i'm working with RTMP. I have captured RTMP packents in wireshark. Then a need take one RTMP chunk and play it. How i can do it? I try save chunk with hex editor in .h264 and play via VLC but it not help.enter image description here
a single h264 chunk is useless. It must be combined with other chunks to make frames. To play h264 you need a minimum of SPS, PPS, and IDR. SPS and PPS are sent as part of the sequence header (different frame, different chunk) and a single chunk may not contain a complete tag.
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I am using "document.querySelector('video').playbackRate=1" as a console input in the edge browser, for setting the playbackspeed on the fly. Is there a similar command to use for customizing the bitrate of that video stream?
No. The bitrate of a video is a result of how source video was encoded. In order to change bitrate, you have to either have multiple source videos that are each encoded at a different bitrate (like Youtube does it) or have the ability to change the encoding on the fly in response to network congestion (like a live stream).
Is there any way to stream mkv files on webpage using javascript/html or any other technologies? I found many questions about this, but I really want to know the answer - is this possible in any way? Maybe ajax, javascript, php, html? Maybe some external libraries? Anything?
I was wondering how youtube works. Is there possible to upload mkv file? Is so, how that videos are streaming to end user?
I know that browser doesn't support mkv natively, but maybe is some way to forcing html to do that?
Any help will be appreciated.
YouTube most probably works using the DASH protocol format. On the server side, the source audio and source video are separately divided into segments of different bitrate/quality. A manifest keeps an index of all possible segments values and their location. This allows to switch quality during playback in the player.
On the client side, the DASH (should be the same with the other main technology: HLS) manifest is used by the player to locate the segments to load in order to feed the content in two separate SourceBuffer, one for audio and one for video and both are being played synchronously together in the MediaSource. For an example player that handles this, see the Shaka Player developed by Google.
Conclusion, there is no need to use a container like mkv but each channel (video, audio) needs to point to a browser supported codec encoded segments.
You don't need anything special for streaming pre-recorded media files. A normal HTTP/1.1 or HTTP/2 server will work just fine. The browser is generally capable of seeking into the file using range requests.
Matroska (MKV) is a container format, and it actually is widely supported because it's basically the same as WebM. WebM is a subset of Matroska... the key differences being that there are suggested codecs for use. (Matroska itself supports almost anything.)
Your audio and video tracks in the file can use a variety of codecs... the key is to use codecs compatible with browsers. Opus for audio and VP8 for video will take you far.
From there, simply reference your video file in a <video> tag.
I have an MP4 video that is variable bitrate, so the average bitrate doesn't necessarily stay consistent throughout the entire file. Because my video is a capture of a computer screen, some parts of the video are very low bitrate because nothing is happening, and other parts are a much higher bitrate because there's a lot of activity on the screen.
How does Chrome decide how much video to buffer for progressive download HTTP(S) videos? I'm running into a problem where Chrome tends to buffer too little, so playback stutters.
If there's no way of convincing Chrome to download a certain time of video (and I don't want to just preload the entire thing), can I author the MP4 some special way to solve the problem? I'm using FFmpeg and MP4Box. Maybe it's up to the HTTP server?
If you want more control over the playback of the video, you should definitely check out MediaSourceExtensions. They define a whole model for video playback, defining sourceBuffers where you can put video data, etc.
Beware it is not a simple to use API still, and the information on how to use it is very fragmented.
In your case, if you go the MSE route, you can either keep using h264 (which is probably the codec your mp4 is wrapping) or switch to webm.
In case of going the MP4, h264 route, you'll need to generate a fragmented MP4 (fMP4) and write some JavaScript to control the way you work with the MP4 fragments (segments in MSE parlance).
MP4Box has support for the -dash protocol, which will segment an MP4 in a way that is suitable for consumption via MSE. FFmpeg also has support for generating fragmented MP4.
I have a very fast connection and it takes about 2-3 seconds before the song actually starts playing. It's a relatively average 128kbps MP3 size (3mb-4mb). I have set preload="auto" but that didn't help much. Is there a way to just start playing the audio right away and continue to buffer it (sort of like YouTube does)?
Here is an example that I am currently working on. It's going to play an audio simultaneously on all connected clients. So if you have 2+ laptops, you can try it out. All computers must be connected before you start playing the audio. (double click on a song to start playing).
Running video and audio without complete buffering is called smooth / adaptive streaming. It can be achieved in players like silverlight and flash.
What it actually does is to create chunks of files and let the user play file chunk by chunk. Since you are downloading chunks, it will not require whole file to download.
Well, I am not giving you the full fledged answer since I haven't studied much but I am giving you the exact idea of how it works.
I had same issue but with HTML5 Video.. I overcame it with using Smooth streaming media Azure..
Here is a tutorial of the same : http://www.wrapcode.com/featured/windows-azure-media-services-mp4-to-smooth-streaming/
I will keep you updated once I find something useful :-)
If you use preload=none, then you have no buffer at the beginning but it will buffer your content "on the fly"
I have an Icecast server which streamsmy contet, and when I use pause and play, it buffers my content, even with preload=none.
Do not use preload=auto. It will take some time to start.
I created a video stream recording application that works fine except the recorded FLVs are corrupt a little bit. :) If I open an FLV in VLC player everything is green but getting "clean" when changes occur. And especially at the beginning of the video is breaking up. (I use Red5 1.0)
For pre-recorded streams, the keyframes are already encoded into the file and they cannot be changed. If you're serving a live stream, the keyframes need to be set in the application that is encoding the live stream (for example, Flash Media Live Encoder).