/* * mp3rtp command line frontend program * * initially contributed by Felix von Leitner * * Copyright (c) 2000 Mark Taylor * 2010 Robert Hegemann * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /* $Id: mp3rtp.c,v 1.36 2011/10/02 17:13:22 robert Exp $ */ /* Still under work ..., need a client for test, where can I get one? */ /* An audio player named Zinf (aka freeamp) can play rtp streams */ /* * experimental translation: * * gcc -I..\include -I..\libmp3lame -o mp3rtp mp3rtp.c ../libmp3lame/libmp3lame.a lametime.c get_audio.c ieeefloat.c timestatus.c parse.c rtp.c -lm * * wavrec -t 14400 -s 44100 -S /proc/self/fd/1 | ./mp3rtp 10.1.1.42 -V2 -b128 -B256 - my_mp3file.mp3 */ #ifdef HAVE_CONFIG_H # include #endif #ifdef HAVE_STDINT_H # include #endif #ifdef STDC_HEADERS # include # include #endif #include #ifdef HAVE_UNISTD_H # include #endif #include "lame.h" #include "main.h" #include "parse.h" #include "lametime.h" #include "timestatus.h" #include "get_audio.h" #include "rtp.h" #include "console.h" #ifdef WITH_DMALLOC #include #endif /* * Encode (via LAME) to mp3 with RTP streaming of the output. * * Author: Felix von Leitner * * mp3rtp ip[:port[:ttl]] [lame encoding options] infile outfile * * examples: * arecord -b 16 -s 22050 -w | ./mp3rtp 224.17.23.42:5004:2 -b 56 - /dev/null * arecord -b 16 -s 44100 -w | ./mp3rtp 10.1.1.42 -V2 -b128 -B256 - my_mp3file.mp3 * */ static unsigned int maxvalue(int Buffer[2][1152]) { int max = 0; int i; for (i = 0; i < 1152; i++) { if (abs(Buffer[0][i]) > max) max = abs(Buffer[0][i]); if (abs(Buffer[1][i]) > max) max = abs(Buffer[1][i]); } return max >> 16; } static void levelmessage(unsigned int maxv, int* maxx, int* tmpx) { char buff[] = "| . | . | . | . | . | . | . | . | . | . | \r"; int tmp = *tmpx, max = *maxx; buff[tmp] = '+'; tmp = (maxv * 61 + 16384) / (32767 + 16384 / 61); if (tmp > sizeof(buff) - 2) tmp = sizeof(buff) - 2; if (max < tmp) max = tmp; buff[max] = 'x'; buff[tmp] = '#'; console_printf(buff); console_flush(); *maxx = max; *tmpx = tmp; } /************************************************************************ * * main * * PURPOSE: MPEG-1,2 Layer III encoder with GPSYCHO * psychoacoustic model. * ************************************************************************/ int lame_main(lame_t gf, int argc, char **argv) { unsigned char mp3buffer[LAME_MAXMP3BUFFER]; char inPath[PATH_MAX + 1]; char outPath[PATH_MAX + 1]; int Buffer[2][1152]; int maxx = 0, tmpx = 0; int ret; int wavsamples; int mp3bytes; FILE *outf; char ip[16]; unsigned int port = 5004; unsigned int ttl = 2; char dummy; if (argc <= 2) { console_printf("Encode (via LAME) to mp3 with RTP streaming of the output\n" "\n" " mp3rtp ip[:port[:ttl]] [lame encoding options] infile outfile\n" "\n" " examples:\n" " arecord -b 16 -s 22050 -w | ./mp3rtp 224.17.23.42:5004:2 -b 56 - /dev/null\n" " arecord -b 16 -s 44100 -w | ./mp3rtp 10.1.1.42 -V2 -b128 -B256 - my_mp3file.mp3\n" "\n"); return 1; } switch (sscanf(argv[1], "%11[.0-9]:%u:%u%c", ip, &port, &ttl, &dummy)) { case 1: case 2: case 3: break; default: error_printf("Illegal destination selector '%s', must be ip[:port[:ttl]]\n", argv[1]); return -1; } rtp_initialization(); if (rtp_socket(ip, port, ttl)) { rtp_deinitialization(); error_printf("fatal error during initialization\n"); return 1; } lame_set_errorf(gf, &frontend_errorf); lame_set_debugf(gf, &frontend_debugf); lame_set_msgf(gf, &frontend_msgf); /* Remove the argumets that are rtp related, and then * parse the command line arguments, setting various flags in the * struct pointed to by 'gf'. If you want to parse your own arguments, * or call libmp3lame from a program which uses a GUI to set arguments, * skip this call and set the values of interest in the gf struct. * (see lame.h for documentation about these parameters) */ argv[1] = argv[0]; parse_args(gf, argc - 1, argv + 1, inPath, outPath, NULL, NULL); /* open the output file. Filename parsed into gf.inPath */ if (0 == strcmp(outPath, "-")) { lame_set_stream_binary_mode(outf = stdout); } else { if ((outf = lame_fopen(outPath, "wb+")) == NULL) { rtp_deinitialization(); error_printf("Could not create \"%s\".\n", outPath); return 1; } } /* open the wav/aiff/raw pcm or mp3 input file. This call will * open the file with name gf.inFile, try to parse the headers and * set gf.samplerate, gf.num_channels, gf.num_samples. * if you want to do your own file input, skip this call and set * these values yourself. */ if (init_infile(gf, inPath) < 0) { error_printf("Can't init infile '%s'\n", inPath); return 1; } /* Now that all the options are set, lame needs to analyze them and * set some more options */ ret = lame_init_params(gf); if (ret < 0) { if (ret == -1) display_bitrates(stderr); rtp_deinitialization(); error_printf("fatal error during initialization\n"); return -1; } lame_print_config(gf); /* print useful information about options being used */ if (global_ui_config.update_interval < 0.) global_ui_config.update_interval = 2.; /* encode until we hit EOF */ while ((wavsamples = get_audio(gf, Buffer)) > 0) { /* read in 'wavsamples' samples */ levelmessage(maxvalue(Buffer), &maxx, &tmpx); mp3bytes = lame_encode_buffer_int(gf, /* encode the frame */ Buffer[0], Buffer[1], wavsamples, mp3buffer, sizeof(mp3buffer)); rtp_output(mp3buffer, mp3bytes); /* write MP3 output to RTP port */ fwrite(mp3buffer, 1, mp3bytes, outf); /* write the MP3 output to file */ } mp3bytes = lame_encode_flush(gf, /* may return one or more mp3 frame */ mp3buffer, sizeof(mp3buffer)); rtp_output(mp3buffer, mp3bytes); /* write MP3 output to RTP port */ fwrite(mp3buffer, 1, mp3bytes, outf); /* write the MP3 output to file */ lame_mp3_tags_fid(gf, outf); /* add VBR tags to mp3 file */ rtp_deinitialization(); fclose(outf); close_infile(); /* close the sound input file */ return 0; } /* end of mp3rtp.c */