I've uploaded & encoded a video as an asset using AMS's AdaptiveStreaming preset to be able to stream the whole video.
I now need to split the video into short clips that will need be streamed as well. How do I go about it?
(The start & end times of the video-clips are known.)
Are your clips going to be at arbitrary in/out points (eg. start at 3 seconds 22 frames into the input video, and stop 18 seconds 12 frames)? If so, you would need to use the original video source, and follow the steps outlined here. As indicated in that article, you would have to start with one of the presets where the bitrates, resolutions etc. are explicitly defined. At this time, we don't support the ability to combine clipping/trimming with the "Adaptive Streaming" preset.
Related
I have two audio files of different durations. I want to play them simultaneously with the shorter duration starting in the middle of the longer duration file.
I've enabled media synchronization with the app launch setting --sout-all --sout #display.
Swapping between input-master and -slave settings results in either the shorter file not playing or nothing played back.
How can this be done in VLC?
As of the date of this response, asynchronized audio playback or recordings where files start and end at different times cannot be done using a single instance of VLC alone without customized addons, if available. This case is not an intended use of the VLC standard application.
It is possible yet cumbersome to have asynchronized playback with two or more instances of VLC and manually starting and ending each audio track.
ALTERNATIVES
Alternatively, there are numerous online audio editing tools, many accessible freely, that permit uploading audio files as separate tracks for playback, editing, mixing, or recording and for downloading that are much more advanced than the features of an uncustomized VLC.
A web search for "audio editors online" will produce a lengthy list of options.
as I said in the title, I need to record my screen from an electron app.
my needs are:
high quality (720p or 1080p)
minimum size
record audio + screen + mic
low impact on PC hardware while recording
no need for any wait after the recorder stopped
by minimum size I mean about 400MB on 720p and 700MB on 1080p for a 3 to 4 hours recording. we already could achieve this by bandicam and obs and it's possible
I already tried:
the simple MediaStreamRecorder API using RecordRTC.Js; produces huge file sizes, like 1GB per hour for 720p video.
compressing the output video using FFmpeg; it can take up to 1 hour for 3 hours recording
save every chunk with 'ondataavailable' event and right after, run FFmpeg and convert and reduce the size and append all the compressed files (also by FFmpeg); there are two problems. 1, because of different PTS but it can be fixed by tunning compress command args. 2, the main problem is the audio data headers are only available in the first chunk and this approach causes a video that only has audio for the first few seconds
recording the video with FFmpeg itself; the end-users need to change some things manually (Stereo Mix), the configs are too complex, it causes the whole PC to work slower while recording (like fps drop; even if I set -threads to 1), in some cases after recording is finished it needs many times to wrap it all up
searched through the internet to find applications that can be used from the command line; I couldn't find much, the famous applications like bandicam and obs have command line args but there are not many args to play with and I can't set many options which leads to other problems
I don't know what else I can do, please tell me if u know a way or simple tool that can be used through CLI to achieve this and guide me through this
I end up using the portable mode of high-level 3d-party applications like obs-studio and adding them to our final package. I also created a js file to control the application using CLI
this way I could pre-set my options (such as crf value, etc) and now our average output size for a 3:30 hour value with 1080p resolution is about 700MB which is impressive
I come to a technical problem and I need you.
Situation data:
I record the screen as well as 1 to 2 audio tracks (microphone and speaker).
These three recordings are done separately (it could be mixed but I don't prefer) and every 10s (this is configurable), I send the chunk of recorded data to my backend. We, therefore, have 2 to 3 chunks sent every 10s.
These data chunks are interdependent. Example: The 1st video chunk starts with the headers and a keyframe. The second chunk can be in the middle of a frame. It's like having the entire video and doing a random one-bit split.
The video stream is in h264 in a WebM container. I don't have a lot of control over it.
The audio stream is in opus in a WebM container. I can't use aac directly, nor do I have much control.
Given the reality, the server may be restarted randomly (crash, update, scaled, ...). It doesn't happen often (4 times a week). In addition, the customer can, once the recording ends on his side, close the application or his computer. This will prevent the end of the recording from being sent. Once it reconnects, the missing data chunks are sent. This, therefore, prevents the use of a "live" stream on the backend side.
