/* * copyright (c) 2001 Fabrice Bellard * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * audio decoding with libavcodec API example * * @example decode_audio.c */ #include #include #include #include "libavutil/channel_layout.h" #include "libavutil/frame.h" #include "libavutil/mem.h" #include "libavcodec/avcodec.h" #define AUDIO_INBUF_SIZE 20480 #define AUDIO_REFILL_THRESH 4096 static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile) { int16_t *interleave_buf; int ret, data_size, i; /* send the packet with the compressed data to the decoder */ ret = avcodec_send_packet(dec_ctx, pkt); if (ret < 0) { fprintf(stderr, "Error submitting the packet to the decoder\n"); exit(1); } /* read all the output frames (in general there may be any number of them */ while (ret >= 0) { ret = avcodec_receive_frame(dec_ctx, frame); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) return; else if (ret < 0) { fprintf(stderr, "Error during decoding\n"); exit(1); } /* the stream parameters may change at any time, check that they are * what we expect */ if (av_get_channel_layout_nb_channels(frame->channel_layout) != 2 || frame->format != AV_SAMPLE_FMT_S16P) { fprintf(stderr, "Unsupported frame parameters\n"); exit(1); } /* The decoded data is signed 16-bit planar -- each channel in its own * buffer. We interleave the two channels manually here, but using * libavresample is recommended instead. */ data_size = sizeof(*interleave_buf) * 2 * frame->nb_samples; interleave_buf = av_malloc(data_size); if (!interleave_buf) exit(1); for (i = 0; i < frame->nb_samples; i++) { interleave_buf[2 * i] = ((int16_t*)frame->data[0])[i]; interleave_buf[2 * i + 1] = ((int16_t*)frame->data[1])[i]; } fwrite(interleave_buf, 1, data_size, outfile); av_freep(&interleave_buf); } } int main(int argc, char **argv) { const char *outfilename, *filename; const AVCodec *codec; AVCodecContext *c= NULL; AVCodecParserContext *parser = NULL; int len, ret; FILE *f, *outfile; uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; uint8_t *data; size_t data_size; AVPacket *pkt; AVFrame *decoded_frame = NULL; if (argc <= 2) { fprintf(stderr, "Usage: %s \n", argv[0]); exit(0); } filename = argv[1]; outfilename = argv[2]; /* register all the codecs */ avcodec_register_all(); pkt = av_packet_alloc(); /* find the MPEG audio decoder */ codec = avcodec_find_decoder(AV_CODEC_ID_MP2); if (!codec) { fprintf(stderr, "codec not found\n"); exit(1); } parser = av_parser_init(codec->id); if (!parser) { fprintf(stderr, "parser not found\n"); exit(1); } c = avcodec_alloc_context3(codec); /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { fprintf(stderr, "could not open codec\n"); exit(1); } f = fopen(filename, "rb"); if (!f) { fprintf(stderr, "could not open %s\n", filename); exit(1); } outfile = fopen(outfilename, "wb"); if (!outfile) { av_free(c); exit(1); } /* decode until eof */ data = inbuf; data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); while (data_size > 0) { if (!decoded_frame) { if (!(decoded_frame = av_frame_alloc())) { fprintf(stderr, "out of memory\n"); exit(1); } } ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size, data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0); if (ret < 0) { fprintf(stderr, "Error while parsing\n"); exit(1); } data += ret; data_size -= ret; if (pkt->size) decode(c, pkt, decoded_frame, outfile); if (data_size < AUDIO_REFILL_THRESH) { memmove(inbuf, data, data_size); data = inbuf; len = fread(data + data_size, 1, AUDIO_INBUF_SIZE - data_size, f); if (len > 0) data_size += len; } } fclose(outfile); fclose(f); avcodec_free_context(&c); av_parser_close(parser); av_frame_free(&decoded_frame); av_packet_free(&pkt); return 0; }