http://blog.csdn.net/relar/article/details/21785673
在ffmpeg的官方例子中有一个muxing.c,这个例子是演示如何用ffmpeg进行打包(muxing),但是这个例子有些问题,说好听点是不完善,说不好听就是有错误。ffmpeg.c是非常完善的,对比ffmpeg.c我发现主要有以下两个错误:
1、在使用avcodec_encode_audio2/avcodec_encode_video2编码前,没有给定时间戳。
2、在main函数的for循环之后,没有flush,也就是还有一些延迟的帧在缓冲中,没有写进输出文件。在编码时,并不是每一个输入帧立即编码得到输出帧,而往往是输入N多帧之后才开始输出帧,我见过最多输入60帧之后才出现第一个输入帧的,那么就出现了一个问题,以输入为循环体,输入结束循环也结束,那么就还有一些帧在缓存中,此时我们需要将其拿出来,编码,再写进输出文件。
下面我以音频为例修改了代码,首先是函数write_audio_frame
- static void write_audio_frame(AVFormatContext *oc, AVStream *st)
- {
- AVCodecContext *c;
- AVPacket pkt = { 0 }; // data and size must be 0;
- AVFrame *frame = NULL;
- int got_packet, ret, dst_nb_samples;
- AVRational r = {1, AV_TIME_BASE};
- av_init_packet(&pkt);
- pkt.data = NULL;
- pkt.size = 0;
- c = st->codec;
- if(!frame && !(frame = avcodec_alloc_frame()))
- return ;
- else
- avcodec_get_frame_defaults(frame);
- get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
- /* convert samples from native format to destination codec format, using the resampler */
- if (swr_ctx) {
- /* compute destination number of samples */
- dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
- c->sample_rate, c->sample_rate, AV_ROUND_UP);
- if (dst_nb_samples > max_dst_nb_samples) {
- av_free(dst_samples_data[0]);
- ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
- dst_nb_samples, c->sample_fmt, 0);
- if (ret < 0)
- exit(1);
- max_dst_nb_samples = dst_nb_samples;
- dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
- c->sample_fmt, 0);
- }
- /* convert to destination format */
- ret = swr_convert(swr_ctx,
- dst_samples_data, dst_nb_samples,
- (const uint8_t **)src_samples_data, src_nb_samples);
- if (ret < 0) {
- fprintf(stderr, "Error while converting\n");
- exit(1);
- }
- } else {
- dst_samples_data[0] = src_samples_data[0];
- dst_nb_samples = src_nb_samples;
- }
- frame->nb_samples = dst_nb_samples;
- avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, dst_samples_data[0], dst_samples_size, 0);
- //下面两句我加的。编码前一定要给frame时间戳
- frame->pts = lastpts;
- lastpts = frame->pts + frame->nb_samples;
- ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet); //如果没有前两句,编码之后的pts是无效的
- if (ret < 0) {
- fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
- exit(1);
- }
- if (!got_packet)
- return;
- pkt.stream_index = st->index;
- //下面两句我加的,加了才不是提示“encoder did not produce proper pts, make some up”错误
- pkt.pts = av_rescale_q(pkt.pts, st->codec->time_base, st->time_base);//
- pkt.dts = av_rescale_q(pkt.dts, st->codec->time_base, st->time_base);//
- pkt.duration = av_rescale_q(pkt.duration, st->codec->time_base, st->time_base);
- /* Write the compressed frame to the media file. */
- ret = av_interleaved_write_frame(oc, &pkt);
- if (ret != 0) {
- fprintf(stderr, "Error while writing audio frame: %s\n",
- av_err2str(ret));
- exit(1);
- }
- avcodec_free_frame(&frame);
- }
- //下面几行是获得delay帧
- c = audio_st->codec;
- for(got_output=1; got_output>0; i++)
- {
- av_init_packet(&pkt);
- pkt.data = NULL;
- pkt.size = 0;
- ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
- if (ret < 0)
- {
- fprintf(stderr, "Error encoding frame\n");
- exit(1);
- }
- if (got_output)
- {
- // audio_st->pts.val += 1024;//av_rescale_q(1, audio_st->codec->time_base, audio_st->time_base);
- pkt.pts = av_rescale_q(pkt.pts, c->time_base, audio_st->time_base);//audio_st->pts.val;
- pkt.dts = av_rescale_q(pkt.dts, c->time_base, audio_st->time_base);//audio_st->pts.val;
- pkt.duration = av_rescale_q(pkt.duration, c->time_base, audio_st->time_base);
- ret = av_interleaved_write_frame(oc, &pkt);
- av_free_packet(&pkt);
- }
- }
这里仅仅修改了音频方面的处理,视频方面可以家有完全参考以上代码。另外我还将原来由程序生成的原始图像,改为了从BMP文件读取画面,成功。有兴趣的朋友可以交流。