Goals :
Store video and audio as it is received on the server in cloud storage.
Be able to start playing the video/audio even when the upload has not finished (so in a live stream)
As soon as the last chunks have been received on the server, I want the entire video to be already available in VoD (Video On Demand) with as little delay as possible.
Everything must be distributed with the audios in AAC. The audios can be mixed or not, and mixed or not with the video.
Current and blocking solution:
The most promising solution I have seen is using HLS to support the Live and VoD mode that I need. It would also bring a lot of optimization possibilities for the future.
Video isn't a problem in this context, here's what I do:
Every time I get a data chunk, I append it to a screen.webm file.
Then I spit the file with ffmpeg
ffmpeg -ss {total_duration_in_storage} -i screen.webm -c: v copy -f hls -hls_time 8 -hls_list_size 0 output.m3u8
I ignore the last file unless it's the last chunk.
I upload all the files to the cloud storage along with a newly updated output.m3u8 with the new file information.
Note: total_duration_in_storage corresponds to the time already uploaded
on cloud storage. So the sum of the parts presents in the last output.m3u8.
Note 2: I ignore the last file in point 3 because it allows me to have keyframes in each song of my playlist and therefore to be able to use a seeking which allows segmenting only the parts necessary for each new chunk.
My problem is with the audio. I can use the same method and it works fine, I don't re-encode. But I need to re-encode in aac to be compatible with HLS but also with Safari.
If I re-encode only the new chunks that arrive, there is an auditory glitch
The only possible avenue I have found is to re-encode and segment all the files each time a new chunk comes along. This will be problematic for long recordings (multiple hours).
Do you have any solutions for this problem or another way to achieve my goal?
Thanks a lot for your help!
During capturing from some audio and video sources and encoding at AVI container for synchronizing audio & video I set audio as a master stream and this gave best result for synchronizing.
http://msdn.microsoft.com/en-us/library/windows/desktop/dd312034(v=vs.85).aspx
But this method gives a higher FPS value as a result. About 40 or 50 instead of 30 FPS.
If this media file just playback - all OK, but if try to recode at different software to another video format appears out of sync.
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
How can I programmatically set dwScale and dwRate values in the AVISTREAMHEADER structure at AVI muxing?
MSDN:
This method works by adjusting the dwScale and dwRate values in the AVISTREAMHEADER structure.
You requested that multiplexer manages the scale/rate values, so you cannot adjust them. You should be seeing more odd things in your file, not just higher FPS. The file itself is perhaps out of sync and as soon as you process it with other applciations that don't do playback fine tuning, you start seeing issues. You might be having video media type showing one frame rate and effectively the rate is different.
Basically I'm trying to replicate YouTube's ability to begin video playback from any part of hosted movie. So if you have a 60 minute video, a user could skip straight to the 30 minute mark without streaming the first 30 minutes of video. Does anyone have an idea how YouTube accomplishes this?
Well the player opens the HTTP resource like normal. When you hit the seek bar, the player requests a different portion of the file.
It passes a header like this:
RANGE: bytes-unit = 10001\n\n
and the server serves the resource from that byte range. Depending on the codec it will need to read until it gets to a sync frame to begin playback
Video is a series of frames, played at a frame rate. That said, there are some rules about the order of what frames can be decoded.
Essentially, you have reference frames (called I-Frames) and you have modification frames (class P-Frames and B-Frames)... It is generally true that a properly configured decoder will be able to join a stream on any I-Frame (that is, start decoding), but not on P and B frames... So, when the user drags the slider, you're going to need to find the closest I frame and decode that...
This may of course be hidden under the hood of Flash for you, but that is what it will be doing...
I don't know how YouTube does it, but if you're looking to replicate the functionality, check out Annodex. It's an open standard that is based on Ogg Theora, but with an extra XML metadata stream.
Annodex allows you to have links to named sections within the video or temporal URIs to specific times in the video. Using libannodex, the server can seek to the relevant part of the video and start serving it from there.
If I were to guess, it would be some sort of selective data retrieval, like the Range header in HTTP. that might even be what they use. You can find more about it here.