Can jlayer/javazoom, or vlcj playback MPEG4, RAW AMR, or THREE GPP?
I have searched, and I cannot find how to correctly use the players, or whether they even support playback of these file formats. Are there any other players that can playback these audio formats?
I've used Xuggler http://www.xuggle.com/xuggler/ for playing AMR from files. It uses ffmpeg with opencore-amr-nb. And I think it can play other formats. You can start from http://xuggle.googlecode.com/svn/trunk/java/xuggle-xuggler/src/com/xuggle/xuggler/demos/DecodeAndPlayAudio.java
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My goal is to work out why a given video file does not play on Macos/Safari/Quicktime.
The background to this question is that it is possible to play HEVC videos with a transparent background/alpha channel on Safari/MacOS. To be playable, a video must meet the specific requirements set out by Apple in this document:
https://developer.apple.com/av-foundation/HEVC-Video-with-Alpha-Interoperability-Profile.pdf
The video that does not play on Apple/Safari/Quicktime is an HEVC video with an alpha transparency channel. Note that VLC for MacOS DOES play this file. Here it is:
https://drive.google.com/file/d/1ZnXjcDbk-_YxTgRuH_D7RSR9SXdY_XTv/view?usp=share_link
I have two example HEVC video files with a transparent background/alpha channel, and they both play fine using either Quicktime player or Safari:
Working video #1:
https://drive.google.com/file/d/1PJAyg_sVKVvb-Py8PAu42c1qm8l2qCbh/view?usp=share_link
Working video #2:
https://drive.google.com/file/d/1kk8ssUyT7qAaK15afp8VPR6mIWPFX8vQ/view?usp=sharing
The first step is to work out in what way my non-working video ( https://drive.google.com/file/d/1ZnXjcDbk-_YxTgRuH_D7RSR9SXdY_XTv/view?usp=share_link ) does not comply with the specification.
Once it is clear which requirements are not met by the non-working video then I can move onto the next phase, which is to try to formulate an ffmpeg command that will output a video meeting the requirements.
I have read Apples requirements document and I am out of my depth in trying to analyse the non working video against the requirements - I don't know how to do it.
Can anyone suggest a way to identify what is wrong with the video?
Additional context is that I am trying to find a way to create Apple/MacOS compatible alpha channel / transparent videos using ffmpeg with hevc_nvenc running on an Intel machine. I am aware that Apple hardware can create such videos, but for a wide variety of reasons it is not practical for me to use Apple hardware to do the job. I have spent many hours trying all sorts of ffmpeg and ffprobe commands to try to work out what is wrong and modify the video to fix it, but to be honest most of my attempts are guesswork.
The Apple specification for an alpha layer in HEVC requires that the encoder process and store the alpha in a certain manner. It also requires that the stream configuration syntax be formed in a specific manner. At time of writing, I'm aware of only the videotoolbox HEVC encoder being capable of emitting such a stream.
I have Live stream of HLS [https://82-80-192-30.vidnt.com/ipbc_IPBCchannel11LVMRepeat/definst/IPBCchannel11LVM_3.stream/playlist.m3u8] and I want to convert it to MPEG-DASH.
What is the best practice?
The stream is already h264 aac therefore I understand I do not need to reencode and I just need to transmux.
What should I use?
ffmpeg? mp4box?
Notes:
I used nginx-rtmp-module (https://github.com/ut0mt8/nginx-rtmp-module/) in order to create DASH from RTMP stream according to this tutorial: https://isrv.pw/html5-live-streaming-with-mpeg-dash
But nginx-rtmp-module can get as input just rtmp streams and it did not work for me with HLS stream.
I used ffmpeg in order to create dash from m3u8 as following:
ffmpeg -i https://82-80-192-30.vidnt.com/ipbc_IPBCchannel11LVMRepeat/_definst_/IPBCchannel11LVM_3.stream/playlist.m3u8 -strict -2 -min_seg_duration 2000 -window_size 5 -extra_window_size 5 -use_template 1 -use_timeline 1 -f dash out.mpd
But this is very limited. I can't control the segment duration.
