ffmpeg重采样resample包含AVAudioFifo
Posted 朱韦刚
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ffmpeg重采样resample包含AVAudioFifo(三)
本篇文章记录ffmpeg重采样和AVAudioFifo一起用的实现,如果看完整的流程请看这边文章:http://blog.csdn.net/zhuweigangzwg/article/details/72481966,本文只讲重采样和AVAudioFifo的用法,具体看代码注释。
SwrContext * ffmpeg_init_pcm_resample(Out_stream_info * out_stream_info,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 = m_icodec->streams[m_in_audio_stream_idx]->codec;
AVSampleFormat sample_fmt;
sample_fmt = (AVSampleFormat)out_stream_info->m_dwBitsPerSample; //样本
int out_channel_layout = av_get_default_channel_layout(out_stream_info->m_dwChannelCount);
if (audio_dec_ctx->channel_layout == 0)
{
audio_dec_ctx->channel_layout = av_get_default_channel_layout(m_icodec->streams[m_in_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", out_channel_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", out_stream_info->m_dwFrequency, 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;
//计算输出的samples 和采样率有关 例如:48000转44100,samples则是从1152转为1059,除法
out_frame->nb_samples = av_rescale_rnd(src_nb_samples, out_stream_info->m_dwFrequency, audio_dec_ctx->sample_rate, AV_ROUND_UP);
int ret = av_samples_alloc(out_frame->data, &out_frame->linesize[0],
out_stream_info->m_dwChannelCount, out_frame->nb_samples,out_stream_info->m_oaudio_st->codec->sample_fmt,1);
if (ret < 0)
{
return NULL;
}
out_stream_info->m_audiofifo = av_audio_fifo_alloc(out_stream_info->m_oaudio_st->codec->sample_fmt, out_stream_info->m_oaudio_st->codec->channels,
out_frame->nb_samples);
return swr_ctx;
}
int ffmpeg_preform_pcm_resample(Out_stream_info * out_stream_info,SwrContext * pSwrCtx,AVFrame *in_frame, AVFrame *out_frame)
{
int ret = 0;
int samples_out_per_size = 0; //转换之后的samples大小
if (pSwrCtx != NULL)
{
//这里注意下samples_out_per_size这个值和 out_frame->nb_samples这个值有时候不一样,ffmpeg里面做了策略不是问题。
samples_out_per_size = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples,
(const uint8_t**)in_frame->data, in_frame->nb_samples);
if (samples_out_per_size < 0)
{
return -1;
}
AVCodecContext * audio_dec_ctx = m_icodec->streams[m_in_audio_stream_idx]->codec;
int buffersize_in = av_samples_get_buffer_size(&in_frame->linesize[0],audio_dec_ctx->channels,
in_frame->nb_samples, audio_dec_ctx->sample_fmt, 1);
//修改分包内存
int buffersize_out = av_samples_get_buffer_size(&out_frame->linesize[0], out_stream_info->m_oaudio_st->codec->channels,
samples_out_per_size, out_stream_info->m_oaudio_st->codec->sample_fmt, 1);
int fifo_size = av_audio_fifo_size(out_stream_info->m_audiofifo);
fifo_size = av_audio_fifo_realloc(out_stream_info->m_audiofifo, av_audio_fifo_size(out_stream_info->m_audiofifo) + out_frame->nb_samples);
av_audio_fifo_write(out_stream_info->m_audiofifo,(void **)out_frame->data,samples_out_per_size);
fifo_size = av_audio_fifo_size(out_stream_info->m_audiofifo);
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;
//测试用
if (out_stream_info->user_stream_id ==11)
{
if (pcm_file == NULL)
{
pcm_file = fopen("11.pcm","wb");
}
int wtiresize = fwrite(out_frame->data[0],buffersize_out,1, pcm_file);
fflush(pcm_file);
}
}
ret = 1;
return ret;
}
void ffmpeg_uinit_pcm_resample(SwrContext * swr_ctx,AVAudioFifo * audiofifo)
{
if (swr_ctx)
{
swr_free(&swr_ctx);
swr_ctx = NULL;
}
if(audiofifo)
{
av_audio_fifo_free(audiofifo);
audiofifo = NULL;
}
}
存储的声音波形pcm如下:
运行后转码文件如下效果:
如有错误请指正:
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