diff options
Diffstat (limited to 'doc/examples/encode_audio.c')
| -rw-r--r-- | doc/examples/encode_audio.c | 211 |
1 files changed, 211 insertions, 0 deletions
diff --git a/doc/examples/encode_audio.c b/doc/examples/encode_audio.c new file mode 100644 index 0000000000..cabe5893b0 --- /dev/null +++ b/doc/examples/encode_audio.c @@ -0,0 +1,211 @@ +/* + * 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 encoding with libavcodec API example. + * + * @example encode_audio.c + */ + +#include <stdint.h> +#include <stdio.h> +#include <stdlib.h> + +#include "libavcodec/avcodec.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(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(AVCodec *codec) +{ + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; +} + +/* select layout with the highest channel count */ +static int select_channel_layout(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; +} + +int main(int argc, char **argv) +{ + const char *filename; + AVCodec *codec; + AVCodecContext *c= NULL; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; + FILE *f; + uint16_t *samples; + float t, tincr; + + if (argc <= 1) { + fprintf(stderr, "Usage: %s <output file>\n", argv[0]); + return 0; + } + filename = argv[1]; + + /* register all the codecs */ + avcodec_register_all(); + + /* find the MP2 encoder */ + codec = avcodec_find_encoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + + /* put sample parameters */ + c->bit_rate = 64000; + + /* check that the encoder supports s16 pcm input */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { + fprintf(stderr, "encoder does not support %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); + } + + /* 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; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); + samples = av_malloc(buffer_size); + if (!samples) { + fprintf(stderr, "could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { + fprintf(stderr, "could not setup audio frame\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++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 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 the samples */ + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + fclose(f); + + av_freep(&samples); + av_frame_free(&frame); + avcodec_free_context(&c); +} |