The min_seg_duration parameter of ffmpeg does not work very well for me, and also it can set the minimum duration while I want to limit the maximum duration of each segment (the segment comes out with ~10 seconds, while I need it to be ~2-4 seconds as I'm playing live).
Firstly it is worth saying that if you can avoid doing this you will be saving yourself a whole lot of work!
Most devices and clients these days can play both HLS and DASH streams, so the usual approach is to add any extra functionality needed in your app or client.
If you do have to convert server side, then its worth being aware that while HLS streams typically used TS segments in the past, recently support for fragmented MP4 has become available within the HLS ecosystem.
If you have TS video streams then you will need to do a conversion along the lines you outline above with ffmpeg.
If you have fragmented MP4 then you should actually have the correct format already and may find you just have to create the manifest file so DASH can access the fragmented mp4 streams.
All the above assumes that your content is not encrypted or that you don't have to support encryption - if it is then you may not be able to convert the media, or you may have to also encrypt the media differently for some streams than others, as currently most deployed windows and chrome devices and browsers use a slightly different encryption approach (a different AES mode) than Apple devices.
I am developing a player based on ffmpeg.
Now I try to decode hls video. The video stream has several programs (AVProgram) separated by quality. I want to select one specific program with desired quality. But ffmpeg reads packets from all programs (all streams).
How can I tell ffmpeg which streams to read?
Solved by using disard field in AVStream structure:
_stream->discard = AVDISCARD_ALL;
Problem:
I have to save live video streams data which come as an RTP packets from RTSP Server.
The data come in two formats : MPEG4 and h264.
I do not want to encode/decode input stream.
Just write to a file which is playable with proper codecs.
Any advice?
Best Wishes
History:
My Solutions and their problems:
Firt attempt: FFmpeg
I use FFmpeg libary to get audio and video rtp packets.
But in order to write packets i have to use av_write_frame :
which seems that decode /encode takes place.
Also, when i give output format as mp4 ( av_guess_format("mp4", NULL, NULL)
the output file is unplayable.
[ any way ffmpeg has bad doc. hard to find what is wrong]
Second attempth: DirectShow
Then i decide to use DirectShow. I found a RTSP Source Filter.
Then a Mux and File Writer.
Cretae Single graph:
RTSP Source --> MPEG MUX ---> File Writer
It worked but the problem is that the output file is not playable
if graph is not stoped. If something happens, graph crashed for example
the output file is not playable
Also i can able to write H264 data, but the video is completely unplayable.
The MP4 file format has an index that is required for correct playback, and the index can only be created once you've finished recording. So any solution using MP4 container files (and other indexed files) is going to suffer from the same problem. You need to stop the recording to finalise the file, or it will not be playable.
One solution that might help is to break the graph up into two parts, so that you can keep recording to a new file while stopping the current one. There's an example of this at www.gdcl.co.uk/gmfbridge.
If you try the GDCL MP4 multiplexor, and you are having problems with H264 streams, see the related question GDCL Mpeg-4 Multiplexor Problem
I am working on ffmpeg and trying to add a audio stream on the fly. I am using AudioQueues and I get raw audio buffer. I am encoding audio with linear PCM and hence the audio I get will be of raw format, which I know ffmpeg does accept it. But I cannot figure out how. I have looked into AVStream, where in we have to create a new stream for this audio channel but how do I encode it to a video which is already initialized in another AVStream structure.
Overall, I would like to have an idea of the architecture of ffmpeg. I found it difficult to work since it is least documented. Any pointers or details are appreciated.
Thanks and Regards,
Raj Pawan G
If you want to use java, you'll find a much better documented API wrapper for FFmpeg with Xuggler.
That said, FFmpeg can support Raw PCM bu tnot all containers can contain it. use the PCM codecs (see avcodec.h) and find the one that has the right size and attributes you want. To add the audio to the same container find a AVFormatContext object that you use for your existing video stream, and use av_new_stream(...) to add a new stream. Then attach your PCM encoder and 'encode' to that and write resulting packets. See output_example.c in the FFmpeg for examples of this API in action.