my_encode_audio.c

#include <stdint.h> #include <stdio.h> #include <stdlib.h> #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <libavutil/channel_layout.h> #include <libavutil/common.h> #include <libavutil/frame.h> #include <libavutil/samplefmt.h> /* check that a given sample format is supported by the encoder */ static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt) { const enum AVSampleFormat *p = codec->sample_fmts; while (*p != AV_SAMPLE_FMT_NONE) { if (*p == sample_fmt) return 1; p++; } return 0; } /* just pick the highest supported samplerate */ static int select_sample_rate(const AVCodec *codec) { const int *p; int best_samplerate = 0; if (!codec->supported_samplerates) return 44100; p = codec->supported_samplerates; while (*p) { if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate)) best_samplerate = *p; p++; } return best_samplerate; } /* select layout with the highest channel count */ static int select_channel_layout(const AVCodec *codec) { const uint64_t *p; uint64_t best_ch_layout = 0; int best_nb_channels = 0; if (!codec->channel_layouts) return AV_CH_LAYOUT_STEREO; p = codec->channel_layouts; while (*p) { int nb_channels = av_get_channel_layout_nb_channels(*p); if (nb_channels > best_nb_channels) { best_ch_layout = *p; best_nb_channels = nb_channels; } p++; } return best_ch_layout; } static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt, FILE *output) { int ret; /* send the frame for encoding */ ret = avcodec_send_frame(ctx, frame); if (ret < 0) { fprintf(stderr, "Error sending the frame to the encoder\n"); exit(1); } /* read all the available output packets (in general there may be any * number of them */ while (ret >= 0) { ret = avcodec_receive_packet(ctx, pkt); if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) return; else if (ret < 0) { fprintf(stderr, "Error encoding audio frame\n"); exit(1); } fwrite(pkt->data, 1, pkt->size, output); av_packet_unref(pkt); } } int main(int argc, char **argv) { const char *filename; const AVCodec *codec; AVCodecContext *c= NULL; AVFrame *frame; AVPacket *pkt; int i, j, k, ret; FILE *f; uint16_t *samples; float t, tincr; av_register_all(); avcodec_register_all(); if (argc <= 1) { fprintf(stderr, "Usage: %s <output file>\n", argv[0]); return 0; } filename = argv[1]; /* find the MP2 encoder */ codec = avcodec_find_encoder(AV_CODEC_ID_MP3); if (!codec) { fprintf(stderr, "Codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); if (!c) { fprintf(stderr, "Could not allocate audio codec context\n"); exit(1); } /* put sample parameters */ c->bit_rate = 64000; /* check that the encoder supports s16 pcm input */ c->sample_fmt = AV_SAMPLE_FMT_S16P; if (!check_sample_fmt(codec, c->sample_fmt)) { fprintf(stderr, "Encoder does not support sample format %s", av_get_sample_fmt_name(c->sample_fmt)); exit(1); } /* select other audio parameters supported by the encoder */ c->sample_rate = select_sample_rate(codec); c->channel_layout = select_channel_layout(codec); c->channels = av_get_channel_layout_nb_channels(c->channel_layout); /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { fprintf(stderr, "Could not open codec\n"); exit(1); } f = fopen(filename, "wb"); if (!f) { fprintf(stderr, "Could not open %s\n", filename); exit(1); } /* packet for holding encoded output */ pkt = av_packet_alloc(); if (!pkt) { fprintf(stderr, "could not allocate the packet\n"); exit(1); } /* frame containing input raw audio */ frame = av_frame_alloc(); if (!frame) { fprintf(stderr, "Could not allocate audio frame\n"); exit(1); } frame->nb_samples = c->frame_size; frame->format = c->sample_fmt; frame->channel_layout = c->channel_layout; /* allocate the data buffers */ ret = av_frame_get_buffer(frame, 0); if (ret < 0) { fprintf(stderr, "Could not allocate audio data buffers\n"); exit(1); } /* encode a single tone sound */ t = 0; tincr = 2 * M_PI * 440.0 / c->sample_rate; for (i = 0; i < 200; i++) { /* make sure the frame is writable -- makes a copy if the encoder * kept a reference internally */ ret = av_frame_make_writable(frame); if (ret < 0) exit(1); samples = (uint16_t*)frame->data[0]; for (j = 0; j < c->frame_size; j++) { samples[2*j] = (int)(sin(t) * 10000); for (k = 1; k < c->channels; k++) samples[2*j + k] = samples[2*j]; t += tincr; } encode(c, frame, pkt, f); } /* flush the encoder */ encode(c, NULL, pkt, f); fclose(f); av_frame_free(&frame); av_packet_free(&pkt); avcodec_free_context(&c); return 0; }
the above file is for encoding the decoded audio file(binary file).

to compile: gcc -g -o my_encode_audio my_encode_audio.c -lavutil -lavformat -lavcodec -lswresample -lz -lm
to run: ./my_encode_audio raw.bin

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