[gst-devel] typefind and modifying a running stream (python bindings)

[Daniel Lenski]

Hi all,
Sorry if this is the wrong place to ask questions about code written with
the Python bindings... there doesn't seem to be a python-gst mailing list,
so I figured I'd give this a shot.

I'm new to gstreamer and trying to write some code that will slurp a stream
off the web into a file, and transcode it to MP3 if not already in that
format.  The application is to automatically convert it to a form that most
portable audio players can understand.

I have some working code, but not exactly sure "why" it works ;-)
Basically, what I do is I set up a gnomevfssrc ! typefind pipeline, and
connect the have-type signal of the typefind element.  Then I start the
pipeline playing.  When the have-type signal is received, I pause the
pipeline, and then:
  (a) if the detected stream is MP3, I make an identity element as the
transcoder: transcoder = identity
  (b) if it's not MP3, I make a real transcoder: transcoder = decodebin !
audioconvert ! lame
Next, I link this transcoder to the end of the pipeline and link that to a
filesink, so I end up with:
  gnomevfssrc ! typefind ! transcoder ! filesink
Finally, I unpause the pipeline, and the MP3 output stream magically goes
into the output file.  Hooray!

Here are a few things I'm trying to understand:

* Are there any subtleties to modifying a running pipeline?  When I pause
the pipeline, append to it, and then unpause it, should I be worried about
data getting lost, or buffer overflows, or anything like that?  This doesn't
seem to be a problem, but maybe I've just got lucky.

* If I add() the transcoder and sink elements to the pipeline BEFORE I'm
ready to actually link their pads, then the pipeline runs, and typefind
works... but I get no output.  Why?

* Typefind identifies MP3 files as application/id3 sometimes, rather than
audio/mpeg.  Is there a more correct and reliable way to identify an MP3
stream?

* How can I identify the duration of the stream processed by the pipeline?
pipeline.get_clock() seems to keep track of the REAL time, rather than the
amount of audio processed.

Any help will be appreciated!  Thanks a lot!

Dan Lenski


PS- The relevant parts of my code so far:

class transcode_to_mp3(gst.Bin):
    # a bin containing decodebin, audioconvert, and lame
    # to auto-convert any audio stream to MP3
    ...

class StreamRecorder:
    def __init__(self, uri, filename):
        # stream reader
        self.stream = gst.element_factory_make('gnomevfssrc')
        self.stream.set_property('location', uri)

        # automatically identify stream type
        self.typefind = gst.element_factory_make('typefind')
        self.typefind.set_property('minimum', 100)
        self.typefind.connect('have-type', self.autoconfigure)

        # convert arbitrary audio stream to MP3, or do nothing to an MP3
        self.transcode = transcode_to_mp3()
        self.identity = gst.element_factory_make('identity')

        # write to file output
        self.sink = gst.element_factory_make('filesink')
        self.sink.set_property('location', filename)

        # partially assemble pipeline
        self.pipeline = gst.Pipeline()
        self.pipeline.add(self.stream, self.typefind)
        self.stream.link(self.typefind)

    def autoconfigure(self, tf, prob, caps):
        self.pipeline.set_state(gst.STATE_PAUSED)

        ismp3 = (caps[0].get_name() == 'audio/mpeg' and caps[0]['layer']==3)
        if ismp3: stage = self.identity
        else: stage = self.mp3ify

        self.pipeline.add(stage, self.sink)
        gst.element_link_many(self.typefind, stage, self.sink)

        self.pipeline.set_state(gst.STATE_PLAYING)

    def start(self):
        self.pipeline.set_state(gst.STATE_PLAYING)

    def stop(self):
        self.pipeline.set_state(gst.STATE_NULL)
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