I am trying to use ffmpeg, and have been doing a lot of experiment last 1 month.
I have not been able to get through. Is it really difficult to use FFmpeg?
My requirement is simple as below.
Can you please guide me if ffmpeg is suitable one or I have implement on my own (using codec libs available).
I have a webm file (having VP8 and OPUS frames)
I will read the encoded data and send it to remote guy
The remote guy will read the encoded data from socket
The remote guy will write it to a file (can we avoid decoding).
Then remote guy should be able to pay the file using ffplay or any player.
Now I will take a specific example.
Say I have a file small.webm, containing VP8 and OPUS frames.
I am reading only audio frames (OPUS) using av_read_frame api (Then checks stream index and filters audio frames only)
So now I have data buffer (encoded) as packet.data and encoded data buffer size as packet.size (Please correct me if wrong)
Here is my first doubt, everytime audio packet size is not same, why the difference. Sometimes packet size is as low as 54 bytes and sometimes it is 420 bytes. For OPUS will frame size vary from time to time?
Next say somehow extract a single frame (really do not know how to extract a single frame) from packet and send it to remote guy.
Now remote guy need to write the buffer to a file. To write the file we can use av_interleaved_write_frame or av_write_frame api. Both of them takes AVPacket as argument. Now I can have a AVPacket, set its data and size member. Then I can call av_write_frame api. But that does not work. Reason may be one should set other members in packet like ts, dts, pts etc. But I do not have such informations to set.
Can somebody help me to learn if FFmpeg is the right choice, or should I write a custom logic like parse a opus file and get frame by frame.
Now remote guy need to write the buffer to a file. To write the file
we can use av_interleaved_write_frame or av_write_frame api. Both of
them takes AVPacket as argument. Now I can have a AVPacket, set its
data and size member. Then I can call av_write_frame api. But that
does not work. Reason may be one should set other members in packet
like ts, dts, pts etc. But I do not have such informations to set.
Yes, you do. They were in the original packet you received from the demuxer in the sender. You need to serialize all information in this packet and set each value accordingly in the receiver.
Related
I'm new to FFmpeg. When learn it with the nice repo(https://github.com/leandromoreira/ffmpeg-libav-tutorial)，in the hello_world example I find avcodec_receive_frame dosen't return the first I frame until it gets the third packet, as following screenshot shows:
I'm wondering why additional packets are needed to receive an I frame.
Most modern video codecs are using I/P/B frames which brings the decoding time stamp (DTS) and presentation time stamp (PTS). So, what hello_world does with ffmpeg's lib is the following:-
Demuxing (av_read_frame)
Demuxes packets based on file format (mp4/avi/mkv etc.) until you have a packet for the stream that you want (eg. video) - (We might could say NAL units as an example here - not sure)
Feeds the decoder with the packet (avcodec_send_packet)
Starts the decoding process until it has enough packets to give you the first frame (decodes based on DTS)
Checks whether a frame is ready to be presented (avcodec_receive_frame)
Asks the decoder if it has a frame to be presented after feeding it. It might not be ready and you need to re-feed it or even it might give you more than 1 frames at once. (Frames comes out based on PTS)
I'm trying to write a decoder for a webrtc app in C. I receive a RTP stream, I parse every packet, reorder them, and put the payload in a AVPacket, as described here (FFmpeg decode raw buffer with avcodec_decode_video2).
The reordering part is not described in this link but I'm pretty sure this part is OK.
The question is, I dont know how to give the decoder information about resolution, pix_fmt etc. Do I need to create an AVstream* and fill it with all information I took from rtp header?
Do someone have a piece of running code that decode a VP8 packet depacketized without the use of rtp_dec etc.?
In this link, no more information seems to be sent to the decoder, is it able to decode without knowing resolution and without any header?
You don't need to feed "resolution, pix_fmt etc." information to decoder, as those get derived by decoder from the input AVPackets.
Encoders need such information like resolution, pix_fmt etc. to generate the compressed byte/bit-stream. And, encoders embed this (resolution, pix_fmt etc.) information in generated bit-stream. Once decoder receives bit-stream in right order, it derives the resolution, pix_fmt info before proceeding to de-compressing it.
