tests/api/gstreamer.py: remove old code

diff --git a/tests/api/gstreamer.py b/tests/api/gstreamer.py
deleted file mode 100644 (file)
index 897ac42..0000000
--- a/tests/api/gstreamer.py
+++ /dev/null
@@ -1,238 +0,0 @@
-# -*- coding: utf-8 -*-
-#
-# Copyright (C) 2007-2009 Parisson
-# Copyright (c) 2007 Olivier Guilyardi <olivier@samalyse.com>
-# Copyright (c) 2007-2009 Guillaume Pellerin <pellerin@parisson.com>
-#
-# This file is part of TimeSide.
-
-# TimeSide is free software: you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation, either version 2 of the License, or
-# (at your option) any later version.
-
-# TimeSide is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-# GNU General Public License for more details.
-
-# You should have received a copy of the GNU General Public License
-# along with TimeSide.  If not, see <http://www.gnu.org/licenses/>.
-
-# Author: Paul Brossier <piem@piem.org>
-
-from timeside.core import Processor, implements, interfacedoc
-from timeside.api import IDecoder, IEncoder
-from numpy import array, frombuffer, getbuffer, float32
-
-import pygst
-pygst.require('0.10')
-import gst
-import gobject
-gobject.threads_init ()
-
-class FileDecoder(Processor):
-    """ gstreamer-based decoder """
-    implements(IDecoder)
-
-    # duration ms, before discovery process times out
-    MAX_DISCOVERY_TIME = 3000
-
-    audioformat = None
-    audiochannels = None
-    audiorate = None
-    audionframes = None
-    mimetype = ''
-
-    # IProcessor methods
-
-    @staticmethod
-    @interfacedoc
-    def id():
-        return "test_gstreamerdec"
-
-    def setup(self, channels = None, samplerate = None, nframes = None):
-        # the output data format we want
-        caps = "audio/x-raw-float, width=32"
-        pipeline = gst.parse_launch('''uridecodebin uri=%s
-            ! audioconvert
-            ! %s
-            ! appsink name=sink sync=False ''' % (self.uri, caps))
-        # store a pointer to appsink in our decoder object 
-        self.sink = pipeline.get_by_name('sink')
-        # adjust length of emitted buffers
-        # self.sink.set_property('blocksize', 0x10000)
-        # start pipeline
-        pipeline.set_state(gst.STATE_PLAYING)
-
-    @interfacedoc
-    def channels(self):
-        return  self.audiochannels
-
-    @interfacedoc
-    def samplerate(self):
-        return self.audiorate 
-
-    @interfacedoc
-    def nframes(self):
-        return self.audionframes
-
-    @interfacedoc
-    def process(self, frames = None, eod = False):
-        try:
-            buf = self.sink.emit('pull-buffer')
-        except SystemError, e:
-            # should never happen
-            print 'SystemError', e
-            return array([0.]), True
-        if buf == None:
-            return array([0.]), True
-        return self.gst_buffer_to_numpy_array(buf), False
-
-    @interfacedoc
-    def release(self):
-        # nothing to do for now 
-        pass
-
-    ## IDecoder methods
-
-    @interfacedoc
-    def __init__(self, uri):
-
-        # is this a file? 
-        import os.path
-        if os.path.exists(uri):
-            # get the absolute path
-            uri = os.path.abspath(uri)
-            # first run the file/uri through the discover pipeline
-            self.discover(uri)
-            # and make a uri of it
-            from urllib import quote
-            self.uri = 'file://'+quote(uri)
-
-    @interfacedoc
-    def format(self):
-        # TODO check
-        return self.mimetype
-
-    @interfacedoc
-    def encoding(self):
-        # TODO check
-        return self.mimetype.split('/')[-1]
-
-    @interfacedoc
-    def resolution(self):
-        # TODO check: width or depth?
-        return self.audiowidth 
-
-    @interfacedoc
-    def metadata(self):
-        # TODO check
-        return self.tags
-
-    ## gst.extend discoverer
-
-    def discover(self, path):
-        """ gstreamer based helper function to get file attributes """
-        from gst.extend import discoverer
-        d = discoverer.Discoverer(path, timeout = self.MAX_DISCOVERY_TIME)
-        d.connect('discovered', self.discovered)
-        self.mainloop = gobject.MainLoop()
-        d.discover()
-        self.mainloop.run()
-
-    def discovered(self, d, is_media):
-        """ gstreamer based helper executed upon discover() completion """
-        if is_media and d.is_audio:
-            # copy the discoverer attributes to self
-            self.audiorate = d.audiorate
-            self.mimetype= d.mimetype
-            self.audiochannels = d.audiochannels
-            self.audiowidth = d.audiowidth
-            # conversion from time in nanoseconds to frames 
-            from math import ceil
-            duration = d.audiorate * d.audiolength * 1.e-9
-            self.audionframes = int (ceil ( duration ) )
-            self.tags = d.tags
-        elif not d.is_audio:
-            print "error, no audio found!"
-        else:
-            print "fail", path
-        self.mainloop.quit()
-
-    def gst_buffer_to_numpy_array(self, buf):
-        """ gstreamer buffer to numpy array conversion """
-        chan = self.audiochannels
-        samples = frombuffer(buf.data, dtype=float32) 
-        samples.resize([len(samples)/chan, chan])
-        return samples
-
-class WavEncoder(Processor):
-    """ gstreamer-based encoder """
-    implements(IEncoder)
-
-    def __init__(self, output):
-        self.file = None
-        if isinstance(output, basestring):
-            self.filename = output
-        else:
-            raise Exception("Streaming not supported")
-
-    @interfacedoc
-    def setup(self, channels=None, samplerate=None, nframes=None):
-        super(WavEncoder, self).setup(channels, samplerate, nframes)
-        # TODO open file for writing
-        # the output data format we want
-        pipeline = gst.parse_launch(''' appsrc name=src
-            ! audioconvert
-            ! wavenc
-            ! filesink location=%s ''' % self.filename)
-        # store a pointer to appsink in our encoder object 
-        self.src = pipeline.get_by_name('src')
-        srccaps = gst.Caps("""audio/x-raw-float,
-            endianness=(int)1234,
-            channels=(int)%s,
-            width=(int)32,
-            rate=(int)%d""" % (int(channels), int(samplerate)))
-        self.src.set_property("caps", srccaps)
-
-        # start pipeline
-        pipeline.set_state(gst.STATE_PLAYING)
-        self.pipeline = pipeline
-
-    @staticmethod
-    @interfacedoc
-    def id():
-        return "test_gstreamerenc"
-
-    @staticmethod
-    @interfacedoc
-    def description():
-        return "Gstreamer based encoder"
-
-    @staticmethod
-    @interfacedoc
-    def file_extension():
-        return "wav"
-
-    @staticmethod
-    @interfacedoc
-    def mime_type():
-        return "audio/x-wav"
-
-    @interfacedoc
-    def set_metadata(self, metadata):
-        #TODO
-        pass
-
-    @interfacedoc
-    def process(self, frames, eod=False):
-        buf = self.numpy_array_to_gst_buffer(frames)
-        self.src.emit('push-buffer', buf)
-        if eod: self.src.emit('end-of-stream')
-        return frames, eod
-
-    def numpy_array_to_gst_buffer(self, frames):
-        """ gstreamer buffer to numpy array conversion """
-        buf = gst.Buffer(getbuffer(frames))
-        return buf