本篇文章将增加AVFifoBuffer和音频样本是av_sample_fmt_is_planar的样式采样率讲解，下面上代码
< name="code" class="cpp">AVFifoBuffer * m_fifo = NULL;SwrContext * init_pcm_resample(AVFrame *in_frame, AVFrame *out_frame){ SwrContext * swr_ctx = NULL; swr_ctx = swr_alloc(); if (!swr_ctx) { printf("swr_alloc error \n"); return NULL; } AVCodecContext * audio_dec_ctx = icodec->streams[audio_stream_idx]->codec; AVSampleFormat sample_fmt; sample_fmt = (AVSampleFormat)m_dwBitsPerSample; //样本 if (audio_dec_ctx->channel_layout == 0) { audio_dec_ctx->channel_layout = av_get_default_channel_layout(icodec->streams[audio_stream_idx]->codec->channels); } /* set options */ av_opt_set_int(swr_ctx, "in_channel_layout", audio_dec_ctx->channel_layout, 0); av_opt_set_int(swr_ctx, "in_sample_rate", audio_dec_ctx->sample_rate, 0); av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", audio_dec_ctx->sample_fmt, 0); av_opt_set_int(swr_ctx, "out_channel_layout", audio_dec_ctx->channel_layout, 0); av_opt_set_int(swr_ctx, "out_sample_rate", audio_dec_ctx->sample_rate, 0); av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", sample_fmt, 0); swr_init(swr_ctx); int64_t src_nb_samples = in_frame->nb_samples; out_frame->nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx,oaudio_st->codec->sample_rate) + src_nb_samples, oaudio_st->codec->sample_rate, oaudio_st->codec->sample_rate, AV_ROUND_UP); int ret = av_samples_alloc(out_frame->data, &out_frame->linesize[0], icodec->streams[audio_stream_idx]->codec->channels, out_frame->nb_samples,oaudio_st->codec->sample_fmt,1); if (ret < 0) { return NULL; } //pcm分包初始化 int buffersize = av_samples_get_buffer_size(NULL, oaudio_st->codec->channels, 2048, oaudio_st->codec->sample_fmt, 1); m_fifo = av_fifo_alloc(buffersize); return swr_ctx;}int form_pcm_resample(SwrContext * pSwrCtx,AVFrame *in_frame, AVFrame *out_frame){ int ret = 0; if (pSwrCtx != NULL) { ret = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples, (const uint8_t**)in_frame->data, in_frame->nb_samples); if (ret < 0) { return -1; } //修改分包内存 int buffersize = av_samples_get_buffer_size(&out_frame->linesize[0], oaudio_st->codec->channels, ret, oaudio_st->codec->sample_fmt, 1); int sss = av_fifo_size(m_fifo); sss = av_fifo_realloc2(m_fifo, av_fifo_size(m_fifo) + out_frame->linesize[0]); sss = av_fifo_size(m_fifo); av_fifo_generic_write(m_fifo, out_frame->data[0], out_frame->linesize[0], NULL); out_frame->pkt_pts = in_frame->pkt_pts; out_frame->pkt_dts = in_frame->pkt_dts; //有时pkt_pts和pkt_dts不同，并且pkt_pts是编码前的dts,这里要给avframe传入pkt_dts而不能用pkt_pts //out_frame->pts = out_frame->pkt_pts; out_frame->pts = in_frame->pkt_dts; } return 0;}void uinit_pcm_resample(AVFrame * poutframe,SwrContext * swr_ctx){ if (poutframe) { avcodec_free_frame(&poutframe); poutframe = NULL; } if (swr_ctx) { swr_free(&swr_ctx); swr_ctx = NULL; } //析构pcm分包结构 if(m_fifo) { av_fifo_free(m_fifo); m_fifo = NULL; }}int perform_code(int stream_type,AVFrame * picture){ AVCodecContext *cctext = NULL; AVPacket pkt_t; av_init_packet(&pkt_t); pkt_t.data = NULL; // packet data will be allocated by the encoder pkt_t.size = 0; int frameFinished = 0 ; if (stream_type == AUDIO_ID) { cctext = oaudio_st->codec; //如果进和出的的声道，样本，采样率不同,需要重采样 