I have been trying to implement HTTP live streaming using mpeg-dash but need guidance on some issues.
Provided :
I have audio and video encoded stream in buffered input.
a direct mpeg-2 transport stream for above is also available in a buffer.
Current approach :
Save the transport stream into chunks of fixed length.
use ffmpeg to extract video stream.
ffmpeg -i latest_chunk.ts -s 720x480 -c:v libx264 -b:v 600k -y -an output_video_stream.mp4
use ffmpeg to extract video stream.
ffmpeg -i latest_chunk.ts -c:a aac -b:a 128k -y -vn output_audio_stream.mp4
use mp4box to create dash segments and mpd.
mp4box -dash 7000 -profile live output_video_stream.mp4 output_audio_stream.mp4 -out manifest.mpd
A server running continuously in another thread serves the generated mpd and segments.
Issues :
The above approach gives a considerable amount of latency. Can this be done more efficiently?
I want to know if there is a method to take directly encoded streams buffer as input and produce mpeg-dash segments and mpd. HTTP server will do the rest. If there is, please provide an example.
Also i provided the length of the transport stream chunks (in sec) in mp4box as argument -mpd-refresh 12, but the player only requests for the mpd once, plays the segments, and stops. It also does not include minimumUpdatePeriod attribute in the generated mpd file
mp4box -dash 7000 -profile live -mpd-refresh 12 output_video_stream.mp4 output_audio_stream.mp4 -out manifest.mpd
Does the mpeg-dash has support for mpeg-2 encoded media streams?
Any advice/solution/reference for the same is appreciated.
Related
I have an RTSP Stream with one video stream and three audio streams as the source. Two of the audio streams are encoded with .mp2 and one is encoded with .ac-3. I want to convert the .mp2 streams to AAC. This would be easy if the .mp2streams would have the same stream identifier every time I start ffmpeg, but unfortunately the stream identifiers change. This means sometimes the two .mp2 streams are 0:a:0 and 0:a:1 and the next time they are 0:a:1 and 0:a:2.
Is there an option to re-encode only the .mp2 streams and keep the .ac-3 stream untouched?
I should probably also mention that this encoding is used for live TV so it is not an option to produce intermediate files or have several ffmpeg commands.
Try
ffprobe -show_entries stream_tags -select_streams a INPUT_URL
and see if there are any stream tags (metadata) that distinguishes mp2 streams. Then you can use the metadata stream specifier to selectively set re-encoding:
ffmpeg ... -c copy -c:a:m:{name}:{value} ac3 ...
where {name} and {value} are the name and value of the tag, respectively.
Reference on stream specifier: https://ffmpeg.org/ffmpeg.html#Stream-specifiers-1
If there isn't any usable tag, your only solution likely is to run ffprobe first to identify the stream # before running ffmpeg.
A bit of history. I am using Plex as my media server, but for reasons unknown, it has issues transcoding the DTS-HD MA 7.1 audio to EAC3 stereo and keeps buffering (the server has plenty of horsepower on all fronts, CPU/RAM/drive space & speed, gigabit networks connections for all devices. The playback device (TCL Roku TV, with a 3rd party soundbar connected via HDMI ARC) doesn't support the built-in 7.1 audio, so I get silence if I play it back directly by putting the file on a USB stick.
Also, I am by no means a ffmpeg guru, I figured out what I do know by Google University and asking questions, so please be kind and forgive me if I ask follow-up questions that may seem n00b-ish, and please provide example commands (preferably in the context of my command below so that I can have a known point of reference to start with).
I have a movie with 4K (HEVC Main 10 HDR) video and DTS-HD MA 7.1 audio that I am looking to leave the video and audio untouched, but to add a 2nd audio track in either EAC3 or if necessary, just AC3 in stereo
So what I am looking for is as follows:
video.mkv
Existing->4k video file (no change)
Existing->7.1 audio (no change)
Convert and add->stereo audio as a 2nd audio track to the output.mkv file
Below is the command I've historically used with ffmpeg to convert and replace the audio file with the stereo audio, but since I'd prefer to leave the 7.1 audio in place, this doesn't work:
ffmpeg -i "D:\video.mkv" -c:v copy -c:a aac -b:a 128k "D:\output.mkv"
And if this cannot be done as a single command, please also let me know what steps I do need to take to be able to do it.
Thanks in advace,
Mike
ffmpeg -i input.mkv -map 0 -map 0:a -c copy -c:a:1 eac3 output.mkv
-map 0 select all streams.
-map 0:a select all audio streams. This combines with -map 0 so now you have 1 video and 2 audio streams selected.
