I encode h264 data by libavcodec.
ex.
while (1) {
...
avcodec_encode_video(pEnc->pCtx, OutBuf, ENC_OUTSIZE, pEnc->pYUVFrame);
...
}
If I directly save OutBuf data as a .264 file, it can`t be play by player. Now I want to save OutBuf
as a mp4 file. Anyone know how to do this by ffmpeg lib? thanks.
You use avformat_write_header, av_interleaved_write_frame, avformat_write_trailer and friends.
Their usage is shown in the muxing example of FFmpeg.
See a similar topic: Raw H264 frames in mpegts container using libavcodec with also writing to a file (different container, same API)
See also links from answer here: FFMpeg encoding RGB images to H264
Related
I'm using ffmpeg library for live streaming via RTMP. I want to know how to give my choice of audio and video codec for the particular format in avformat_alloc_output_context2.
In Detail:
The following command works perfectly for me.
ffmpeg -re -stream_loop -1 -i ~/Downloads/Microsoft_Surface.mp4 -vcodec copy -c:a aac -b:a 160k -ar 44100 -strict -2 -f flv -flvflags no_duration_filesize rtmp://192.168.1.7/live/surface
In the output, I have set my audio codec to be aac and copied the video codec from input, which is H264.
I want to emulate this in the library, but don't know how to.
avformat_alloc_output_context2(&_ctx, NULL, "flv", NULL);
Above code sets oformat audio codec to ADPCM_SWF and video codec to FLV1. How to change that to AAC and H264 ?
So far, used av_guess_format to construct AVOutputFormat. It accepts only format as input. And I don't know where to mention audio and video codec.
AVOutputFormat* output_format = av_guess_format("flv", NULL, NULL);
Also tried giving filename to avformat_alloc_output_context2 with the rest of the parameters NULL.
AVOutputFormat* output_format = av_guess_format(NULL, "flv_acc_sample.flv", NULL);
This file has AAC audio and H264 video. But still ffmpeg loads oformat with ADPCM_SWF audio and FLV1 video codecs.
Searched stackoverflow for similar questions, but could not find the solution I was looking for.
Any hint/guidance is hugely appreciated. Thank you.
I wrote code to create H.264 stream, which has a loop to generate H.264 encoded frame.
while(true) {
...
x264_encoder_encode(encoder, &buffer, &i_buffer, &pic_in, &pic_out);
...
/*TODO: Write one frame in the buffer to a streamable mp4 file*/
}
Every single time, an H.264 encoded frame is generated and stored in the buffer. How can I write it into a streamable mp4 file directly through the buffer?
I spent lots of time searching for the solution. All I can find is to read stream from a file using
avformat_open_input(&fmtCtx, in_filename, 0, 0)
Is there any way to read directly from buffer without a file?
MP4 is actually not streamable. So in other words, you can't do it at all. I ran in that very problem.
The reason why it won't work is because when you open an mp4 file, you have to have all sorts of parameters, which by default get saved at the end of the file. When you create an MP4, you can always forcibly save that info at the start. However, to know what those parameters are, you need all the data. And without those parameters, the software trying to load the mp4 fails very early on. This is true for some other formats such as webm videos and .m4a or .wav for audio.
What you have to do is stream the actual H.264, possibly using RTSP or a format of your own if you're in control of both sides.
I need to encode an mpegts video using the ffmpeg C++ API. The output video shall have two streams: the first one shall be of type AVMEDIA_TYPE_VIDEO; the second one shall be of type AVMEDIA_TYPE_DATA and shall contain a set of KLV data.
I have written my own KLV library to manage the KLV format.
However I'm not able to create "from scratch" a new video by combining the two streams. Following the implementation as in FFMPEG C api h.264 encoding / MPEG2 ts streaming problems I can successfully encode a mpegts video with a single video stream.
However I'm not able to add a new AVMEDIA_TYPE_DATA stream to the output video since, as soon as I add a new data stream using methods like avformat_new_stream(...) the output video is empty: neither the data stream nor the video one are produced and the output file is empty.
Can anyone suggest me a tutorial page or a sample on how to properly add a data stream to my output video in mpegts format?
Thanks a lot!
