size); convert_buf = (int16_t*) malloc(input_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"); prg_exit(1); } int loops = 0; 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, "outbuf_size: %d\n", outbuf_size); //outbuf_size += (4 * subchannel_index * (8*8)/8) - outbuf_size/5; //fprintf(stderr, "outbuf_size: %d\n", outbuf_size); int frame=0; int send_error_count = 0; while (1) { memset(outbuf, 0x00, outbuf_size); AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; AACENC_InArgs in_args = { 0 }; AACENC_OutArgs out_args = { 0 }; int in_identifier = IN_AUDIO_DATA; int in_size, in_elem_size; int out_identifier = OUT_BITSTREAM_DATA; int out_size, out_elem_size; int read=0, i; int send_error; void *in_ptr, *out_ptr; AACENC_ERROR err; read = alsa_read(input_buf, info.frameLength); if (read != info.frameLength) { fprintf(stderr, "Unable to read enough data from input!\n"); break; } for (i = 0; i < read/2; i++) { const uint8_t* in = &input_buf[2*i]; convert_buf[i] = in[0] | (in[1] << 8); } if (read <= 0) { in_args.numInSamples = -1; } else { in_ptr = convert_buf; in_size = read; in_elem_size = 2; in_args.numInSamples = read/2; in_buf.numBufs = 1; in_buf.bufs = &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; if ((err = aacEncEncode(handle, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) { if (err == AACENC_ENCODE_EOF) break; fprintf(stderr, "Encoding failed\n"); prg_exit(1); } if (out_args.numOutBytes == 0) continue; #if 0 unsigned char au_start[6]; unsigned char* sfbuf = outbuf; au_start[0] = 6; au_start[1] = (*(sfbuf + 3) << 4) + ((*(sfbuf + 4)) >> 4); au_start[2] = ((*(sfbuf + 4) & 0x0f) << 8) + *(sfbuf + 5); fprintf (stderr, "au_start[0] = %d\n", au_start[0]); fprintf (stderr, "au_start[1] = %d\n", au_start[1]); fprintf (stderr, "au_start[2] = %d\n", au_start[2]); #endif 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); } } send_error = zmq_send(zmq_sock, outbuf, outbuf_size, ZMQ_DONTWAIT); if (send_error < 0) { fprintf(stderr, "ZeroMQ send failed! %s\n", zmq_strerror(errno)); send_error_count ++; } if (send_error_count > 10) { fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n"); break; } //fwrite(outbuf, 1, /*out_args.numOutBytes*/ outbuf_size, out_fh); //fprintf(stderr, "Written %d/%d bytes!\n", out_args.numOutBytes + row*10, outbuf_size); if(out_args.numOutBytes + row*10 == outbuf_size) fprintf(stderr, "."); // if(frame > 10) // break; frame++; } zmq_close(zmq_sock); free_rs_char(rs_handler); aacEncClose(&handle); zmq_ctx_term(zmq_context); prg_exit(0); }