ffmpeg 保存数据流到文件

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我是靠谱客的博主可靠背包，最近开发中收集的这篇文章主要介绍ffmpeg 保存数据流到文件，觉得挺不错的，现在分享给大家，希望可以做个参考。

概述

av_register_all();
	
std::string filename = "d6.mp4";
AVFormatContext *outContext = NULL;
avformat_alloc_output_context2(&outContext, NULL, "mp4", filename.c_str());

AVStream *stream = avformat_new_stream(outContext, NULL);
stream->codec->codec_id = AV_CODEC_ID_H264;
stream->codec->codec_type = AVMEDIA_TYPE_VIDEO;
//stream->codec->pix_fmt = AV_PIX_FMT_YUV420P;
stream->codec->width = 1280;
stream->codec->height = 720;
//stream->codec->bit_rate = context->bit_rate;
stream->codec->time_base.den = 25;
stream->codec->time_base.num = 1;

stream->nb_frames = 1050;
stream->codec->qmin = 10;
stream->codec->qmax = 51;
stream->codec->qcompress = 0.6;

//stream->time_base.den = 25;
//stream->time_base.num = 1;

AVStream *audioStream = avformat_new_stream(outContext, NULL);
if (outContext->oformat->flags & AVFMT_GLOBALHEADER)
{
	stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
	audioStream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}

audioStream->codec->codec_id = AV_CODEC_ID_AAC;
audioStream->codec->codec_type = AVMEDIA_TYPE_AUDIO;
audioStream->codec->sample_rate = 16000;
audioStream->codec->channels = 1;
audioStream->codec->bits_per_raw_sample = 16;
audioStream->codec->sample_fmt = AV_SAMPLE_FMT_S16;
audioStream->codec->time_base.den = 30;
audioStream->codec->time_base.num = 1;
audioStream->codec->frame_size = 1024;//av_samples_get_buffer_size(NULL, 
//	audioStream->codec->channels, 1024, audioStream->codec->sample_fmt, 1);;
//audioStream->time_base.den = 30;
//audioStream->time_base.num = 1;
AVBitStreamFilterContext* aacbsfc = av_bitstream_filter_init("aac_adtstoasc");

AVBitStreamFilterContext* h264bsfc = av_bitstream_filter_init("h264_mp4toannexb");
	
avio_open(&outContext->pb, filename.c_str(), AVIO_FLAG_WRITE);

av_dump_format(outContext, 0, NULL, 1);

AVDictionary* opt = NULL;
//av_dict_set_int(&opt, "video_track_timescale", 25, 0);
avformat_write_header(outContext, &opt);

MediaFormat informat;
informat.m_audioCodec = IMS_FRAME_G711A;
informat.m_audioChannel = 1;
informat.m_audioSampleRate = 16000;
informat.m_audioPropSize = 16;
MediaFormat outformat;
outformat.m_audioCodec = IMS_FRAME_AAC_ADTS;
outformat.m_audioChannel = 1;
outformat.m_audioSampleRate = 16000;
outformat.m_audioPropSize = 16;
AudioConvert *convert = new AudioConvert();
convert->init(informat, outformat);


while ((!feof(fr)) && (len == sizeof(info)))
{

	.........

	if (视频帧)
	{
		AVPacket pkt;
		av_init_packet(&pkt);
		pkt.data = (unsigned char *)(pBuf);
		pkt.size = info.len;
		pkt.stream_index = stream->index;
		pkt.pts = stream->pts.val * stream->time_base.num / stream->time_base.den;// info.timestamp / 25;
		//int64_t calc_duration = (double)AV_TIME_BASE / av_q2d(stream->time_base);
		//pkt.dts = pkt.pts = (double)(count*calc_duration) / (double)(av_q2d(stream->time_base)*AV_TIME_BASE);
		int type = pBuf[4];
		//printf("pts:%.2X, %dn", type,  type & 0x1F);
		if (info.flag == 0x22)
		{
			pkt.flags |= AV_PKT_FLAG_KEY;
		}
		if (stream->codec->extradata == NULL && info.flag == 0x22)
		{
			int i = 0;
			char *pbuffer = pBuf;
			for (;i < info.len - 4; i++)
			{
				UINT32 data = *((uint32_t*)pbuffer);
				if (data == 0x01000000 && *(pbuffer + 5) == 0x65)
				{
					break;
				}
			}
			stream->codec->extradata_size = i;
			stream->codec->extradata = new uint8_t[i + FF_INPUT_BUFFER_PADDING_SIZE];
			memcpy(stream->codec->extradata, pBuf, i);
			memset(stream->codec->extradata + i, 0, FF_INPUT_BUFFER_PADDING_SIZE);
		}
		//av_bitstream_filter_filter(h264bsfc, stream->codec, NULL, &pkt.data, &pkt.size, pkt.data, pkt.size, 0);
		if (av_interleaved_write_frame(outContext, &pkt) < 0)
			printf("write frame error");
		count++;
		Sleep(20);
		//fwrite(pBuf, info.len, 1, fv);
	}
	else
	{
		//fwrite(pBuf, info.len, 1, fa);
		if (convert)
		{
			outlen = 0;
			convert->convert(pBuf, info.len, buffer, outlen);
			
			
		}
		if (outlen > 0)
		{
			AVPacket pkt;
			av_init_packet(&pkt);
			pkt.data = (unsigned char *)(buffer);
			pkt.size = outlen;
			pkt.stream_index = audioStream->index;
			pkt.pts = audioStream->pts.val * audioStream->time_base.num / audioStream->time_base.den;
			av_bitstream_filter_filter(aacbsfc, audioStream->codec, NULL, &pkt.data, &pkt.size, pkt.data, pkt.size, 0);
			if (av_interleaved_write_frame(outContext, &pkt) < 0)
				printf("write frame error");
		}
		
	}

	delete[]pBuf;

	......

}

......

av_write_trailer(outContext);

avcodec_close(outContext->streams[0]->codec);
av_freep(&outContext->streams[0]->codec);
av_freep(&outContext->streams[0]);

avcodec_close(outContext->streams[1]->codec);
av_freep(&outContext->streams[1]->codec);
av_freep(&outContext->streams[1]);

avio_close(outContext->pb);
av_free(outContext);