流程
下面附一张使用FFmpeg编码音频的流程图。使用该流程,不仅可以编码AAC的音频,而且可以编码MP3,MP2等等各种FFmpeg支持的音频。图中蓝色背景的函数是实际输出数据的函数。浅绿色的函数是音频编码的函数。
简单介绍一下流程中各个函数的意义:
av_register_all():注册FFmpeg所有编解码器。avformat_alloc_output_context2():初始化输出码流的AVFormatContext。avio_open():打开输出文件。av_new_stream():创建输出码流的AVStream。avcodec_find_encoder():查找编码器。avcodec_open2():打开编码器。avformat_write_header():写文件头(对于某些没有文件头的封装格式,不需要此函数。比如说MPEG2TS)。avcodec_encode_audio2():编码音频。即将AVFrame(存储PCM采样数据)编码为AVPacket(存储AAC,MP3等格式的码流数据)。av_write_frame():将编码后的视频码流写入文件。av_write_trailer():写文件尾(对于某些没有文件头的封装格式,不需要此函数。比如说MPEG2TS)。
代码
#include
#include "libavutil/avutil.h"#include "libavcodec/avcodec.h"#include "libavformat/avformat.h"int flush_encoder(AVFormatContext *fmt_ctx,unsigned int stream_index){int ret;int got_frame;AVPacket enc_pkt;if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities &AV_CODEC_CAP_DELAY))return 0;while (1) {enc_pkt.data &#61; NULL;enc_pkt.size &#61; 0;av_init_packet(&enc_pkt);ret &#61; avcodec_encode_audio2 (fmt_ctx->streams[stream_index]->codec, &enc_pkt,NULL, &got_frame);av_frame_free(NULL);if (ret < 0)break;if (!got_frame){ret&#61;0;break;}printf("Flush Encoder: Succeed to encode 1 frame!\tsize:%5d\n",enc_pkt.size);ret &#61; av_write_frame(fmt_ctx, &enc_pkt);if (ret < 0)break;}return ret;
}int main(int argc, char* argv[])
{AVFormatContext* pFormatCtx;AVOutputFormat* fmt;AVStream* audio_st;AVCodecContext* pCodecCtx;AVCodec* pCodec;uint8_t* frame_buf;AVFrame* pFrame;AVPacket pkt;int got_frame&#61;0;int ret&#61;0;int size&#61;0;FILE *in_file&#61;NULL; int framenum&#61;1000; const char* out_file &#61; "tdjm.aac"; int i;in_file&#61; fopen("2000-01-01_015202.pcm", "rb");av_register_all();pFormatCtx &#61; avformat_alloc_context();fmt &#61; av_guess_format(NULL, out_file, NULL);pFormatCtx->oformat &#61; fmt;if (avio_open(&pFormatCtx->pb,out_file, AVIO_FLAG_READ_WRITE) < 0){printf("Failed to open output file!\n");return -1;}audio_st &#61; avformat_new_stream(pFormatCtx, 0);if (audio_st&#61;&#61;NULL){return -1;}pCodecCtx &#61; audio_st->codec;pCodecCtx->codec_id &#61; fmt->audio_codec;pCodecCtx->codec_type &#61; AVMEDIA_TYPE_AUDIO;pCodecCtx->sample_fmt &#61; AV_SAMPLE_FMT_FLT;pCodecCtx->sample_rate&#61; 16000;pCodecCtx->channel_layout&#61;AV_CH_LAYOUT_MONO;pCodecCtx->channels &#61; av_get_channel_layout_nb_channels(pCodecCtx->channel_layout);pCodecCtx->bit_rate &#61; 32000;av_dump_format(pFormatCtx, 0, out_file, 1);pCodec &#61; avcodec_find_encoder(pCodecCtx->codec_id);if (!pCodec){printf("Can not find encoder!\n");return -1;}if (avcodec_open2(pCodecCtx, pCodec,NULL) < 0){printf("Failed to open encoder!\n");return -1;}pFrame &#61; av_frame_alloc();pFrame->nb_samples&#61; pCodecCtx->frame_size;pFrame->format&#61; pCodecCtx->sample_fmt;size &#61; av_samples_get_buffer_size(NULL, pCodecCtx->channels,pCodecCtx->frame_size,pCodecCtx->sample_fmt, 1);frame_buf &#61; (uint8_t *)av_malloc(size);avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt,(const uint8_t*)frame_buf, size, 1);avformat_write_header(pFormatCtx,NULL);av_new_packet(&pkt,size);for (i&#61;0; i<framenum; i&#43;&#43;){if (fread(frame_buf, 1, size, in_file) <&#61; 0){printf("Failed to read raw data! \n");return -1;}else if(feof(in_file)){break;}pFrame->data[0] &#61; frame_buf; pFrame->pts&#61;i*100;got_frame&#61;0;ret &#61; avcodec_encode_audio2(pCodecCtx, &pkt,pFrame, &got_frame);if(ret < 0){printf("Failed to encode!\n");}if (got_frame&#61;&#61;1){printf("Succeed to encode 1 frame! \tsize:%5d\n",pkt.size);pkt.stream_index &#61; audio_st->index;ret &#61; av_write_frame(pFormatCtx, &pkt);av_free_packet(&pkt);}}ret &#61; flush_encoder(pFormatCtx,0);if (ret < 0) {printf("Flushing encoder failed\n");return -1;}av_write_trailer(pFormatCtx);if (audio_st){avcodec_close(audio_st->codec);av_free(pFrame);av_free(frame_buf);}avio_close(pFormatCtx->pb);avformat_free_context(pFormatCtx);fclose(in_file);return 0;
}
参考文献
1、https://blog.csdn.net/leixiaohua1020/article/details/25430449