Probably, the packet order that you are feeding to decoder is the cause in your case.
I have following doubts in ffmpeg.Please clarify.
1.I am reading mp4 file using ffmpeg and after doing av_read_frame
i am getting video(stream_index = 0),audio(stream_index = 1) packets
First there is no order of coming of video/audio packets.Is it the standard case.
2.Video packets are coming of various sizes from the minimum of 14 to the maximum of 21824.Please point why the video packet size varies,somewhere it
is written that for video one packet means one frame,so if for video pkt->size = 14 also equals one frame of video.(pkt is of type AVPacket).
3.If we demux the incoming mp4 stream in Probe function while parsing it and store the info in some buffer or calling av_read_frame demuxes it.
4.Is it possible in ffmpeg to demux the mp4 file and then assign both a/v in one stream of packets where video packet has stream_index = 1,and for audio = 0.
or it has to be in separate stream.
5.Diff between ffmpeg processing for transport stream and mp4 file.
if both are demuxed,decoded in same way or is it different.
Regards
Mayank
A media file is created from mutipule streams. A stream can be of mnay types. Audio, video, captions, metadata etc. But a stream can NOT be of multiple types.
1) av_read_frame will (usually) return the frames in the order they are written to the file. If the software that created the file did not mux them monotonically, you can not read it monotonically.
2) This is precisely how video compression works. The codec stores only the changes between frames. If there is very little motion, then one frame may be very similar to the previous frame, so the delta is very small.
3) This is not a question.
4) No.
5) Largely, no. But there is some difference in the file types. mp4 requires random access, while TS does not.
I want to stream video (no audio) from a server to a client. I will encode the video using libx264 and decode it with ffmpeg. I plan to use fixed settings (at the very least they will be known in advance by both the client and the server). I was wondering if I can avoid wrapping the compressed video in a container format (like mp4 or mkv).
Right now I am able to encode my frames using x264_encoder_encode. I get a compressed frame back, and I can do that for every frame. What extra information (if anything at all) do I need to send to the client so that ffmpeg can decode the compressed frames, and more importantly how can I obtain it with libx264. I assume I may need to generate NAL information (x264_nal_encode?). Having an idea of what is the minimum necessary to get the video across, and how to put the pieces together would be really helpful.
I found out that the minimum amount of information are the NAL units from each frame, this will give me a raw h264 stream. If I were to write this to a file, I could watchit using VLC if adding a .h264
I can also open such a file using ffmpeg, but if I want to stream it, then it makes more sense to use RTSP, and a good open source library for that is Live555: http://www.live555.com/liveMedia/
In their FAQ they mention how to send the output from your encoder to live555, and there is source for both a client and a server. I have yet to finish coding this, but it seems like a reasonable solution
I have to write a RTMP parser which will handle the packets captured form a RTMP stream on wireshark and i will extract the data from the pcap.
I have gone through the specs ad i am able to understand the handshake process and also able to locate the media in TCP packets but i am confused in case of Multiple Audio/Video session which are interleaved within a single pcap, how we can handle that in the parsing so as make our parser able to parse multiple stream simultaneously. Any uniqueness will be very helpful for the simultaneous parsing of the different RTMP streams.
EDIT (after #Martin Redmond's answer): yeah that I am able to figure out but it seems like some FLV data is being streamed over the RTMp but that FLV header is missing and there seems to be different handshake and FLV data is streaming for same IP with different ports. So, i am not able to find if its the real FLV file or only header as if i extract only the header and the other data, i am not able to make a FLV file from it.
Any way to validate or extract the media from that RTMP stream???
The header information for each chunk of data lets you figure out which stream the chunk belongs to. It's not straight forward though. The header information gets compressed and the relevant info may have only been sent at the begining of the stream so you need have a context for each chunk.
The important part is the streamid. Video and audio from the same source will have the same streamid but will have different channel numbers and datatypes.
In the spec. the streamid is referred to as the message stream id (section 6.1.2.1) and is only sent with a type 0 header.