FFmpeg音频编码实例

/************************************************************************ * Copyright(c) 2012 Yang Xian * All rights reserved. * * File: AudioEncode.cpp * Brief: 音频的编码 * Version: 1.0 * Author: Yang Xian * Email: yang_xian521@163.com * Date: 2012/06/28 * History: ************************************************************************/ #include <stdlib.h> #include <stdio.h> extern "C" { #include <libavformat/avformat.h> #include <libswscale/swscale.h> }; void main() { int16_t *samples; uint8_t *audio_outbuf; int audio_outbuf_size; int audio_input_frame_size; double audio_pts; const char* filename = "test.wav"; FILE *fin = fopen("result.pcm", "rb"); //音频源文件 AVOutputFormat *fmt; AVFormatContext *oc; AVStream * audio_st; av_register_all(); fmt = guess_format(NULL, filename, NULL); oc = av_alloc_format_context(); oc->oformat = fmt; snprintf(oc->filename, sizeof(oc->filename), "%s", filename); audio_st = NULL; if (fmt->audio_codec != CODEC_ID_NONE) { AVCodecContext *c; audio_st = av_new_stream(oc, 1); c = audio_st->codec; c->codec_id = fmt->audio_codec; c->codec_type = CODEC_TYPE_AUDIO; c->bit_rate = 128000; c->sample_rate = 44100; c->channels = 2; } if (av_set_parameters(oc, NULL) < 0) { return; } dump_format(oc, 0, filename, 1); if (audio_st) { AVCodecContext* c; AVCodec* codec; c = audio_st->codec; codec = avcodec_find_encoder(c->codec_id); avcodec_open(c, codec); audio_outbuf_size = 10000; audio_outbuf = (uint8_t*)av_malloc(audio_outbuf_size); if (c->frame_size <= 1) { audio_input_frame_size = audio_outbuf_size / c->channels; switch (audio_st->codec->codec_id) { case CODEC_ID_PCM_S16LE: case CODEC_ID_PCM_S16BE: case CODEC_ID_PCM_U16LE: case CODEC_ID_PCM_U16BE: audio_input_frame_size >>= 1; break; default: break; } } else { audio_input_frame_size = c->frame_size; } samples = (int16_t*)av_malloc(audio_input_frame_size*2*c->channels); } if (!fmt->flags & AVFMT_NOFILE) { if (url_fopen(&oc->pb, filename, URL_WRONLY) < 0) { return; } } av_write_header(oc); for (;;) { if (audio_st) { audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den; } else { audio_pts = 0.0; } if (!audio_st || audio_pts >= 360.0) { break; } if (fread(samples, 1, audio_input_frame_size*2*audio_st->codec->channels, fin) <= 0) { break; } AVCodecContext* c; AVPacket pkt; av_init_packet(&pkt); c = audio_st->codec; pkt.size = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples); pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, audio_st->time_base); pkt.flags |= PKT_FLAG_KEY; pkt.stream_index = audio_st->index; pkt.data = audio_outbuf; if (av_write_frame(oc, &pkt) != 0) { return; } } if (audio_st) { avcodec_close(audio_st->codec); av_free(samples); av_free(audio_outbuf); } av_write_trailer(oc); for (int i=0; i<oc->nb_streams; i++) { av_freep(&oc->streams[i]->codec); av_freep(&oc->streams[i]); } if (!(fmt->flags & AVFMT_NOFILE)) { url_fclose(oc->pb); } av_free(oc); fclose(fin); }