我正在尝试创建一个将音频文件和视频文件组合在一起并将其输出到mp4的功能。我成功地做到了这一点，只是输出的帧速率不正确。这与原版有很小的不同。 30.13而它应该是30。当我将这些文件与ffmpeg程序结合使用时，结果应该是30。
我确信它在收到无序数据时与dts / pts校正有关，但ffmpeg程序也以类似的方式做到这一点。所以我不确定从哪里开始。我查看了ffmpeg源代码并复制了一些他们的dts校正，但仍然没有运气。我在这做错了什么？
bool mux_audio_video(const char* audio_filename, const char* video_filename, const char* output_filename){
av_register_all();
AVOutputFormat* out_format = NULL;
AVFormatContext* audio_context = NULL, *video_context = NULL, *output_context = NULL;
int video_index_in = -1, audio_index_in = -1;
int video_index_out = -1, audio_index_out = -1;
if(avformat_open_input(&audio_context, audio_filename, 0, 0) < 0)
return false;
if(avformat_find_stream_info(audio_context, 0) < 0){
avformat_close_input(&audio_context);
return false;
}
if(avformat_open_input(&video_context, video_filename, 0, 0) < 0){
avformat_close_input(&audio_context);
return false;
}
if(avformat_find_stream_info(video_context, 0) < 0){
avformat_close_input(&audio_context);
avformat_close_input(&video_context);
return false;
}
if(avformat_alloc_output_context2(&output_context, av_guess_format("mp4", NULL, NULL), NULL, output_filename) < 0){
avformat_close_input(&audio_context);
avformat_close_input(&video_context);
return false;
}
out_format = output_context->oformat;
//find first audio stream in the audio file input
for(size_t i = 0;i < audio_context->nb_streams;++i){
if(audio_context->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO){
audio_index_in = i;
AVStream* in_stream = audio_context->streams[i];
AVCodec* codec = avcodec_find_encoder(in_stream->codecpar->codec_id);
AVCodecContext* tmp = avcodec_alloc_context3(codec);
avcodec_parameters_to_context(tmp, in_stream->codecpar);
AVStream* out_stream = avformat_new_stream(output_context, codec);
audio_index_out = out_stream->index;
if(output_context->oformat->flags & AVFMT_GLOBALHEADER){
tmp->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
tmp->codec_tag = 0;
avcodec_parameters_from_context(out_stream->codecpar, tmp);
avcodec_free_context(&tmp);
break;
}
}
//find first video stream in the video file input
for(size_t i = 0;i < video_context->nb_streams;++i){
if(video_context->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO){
video_index_in = i;
AVStream* in_stream = video_context->streams[i];
AVCodec* codec = avcodec_find_encoder(in_stream->codecpar->codec_id);
AVCodecContext* tmp = avcodec_alloc_context3(codec);
avcodec_parameters_to_context(tmp, in_stream->codecpar);
AVStream* out_stream = avformat_new_stream(output_context, codec);
video_index_out = out_stream->index;
if(output_context->oformat->flags & AVFMT_GLOBALHEADER){
tmp->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
tmp->codec_tag = 0;
avcodec_parameters_from_context(out_stream->codecpar, tmp);
avcodec_free_context(&tmp);
break;
}
}
//setup output
if(!(out_format->flags & AVFMT_NOFILE)){
if(avio_open(&output_context->pb, output_filename, AVIO_FLAG_WRITE) < 0){
avformat_free_context(output_context);
avformat_close_input(&audio_context);
avformat_close_input(&video_context);
return false;
}
}
if(avformat_write_header(output_context, NULL) < 0){
if(!(out_format->flags & AVFMT_NOFILE)){
avio_close(output_context->pb);
}
avformat_free_context(output_context);
avformat_close_input(&audio_context);
avformat_close_input(&video_context);
return false;
}
int64_t video_pts = 0, audio_pts = 0;
int64_t last_video_dts = 0, last_audio_dts = 0;
while(true){
AVPacket packet;
av_init_packet(&packet);
packet.data = NULL;
packet.size = 0;
int64_t* last_dts;
AVFormatContext* in_context;
int stream_index = 0;
AVStream* in_stream, *out_stream;
//Read in a frame from the next stream
if(av_compare_ts(video_pts, video_context->streams[video_index_in]->time_base,
audio_pts, audio_context->streams[audio_index_in]->time_base) <= 0)
{
//video
last_dts = &last_video_dts;
in_context = video_context;
stream_index = video_index_out;
if(av_read_frame(in_context, &packet) >= 0){
do{
if(packet.stream_index == video_index_in){
video_pts = packet.pts;
break;
}
av_packet_unref(&packet);
}while(av_read_frame(in_context, &packet) >= 0);
}else{
break;
}
}else{
//audio
last_dts = &last_audio_dts;
in_context = audio_context;
stream_index = audio_index_out;
if(av_read_frame(in_context, &packet) >= 0){
do{
if(packet.stream_index == audio_index_in){
audio_pts = packet.pts;
break;
}
av_packet_unref(&packet);
}while(av_read_frame(in_context, &packet) >= 0);
}else{
break;
}
}
in_stream = in_context->streams[packet.stream_index];
out_stream = output_context->streams[stream_index];
av_packet_rescale_ts(&packet, in_stream->time_base, out_stream->time_base);
//if dts is out of order, ffmpeg throws an error. So manually fix. Similar to what ffmpeg does in ffmpeg.c
if(packet.dts < (*last_dts + !(output_context->oformat->flags & AVFMT_TS_NONSTRICT)) && packet.dts != AV_NOPTS_VALUE && (*last_dts) != AV_NOPTS_VALUE){
int64_t next_dts = (*last_dts)+1;
if(packet.pts >= packet.dts && packet.pts != AV_NOPTS_VALUE){
packet.pts = FFMAX(packet.pts, next_dts);
}
if(packet.pts == AV_NOPTS_VALUE){
packet.pts = next_dts;
}
packet.dts = next_dts;
}
(*last_dts) = packet.dts;
packet.pos = -1;
packet.stream_index = stream_index;
//output packet
if(av_interleaved_write_frame(output_context, &packet) < 0){
break;
}
av_packet_unref(&packet);
}
av_write_trailer(output_context);
//cleanup
if(!(out_format->flags & AVFMT_NOFILE)){
avio_close(output_context->pb);
}
avformat_free_context(output_context);
avformat_close_input(&audio_context);
avformat_close_input(&video_context);
return true;
}
我发现了这个问题。我只需要将last_video_dts和last_audio_dts初始化为int64_t的最小值而不是0。
int64_t last_video_dts, last_audio_dts;
last_video_dts = last_audio_dts = std::numeric_limits<int64_t>::lowest();
现在输出基本上与ffmpeg程序的输出相同。
编辑：
正如kamilz所提到的，使用AV_NOPTS_VALUE更好，更便携。
int64_t last_video_dts, last_audio_dts;
last_video_dts = last_audio_dts = AV_NOPTS_VALUE;