[gst-devel] Integrating Swfdec and GStreamer

[in7y118 at public.uni-hamburg.de]

Hi,

I've recently been looking at integrating Swfdec and GStreamer. There  
are some issues I'd like advice on. In order of importance for me:

1) Using GStreamer codecs
Swfdec currently has a simple API for decoding data that looks like
GstBuffer *decode_next_buffer (Decoder *dec, GstBuffer *input); [1]
Which is used to decode video (VP6 and Sorensen Squeeze/H263 to RGBA)  
and audio (MP3, Nellymoser to 44kHz raw audio). The decoding API can't  
include demuxers (like FLV), since those demuxers trigger script  
events [2], and I need to ensure they do happen and happen in the  
right order. And it can't pass the data on to the API. Video is  
rendered just like every other object, and as such can be rotated,  
scaled, affected by a clipping region or place below other object.  
Audio can be modified by scripts or the file (volume, panning). [3]
My idea to implement this has been to spawn a pipeline like: appsrc !  
right/caps,go=here ! decodebin ! right/caps,go=here ! appsink
But as you can see above, the current Swfdec API decodes only one  
buffer at a time and then wants all available output. Since the  
available output is not necessary one buffer (it is for video), but  
may be 0-n (in the mp3 case), how do I make sure I got all available  
output before returning from the decode function?
Or even better: Does someone of you have a better idea on how to  
implement the Swfdec codec API to make it more suited for APIs like  
GStreamer's?

2) Using GStreamer as audio output
The current audio API in Swfdec is modelled after how the official  
player works and I've had the best results regarding start/stop  
latency in ALSA with this approach. When a new audio stream [4]  
becomes available [5], a new ALSA pcm is openened, and when the stream  
gets removed by the player, I close the pcm. This way (as opposed to  
mixing the audio myself), the open/close operations avoid the sound  
card's buffer and happen way faster.
So I've been thinking about opening a new pipeline with appsrc !  
autoaudiosink  for every new stream and run it for as long as the  
stream plays. Since you can easily find Flash files which have 5 of  
those streams playing, I've been wondering if this might cause issues  
for the varying sound backends. I know it works on Linux/ALSA with  
dmix and it'll probably work on $SOUND_SERVER_OF_CHOICE, but what  
about Solaris or BSD? Any opinions on this?

3) Writing a Swfdec plugin for GStreamer
People have asked me why I removed the GStreamer plugin from Swfdec.  
This is motivated by Swfdec now supporting Actionscript and what the  
Actionscript stuff can do to one's Flash files. This whole new can of  
worms would make lots of Flash files complicated to get right,  
especially in the autoplugger. Since I don't have the time to hack two  
output frontends and the GStreamer one was problematic, I removed the  
plugin. Btw, I think these problems would apply to other formats  
interesting for GStreamer, too, like for example SMIL. In no  
particular order:

- Flash files can load (multiple) files (at once). Currently there's  
no good way to request data from elsewhere. I've been told there is an  
event to request a different URI that is used by some demuxers, but  
that requires the current file to be complete and does not support  
more than one file.
- If Flash wants to open new files, it needs the URL of the input for  
local URL resolving and for security management. Last I looked the URL  
was not part of the (meta)data in the stream.
- Rendering a Flash movie requires a lot of drawing operations. Swfdec  
uses cairo for this. The only available solution for cairo in  
GStreamer is to use image buffers. Rendering to images is pretty much  
the slowest rendering method cairo offers.
- A lot of Flash files take 100% CPU and require frame dropping to get  
smooth audio playback and OK video. Back when I dropped the plugin,  
the QoS framework did not exist. I haven't looked if the GStreamer QoS  
would be up to the task now.
- Modern Flash relies a lot on user input. Particular in 0.8 there was  
a lot of buffering taking place inside pipelines, which made user  
input feel very unresponsive. I have no idea how much that is still  
the case in 0.10.
- The audio issues pointed out in 2) might be interesting, too, in  
particular when thinking about autoplugging.

Nonetheless I am very interested in a working GStreamer plugin for  
Swfdec. And I bet a lot of other people are, too. While designing the  
Swfdec API [6], I kept in mind that people want to use Swfdec inside  
GStreamer, so I hope there are no obstacles in implementing a  
GStreamer plugin. If there are, I'm of course open to change the API,  
since it's by no means frozen.
So if anyone is looking for an interesting project to do, this might  
be it. I'll gladly help everyone trying to make this happen, but I  
certainly do not have enough time to do it myself, since making Swfdec  
correctly decode SWF files takes up 150% of my time already.


Cheers,
Benjamin



[1]  
http://gitweb.freedesktop.org/?p=swfdec.git;a=blob;f=libswfdec/swfdec_codec.h
[2] http://brajeshwar.com/reference/as2/NetStream.html
[3] http://brajeshwar.com/reference/as2/Sound.html
[4] http://swfdec.freedesktop.org/documentation/swfdec/SwfdecAudio.html
[5]  
http://swfdec.freedesktop.org/documentation/swfdec/SwfdecPlayer.html#SwfdecPlayer-audio-added
[6] http://swfdec.freedesktop.org/documentation/swfdec/