-c copy stream copy all streams.
-c:a:1 eac3 encode output audio stream #1 with eac3 encoder. This overrides -c copy for this particular stream.
What I want is to be able to create a livestream from a Ubuntu v14.04 server to a RTMP Server (like Twitch) and to be able to use NodeJS to control visual aspects (adding layers, text, images) and add different sources (video files, others livestreams, etc). Like having OBS running on a server.
What I've done/researched so far:
FFmpeg
With ffmpeg I can can create video files streams like that:
ffmpeg -re -i video.mp4 -c:v libx264 -preset fast -c:a aac -ab 128k -ar 44100 -f flv rtmp://example.com
Also using the filter_complex I can create something near to a layer like this tutorial explains:
https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20input%20videos
But I found the following problems:
The streams that I create with ffmpeg only last until the video file is over, if I wanted to stream multiple video files (dynamic playlist) it would interrupt the stream between each file;
The manipulation is very limited as far as I am concerned, I can't edit filter_complex once ffmpeg is executing;
Can't display text and create animated overlays, like sliding text.
I tried to search for any cli/nodejs package that is able to create a continuos video stream and manipulate it to use as input source for ffmpeg which streams to the RTMP server.
Can someone give me more information about what I am trying to do?
I'm playing with github.com/fluent-ffmpeg/node-fluent-ffmpeg to see if I have a different outcome.
With FFMPEG, I'm sending a stream from Computer A over to Computer B, via UDP.
This is done over a MPEGTS stream, encoded with libx264 and aac.
Computer B takes this stream with FFMPEG and puts it into an m3u8 playlist.
After a random time (2-35 minutes), the message
[mpegts # 0533f000] AAC bitstream not in ADTS format and extradata missing
av_interleaved_write_frame(): Invalid data found when processing input
appears.
What I figures is that the receiving FFMPEG can't read the header file of the audio part for this particular package, and since it can't put video and audio together anymore, it stops creating the .ts files and just stops running.
Here's the cmdline of the receiving stream:
ffmpeg -i udp://address -vcodec copy -acodec copy -map 0 -f segment -segment_list playlist.m3u8 -analyzeduration 100000 -probesize 100000-segment_list_flags +live-cache -segment_time 8 -segment_wrap 10 out%03d.ts
Now I need to know the answer to either one of these 2 questions:
1) Can I put something in my commandline in order to avoid this particular problem or
2) Can I tell FFMPEG to just ignore it for this particular message, quite possibly creating weird audio or none at all, and to simply move on to the next one?
I know that there are a million ways to download a video from youtube and then convert it to audio or do further processing on it. But recently I was surprised to see an app called YoutubeToMp3 on mac actually showing "Skipping X mb of video" and supposedly only downloading the audio from the video, without the need to use bandwith to download the entire video and then convert it. I was wondering if this is actually correct and possible at all because I cant find any way to do that. Do you have any ideas ?
EDIT:
After some tests here is some additional information on the topic. The video which I tried to get the audio from is just a sample mp4 file from the internet:
http://download.wavetlan.com/SVV/Media/HTTP/MP4/ConvertedFiles/MediaCoder/MediaCoder_test6_1m9s_XVID_VBR_306kbps_320x240_25fps_MPEG1Layer3_CBR_320kbps_Stereo_44100Hz.mp4
I tried
ffmpeg -i "input" out.mp3
ffmpeg -i "input" -vn out.mp3
ffmpeg -i “input” -vn -ac 2 -ar 44100 -ab 320k -f mp3 output.mp3
ffmpeg -i “input” -vn -acodec copy output.mp3
Unfortunately non of these commands seems to be using less bandwith. They all download the entire video. Now that you have the video can you confirm if there is actually a command that downloads only the audio stream from it and lowers the bandwith usage? Thanks!
After a lot of research I found out that this is not possible and developed an alternative approach:
Download the mp4 header
Parse the header and get the locations of the audio bytes
Download the audio bytes with http range requests and offsets
Assemble the audio bytes and wrap them in a simple ADTS container to produce a playing m4a file
That way only bandwidth for the audio bytes is used. If you find a better approach of doing it please let me know.
For a sample Android APP and implementation check out:
https://github.com/feribg/audiogetter/blob/master/audiogetter/src/main/java/com/github/feribg/audiogetter/tasks/download/VideoTask.java
FFmpeg is capable of accepting an URL as input. If the URL is seekable, then FFmpeg could theoretically skip all the video frames, and thus it would need to download only the data for the audio stream.
Try using
ffmpeg -i http://myvideo.avi out.mp3
and see if it takes less bandwidth.