I was able to get a KLV stream added to a muxed output by starting with the "muxing.c" example that comes with the FFmpeg source, and modifying it as follows.
First, I created the AVStream as follows, where "oc" is the AVFormatContext (muxer) variable:
AVStream *klv_stream = klv_stream = avformat_new_stream(oc, NULL);
klv_stream->codec->codec_type = AVMEDIA_TYPE_DATA;
klv_stream->codec->codec_id = AV_CODEC_ID_TIMED_ID3;
klv_stream->time_base = AVRational{ 1, 30 };
klv_stream->id = oc->nb_streams - 1;
Then, during the encoding/muxing loop:
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = (uint8_t*)GetKlv(pkt.size);
auto res = write_frame(oc, &video_st.st->time_base, klv_stream, &pkt);
free(pkt.data);
(The GetKlv() function returns a malloc()'ed array of binary data that would be replaced by whatever you're using to get your encoded KLV. It sets pkt.size to the length of the data.)
With this modification, and specifying a ".ts" target file, I get a three-stream file that plays just fine in VLC. The KLV stream has a stream_type of 0x15, indicating synchronous KLV.
Note the codec_id value of AV_CODEC_ID_TIMED_ID3. According to the libavformat source file "mpegtsenc.c", a value of AV_CODEC_ID_OPUS should result in stream_type 6, for asynchronous KLV (no accompanying PTS or DTS). This is actually important for my application, but I'm unable to get it to work -- the call to avformat_write_header() throws a division by zero error. If I get that figured out, I'll add an update here.
I was writing as I could not find the answer in previous topics. I am using live555 to stream live video (h264) and audio(g723), which are being recorded by a web camera. The video part is already done and it works perfectly, but I have no clue about the audio task.
As long as I have read I have to create a ServerMediaSession to which I should add two subsessions: one for the video and one for the audio. For the video part I created a subclass of OnDemandServerMediaSubsession, a subclass of FramedSource and the Encoder class, but for the audio aspect I do not know on which classes should I base the implementation.
The web camera records and delivers audio frames in g723 format separatedly from the video. I would say the audio is raw as when I try to play it in VLC it says that it could not find any startcode; so I suppose it is the raw audio stream what is recorded by the web cam.
I was wondering if someone could give me a hint.
For an audio stream ,your override of OnDemandServerMediaSubsession::createNewRTPSink should create a SimpleRTPSink.
Something like :
RTPSink* YourAudioMediaSubsession::createNewRTPSink(Groupsock* rtpGroupsock, unsigned char rtpPayloadTypeIfDynamic, FramedSource* inputSource)
{
return SimpleRTPSink::createNew(envir(), rtpGroupsock,
4,
frequency,
"audio",
"G723",
channels );
}
The frequency and the number of channels should comes from the inputSource.
My C++ application receives a H.264 RTP video stream.
Right now it decodes the stream, saves it into a YUV file and later I use ffmpeg to re-ecode the file into something suitable to watch on a Windows PC (eg. Mpeg4 AVI).
Shouldn't it be possible to save the H.264 stream into a AVI (or similar) container without having to decode and re-encode it ? That would require some H.264 decoder on the PC to watch, but it should be much more efficient.
How could that be done ? Are there any libraries supporting that ?
using ffmpeg is correct but the answers posted so far dont look right to me.
the correct switch should be:
-vcodec copy
Your program could pipe the rtp itself through ffmpeg - even invoking it using popen3().
It seems that you need to use an intermediate SDP file - I speculate that you can specify a file you created as a named pipe or with tmpfile() which your application writes to - using the file as an intermediary.
The command-line would be something like:
int p[3];
const char* const out_fmt = "avi";
const char* cmd[] = {"ffmpeg","-f",,"-i",temp_sdp_filename,"-vcodec","copy","-f",out_fmt,"-",NULL};
if(-1 == popen3(p,cmd)) ...
// write the rtp that you receive to p[STDIN_FILENO]
// read the avi from p[STDOUT_FILENO]
// read any messages and error text from p[STDERR_FILENO]
I believe that in this circumstance ffmpeg is clever enough to repackage the container (rtp stream vs AVI) without transcoding the video and audio (this is the -vcodec copy switch); therefore, you'd have no loss of quality and it'd be blazingly fast.