mirror of https://github.com/FFmpeg/FFmpeg.git
The four examples (audio/video encoding/decoding) are completely independent so it makes little sense to have them all in one file.pull/258/head
Anton Khirnov
8 years ago
parent
064f19f39e
commit
40aaa8dadf
4 changed files with 215 additions and 171 deletions
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/*
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* copyright (c) 2001 Fabrice Bellard |
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* |
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* This file is part of Libav. |
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* |
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* Libav is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* Libav is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with Libav; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/**
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* @file |
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* audio encoding with libavcodec API example. |
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* |
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* @example encode_audio.c |
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*/ |
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#include <stdint.h> |
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#include <stdio.h> |
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#include <stdlib.h> |
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#include "libavcodec/avcodec.h" |
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#include "libavutil/channel_layout.h" |
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#include "libavutil/common.h" |
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#include "libavutil/frame.h" |
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#include "libavutil/samplefmt.h" |
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/* check that a given sample format is supported by the encoder */ |
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static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) |
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{ |
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const enum AVSampleFormat *p = codec->sample_fmts; |
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while (*p != AV_SAMPLE_FMT_NONE) { |
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if (*p == sample_fmt) |
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return 1; |
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p++; |
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} |
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return 0; |
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} |
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/* just pick the highest supported samplerate */ |
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static int select_sample_rate(AVCodec *codec) |
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{ |
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const int *p; |
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int best_samplerate = 0; |
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if (!codec->supported_samplerates) |
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return 44100; |
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p = codec->supported_samplerates; |
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while (*p) { |
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best_samplerate = FFMAX(*p, best_samplerate); |
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p++; |
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} |
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return best_samplerate; |
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} |
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/* select layout with the highest channel count */ |
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static int select_channel_layout(AVCodec *codec) |
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{ |
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const uint64_t *p; |
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uint64_t best_ch_layout = 0; |
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int best_nb_channels = 0; |
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if (!codec->channel_layouts) |
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return AV_CH_LAYOUT_STEREO; |
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p = codec->channel_layouts; |
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while (*p) { |
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int nb_channels = av_get_channel_layout_nb_channels(*p); |
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if (nb_channels > best_nb_channels) { |
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best_ch_layout = *p; |
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best_nb_channels = nb_channels; |
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} |
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p++; |
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} |
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return best_ch_layout; |
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} |
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int main(int argc, char **argv) |
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{ |
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const char *filename; |
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AVCodec *codec; |
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AVCodecContext *c= NULL; |
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AVFrame *frame; |
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AVPacket pkt; |
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int i, j, k, ret, got_output; |
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int buffer_size; |
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FILE *f; |
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uint16_t *samples; |
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float t, tincr; |
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if (argc <= 1) { |
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fprintf(stderr, "Usage: %s <output file>\n", argv[0]); |
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return 0; |
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} |
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filename = argv[1]; |
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/* register all the codecs */ |
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avcodec_register_all(); |
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/* find the MP2 encoder */ |
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codec = avcodec_find_encoder(AV_CODEC_ID_MP2); |
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if (!codec) { |
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fprintf(stderr, "codec not found\n"); |
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exit(1); |
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} |
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c = avcodec_alloc_context3(codec); |
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/* put sample parameters */ |
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c->bit_rate = 64000; |
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/* check that the encoder supports s16 pcm input */ |
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c->sample_fmt = AV_SAMPLE_FMT_S16; |
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if (!check_sample_fmt(codec, c->sample_fmt)) { |
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fprintf(stderr, "encoder does not support %s", |
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av_get_sample_fmt_name(c->sample_fmt)); |
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exit(1); |
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} |
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/* select other audio parameters supported by the encoder */ |
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c->sample_rate = select_sample_rate(codec); |
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c->channel_layout = select_channel_layout(codec); |
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout); |
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/* open it */ |
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if (avcodec_open2(c, codec, NULL) < 0) { |
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fprintf(stderr, "could not open codec\n"); |
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exit(1); |
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} |
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f = fopen(filename, "wb"); |
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if (!f) { |
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fprintf(stderr, "could not open %s\n", filename); |
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exit(1); |
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} |
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/* frame containing input raw audio */ |
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frame = av_frame_alloc(); |
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if (!frame) { |
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fprintf(stderr, "could not allocate audio frame\n"); |
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exit(1); |
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} |
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frame->nb_samples = c->frame_size; |
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frame->format = c->sample_fmt; |
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frame->channel_layout = c->channel_layout; |
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/* the codec gives us the frame size, in samples,
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* we calculate the size of the samples buffer in bytes */ |
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buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, |
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c->sample_fmt, 0); |
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samples = av_malloc(buffer_size); |
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if (!samples) { |
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fprintf(stderr, "could not allocate %d bytes for samples buffer\n", |
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buffer_size); |
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exit(1); |
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} |
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/* setup the data pointers in the AVFrame */ |
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ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, |
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(const uint8_t*)samples, buffer_size, 0); |
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if (ret < 0) { |
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fprintf(stderr, "could not setup audio frame\n"); |
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exit(1); |
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} |
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/* encode a single tone sound */ |
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t = 0; |
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tincr = 2 * M_PI * 440.0 / c->sample_rate; |
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for(i=0;i<200;i++) { |
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av_init_packet(&pkt); |
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pkt.data = NULL; // packet data will be allocated by the encoder
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pkt.size = 0; |
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for (j = 0; j < c->frame_size; j++) { |
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samples[2*j] = (int)(sin(t) * 10000); |
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for (k = 1; k < c->channels; k++) |
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samples[2*j + k] = samples[2*j]; |
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t += tincr; |
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} |
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/* encode the samples */ |
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ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); |
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if (ret < 0) { |
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fprintf(stderr, "error encoding audio frame\n"); |
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exit(1); |
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} |
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if (got_output) { |
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fwrite(pkt.data, 1, pkt.size, f); |
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av_packet_unref(&pkt); |
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} |
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} |
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fclose(f); |
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av_freep(&samples); |
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av_frame_free(&frame); |
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avcodec_free_context(&c); |
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} |
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