if(icodec->streams[audio_stream_idx]->codec->sample_fmt != (AVSampleFormat)m_dwBitsPerSample || icodec->streams[audio_stream_idx]->codec->channels != m_dwChannelCount || icodec->streams[audio_stream_idx]->codec->sample_rate != m_dwFrequency) { int64_t pts_t = picture->pts; int duration_t = (double)cctext->frame_size * (icodec->streams[audio_stream_idx]->time_base.den /icodec->streams[audio_stream_idx]->time_base.num)/ icodec->streams[audio_stream_idx]->codec->sample_rate; int frame_bytes = cctext->frame_size * av_get_bytes_per_sample(cctext->sample_fmt)* cctext->channels; AVFrame * pFrameResample = avcodec_alloc_frame(); uint8_t * readbuff = new uint8_t[frame_bytes]; if(av_sample_fmt_is_planar(cctext->sample_fmt)) { frame_bytes /= cctext->channels; } while (av_fifo_size(m_fifo) >= frame_bytes) //取出写入的未读的包 { pFrameResample->nb_samples = cctext->frame_size; av_fifo_generic_read(m_fifo, readbuff, frame_bytes, NULL); //这里一定要考虑音频分片的问题 //如果是分片的avcodec_fill_audio_frame传入的buf是单声道的，但是buf_size 是两个声道加一起的数据量 //如果不是分片的avcodec_fill_audio_frame传入的buf是双声道的，buf_size 是两个声道加一起的数据量 if(av_sample_fmt_is_planar(cctext->sample_fmt)) { avcodec_fill_audio_frame(pFrameResample,cctext->channels,cctext->sample_fmt,readbuff,frame_bytes * cctext->channels,1); } else { avcodec_fill_audio_frame(pFrameResample,cctext->channels,cctext->sample_fmt,readbuff,frame_bytes,0); } if(m_is_first_audio_pts == 0) { m_first_audio_pts = pts_t; m_is_first_audio_pts = 1; } pFrameResample->pts = m_first_audio_pts; m_first_audio_pts += duration_t; pFrameResample->pts = av_rescale_q_rnd(pFrameResample->pts, icodec->streams[audio_stream_idx]->codec->time_base, oaudio_st->codec->time_base, AV_ROUND_NEAR_INF); nRet = avcodec_encode_audio2(cctext,&pkt_t,pFrameResample,&frameFinished); if (nRet>=0 && frameFinished) { write_frame(ocodec,AUDIO_ID,pkt_t); av_free_packet(&pkt_t); } } if (readbuff) { delete []readbuff; } if (pFrameResample) { av_free(pFrameResample); pFrameResample = NULL; } } else { nRet = avcodec_encode_audio2(cctext,&pkt_t,picture,&frameFinished); if (nRet>=0 && frameFinished) { write_frame(ocodec,AUDIO_ID,pkt_t); av_free_packet(&pkt_t); } } } else if (stream_type == VIDEO_ID) { cctext = ovideo_st->codec; if(icodec->streams[video_stream_idx]->codec->ticks_per_frame != 1) { AVRational time_base_video_t; time_base_video_t.num = icodec->streams[video_stream_idx]->codec->time_base.num; time_base_video_t.den = icodec->streams[video_stream_idx]->codec->time_base.den /icodec->streams[video_stream_idx]->codec->ticks_per_frame; picture->pts = av_rescale_q_rnd(picture->pts, time_base_video_t, ovideo_st->codec->time_base, AV_ROUND_NEAR_INF); } else { picture->pts = av_rescale_q_rnd(picture->pts, icodec->streams[video_stream_idx]->codec->time_base, ovideo_st->codec->time_base, AV_ROUND_NEAR_INF); } avcodec_encode_video2(cctext,&pkt_t,picture,&frameFinished); picture->pts++; if (frameFinished) { write_frame(ocodec,VIDEO_ID,pkt_t); av_free_packet(&pkt_t); } } return 1;}1：由于mp3的sample是1152 aac是1024 有时候将解码的mp3编码成aac时如果不做AVFifoBuffer操作，编码的aac音频sample会比原来的少很多，生成的音频会一卡一卡的明显少声音。
2：当要编码的音频样本是av_sample_fmt_is_planar分片的时候需要将解码后的视频添加到AVFrame结构体中：但是如图
不知道ffmpeg什么这么设计或者我用的不对。不过这样用是成功的。
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