How to implement the Adobe HTTP Streaming spec wit

Question

As of Flash 10.1, they have added the ability to add bytes into the NetStream object via the appendBytes method (described here http://www.bytearray.org/?p=1689). The main reason for this addition is that Adobe is finally supporting HTTP streaming of video. This is great, but it seems that you need to use the Adobe Media Streaming Server (http://www.adobe.com/products/httpdynamicstreaming/) to create the correct video chunks from your existing video to allow for smooth streaming.

I have tried to do a hacked version of HTTP streaming in the past where I swap out the NetStream objects (similar to here http://video.leizhu.com/video.html), but there is always a momentary pause between the chunks. With the new appendBytes, I tried to do a quick mock up with the two sections of video from the preceding site, but even then, the skip still remains.

Does anyone know how the two consecutive .FLV files needs to be formated in order for the appendBytes method on the NetStream object to create a nice smooth video without a noticeable skip between the segments?

Answer 1

You don't need to use their server. Wowza supports Adobe's version of HTTP Streaming and you can implement it yourself by segmenting the videos properly and loading all the segments on a standard HTTP server.

Links to all the specs for Adobe's HTTP Streaming are here:

http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS9463dbe8dbe45c4c-1ae425bf126054c4d3f-7fff.html

Trying to hack the client to do some custom style http streaming will be a lot more troublesome.

Note that HTTP Streaming does not support streaming several different videos but streams a single file that was broken off into separate segments.

File Packager

A command-line tool that translates on-demand media files into fragments and writes the fragments to F4F files. The File Packager is an offline tool. You can use the File Packager to encrypt files for use with Flash Access. For more information, see Packaging on-demand media.

The File Packager is available from adobe.com and is installed with Adobe® Flash® Media Server to the rootinstall/tools/f4fpackager folder.

Packager download link is on right here: Download File Packager for HTTP Dynamic Streaming

http://www.adobe.com/products/httpdynamicstreaming/

Answer 2

Adobe's F4F format is based on MP4 files, are you able to use F4V or MP4 instead of FLV files? There are plenty of MP4 file splitters around but you would need to make sure the timestamps in the files are continuous, maybe the pause happens when it sees a zero timestamp within the audio or video stream inside the file.

Answer 3

I recently found a similar project for node.js to achieve m3u8 transcoding (https://github.com/andrewschaaf/media-server) but have yet to hear of one besides Wowza doing it outside of Origin module for Apache. Since the payloads are nearly identical you're better off looking for a good mp4 segmenting solution (plenty out there) than looking for f4m segmenting. The problem is moov atoms especially on larger mp4 video are difficult to manage and put in their proper initial (near beginning of file) location. Even using optimal ffmpeg settings and 'qtfaststart' you end up with noticeably slower seeking, inefficient bandwidth usage (usually greedy), and a few minor headaches relating to scrubbing/time that you don't get with flv/f4v playback.

In my player I have or intend to switch between HTTP Dynamic Streaming (HDS) and MP4 based on load and realtime log parsing Apache using awk/cron instead of licensing Adobe's Access product for stream protection .. both have unique 'onmetadata' handlers.. but in the end I receive sequenced time/byte hashes virtually equivalent. Just MP4 is slower. So mod_origin is just a synchronizer / request router for Flash clients (over http). I'm still looking for ways to speed up mp4-container-based playback. One incredible solution I read this recently and was rather awestruck by it http://zehfernando.com/2011/flash-video-frame-time-woes/ where a video editor (guy) and flash developer came up with their own mp4 timecoding solution that literally added (via Adobe Premiere script) about 50 pixels to the bottom of every video frame with a visual 'binary' stamp like a frame barcode.. and those binary values translate into highly-accurate timecode values. So Flash could analyze the video frames as they were painted (realtime) and determine precisely where the player was and what bytes were needed from any kind of mp4 byte-segmenting-friendly webserver. The thing is (and perhaps I'm wrong here) Flash seems to arbitrarily choose when it gets to moov data, especially on large video files (.5-1.5gigs). Even if you make sure to run your mp4 through MP4Box (i.e. MP4Box -frag 10000 -inter 0 movie.mp4) I guess this has been a problem OSMF and HDS have worked on quite well now, though it is annoying that you need Apache and a proprietary closed-source module to use it imo. Its probably just a matter of time before open source implementations arrive as HDS is only 1-2 years old, and it just needs a little reverse engineering like that Andrew Chaaf guy with node.js + mpegts streaming (live or not). In the end I may just end up using OSMF exclusively beneath my UI as it seems to have similar virtues to HDS if not more so i.e. Strobe if you need sick extensible HDS or MP4 open player platform to hack from to realize your own custom player.

Answer 4

The place in the OSMF code where this happens is the timer-fired state machine inside of the HTTPNetStream class implementation... might be an informative read. I think I even put some helpful comments in there when I wrote it.

As far as the general question:

If you read an entire FLV file into a ByteArray and pass it to appendBytes, it will play. If you break that FLV file in half, and pass the first half as a byte array and then the second half as a byte array, that will play as well.

If you want to be able to switch around between bitrates without a gap, you need to split up your FLV files at matching keyframe points... and remember that only the first call to appendBytes has the initial FLV file header ('F', 'L', 'V', flags, offset)... the rest just expect a continuation of the FLV byte sequence.

Answer 5

I was able to get this working using Adobe's File Packager Tool which Samuel described. I didn't use the NetStream object but I used the OSMF Sample Player which I assume uses this internally. Here's how to do with without using FMS:

1. Get Adobe's File Packager for Http Dynamic Streaming from http://www.adobe.com/products/httpdynamicstreaming/
2. Run the File Packager on an existing MP4 file containing H.264/AAC like this: C:\Program Files\Adobe\Flash Media Server 4\tools\f4fpackager> f4fpackager.exe --input-file="MyFile.mp4" --segment-duration=30

This will result in 30 second long F4F files, also F4X and a F4M file. The F4F files are your correctly segmented (and fragmented) MP4 files that should play. If you want to test this using the OSMF Player also do the following:

1. Get Apache Server
2. Get Adobe's Http Origin Module for Apache from http://www.adobe.com/products/httpdynamicstreaming/
3. Install the module according to http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS8d6ed60bd880807c48597a9e1265edd6cc0-8000.html
4. Put the F4F, F4X and F4M file into the vod directory under httpdocs
5. Get the “OSMF Sample Player for HTTP Dynamic Streaming” from http://www.osmf.org/downloads/OSFMPlayer_zeri2.zip
6. Put the Sample Player in the httpdocs directory
7. Load the html file from the Sample Player in a browser eg http://localhost/OSMFPlayer.html
8. Press the eject button and put in the URL of your F4M file, it should play

So to answer the original question Adobe's File Packager is the file splitter to use, you don't need to buy FMS to use it and it works for FLV and MP4/F4V files.

Answer 6

You could use F4Pack, it's a GUI around the commandline-tool from Adobe, that lets you process your flv/f4v file so they can be used for HTTP Dynamic Streaming.