return "dB"
def process(self, frames, eod=False):
- max = numpy.round(20*numpy.log(frames.max()), 3)
+ max = numpy.round(20*numpy.log(frames.max()), 2)
if max > self.value:
self.value = max
return frames, eod
return "%s %s" % (str(self.value), unit())
def process(self, frames, eod=False):
- value = numpy.round(20*numpy.log10(numpy.mean(numpy.sqrt(numpy.square(frames)))), 3)
+ value = numpy.round(20*numpy.log10(numpy.mean(numpy.sqrt(numpy.square(frames)))), 2)
if value > self.value:
self.value = value
return frames, eod
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"
+ self.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')
+ self.sink = self.pipeline.get_by_name('sink')
# adjust length of emitted buffers
# self.sink.set_property('blocksize', 0x10000)
# start pipeline
- pipeline.set_state(gst.STATE_PLAYING)
+ self.pipeline.set_state(gst.STATE_PLAYING)
@interfacedoc
def channels(self):
self.buffer_size = 0xFFFF
if not stdin:
- stdin = subprocess.PIPE
+ stdin = subprocess.PIPE
self.proc = subprocess.Popen(command.encode('utf-8'),
shell = True,
from wav import *
from mp3 import *
from m4a import *
-from mp3_stream import *
-from ogg_stream import *
#from timeside.encoder.flac import *
""" gstreamer-based mp3 encoder """
implements(IEncoder)
- def __init__(self, output):
- self.file = None
+ def __init__(self, output, streaming=False):
if isinstance(output, basestring):
self.filename = output
else:
- raise Exception("Streaming not supported")
+ self.filename = None
+ self.streaming = streaming
+
+ if not self.filename and self.streaming:
+ raise Exception('Must give an output')
@interfacedoc
def setup(self, channels=None, samplerate=None, nframes=None):
super(Mp3Encoder, self).setup(channels, samplerate, nframes)
#TODO: open file for writing
- # the output data format we want
- pipeline = gst.parse_launch(''' appsrc name=src
- ! audioconvert
- ! lame name=enc vbr=0 bitrate=256 ! id3v2mux
- ! filesink location=%s ''' % self.filename)
+ # the output data format we want
+ pipe = ''' appsrc name=src max-bytes=32768 block=true
+ ! audioconvert
+ ! lame name=enc vbr=0 bitrate=256 ! id3v2mux
+ '''
+ if self.filename and self.streaming:
+ pipe += '''
+ ! queue2 name=q0 ! tee name=tee
+ tee. ! queue name=q1 ! appsink name=app
+ tee. ! queue name=q2 ! filesink location=%s
+ ''' % self.filename
+
+ elif self.filename :
+ pipe += '! filesink location=%s' % self.filename
+ else:
+ pipe += '! appsink name=app'
+
+ self.pipeline = gst.parse_launch(pipe)
+ # store a pointer to appsrc in our encoder object
+ self.src = self.pipeline.get_by_name('src')
# store a pointer to appsink in our encoder object
- self.src = pipeline.get_by_name('src')
+ self.app = self.pipeline.get_by_name('app')
+
srccaps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
self.src.set_property("caps", srccaps)
# start pipeline
- pipeline.set_state(gst.STATE_PLAYING)
- self.pipeline = pipeline
+ self.pipeline.set_state(gst.STATE_PLAYING)
@staticmethod
@interfacedoc
def process(self, frames, eod=False):
buf = self.numpy_array_to_gst_buffer(frames)
self.src.emit('push-buffer', buf)
+ if self.streaming:
+ frames = self.app.emit('pull-buffer')
if eod: self.src.emit('end-of-stream')
return frames, eod
+++ /dev/null
- # -*- coding: utf-8 -*-
-#
-# Copyright (C) 2007 Parisson SARL
-# Copyright (c) 2006-2007 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/>.
-
-# Authors: Guillaume Pellerin <yomguy@parisson.com>
-# Paul Brossier <piem@piem.org>
-
-from timeside.core import Processor, implements, interfacedoc
-from timeside.api import IEncoder
-from numpy import array, frombuffer, getbuffer, float32
-
-import pygst
-pygst.require('0.10')
-import gst
-import gobject
-gobject.threads_init()
-
-
-class Mp3EncoderStream(Processor):
- """ gstreamer-based streaming mp3 encoder with an appsink tee"""
- implements(IEncoder)
-
- def __init__(self, output=None):
- self.filename = output
-
- @interfacedoc
- def setup(self, channels=None, samplerate=None, nframes=None):
- super(Mp3EncoderStream, self).setup(channels, samplerate, nframes)
- #TODO: open file for writing
- # the output data format we want
- pipe = '''appsrc name=src ! audioconvert
- ! lame name=enc vbr=0 bitrate=256 ! id3v2mux
- '''
- if self.filename:
- pipe += '''
- ! queue2 name=q0 ! tee name=tee
- tee. ! queue name=q1 ! appsink name=app
- tee. ! queue name=q2 ! filesink location=%s
- ''' % self.filename
- else:
- pipe += '! appsink name=app'
-
- pipeline = gst.parse_launch(pipe)
- # store a pointer to appsrc in our encoder object
- self.src = pipeline.get_by_name('src')
- # store a pointer to appsink in our encoder object
- self.app = pipeline.get_by_name('app')
-
- 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 "gst_mp3_enc_stream"
-
- @staticmethod
- @interfacedoc
- def description():
- return "MP3 GStreamer based encoder and streamer"
-
- @staticmethod
- @interfacedoc
- def format():
- return "MP3"
-
- @staticmethod
- @interfacedoc
- def file_extension():
- return "mp3"
-
- @staticmethod
- @interfacedoc
- def mime_type():
- return "audio/mpeg"
-
- @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)
- appbuffer = self.app.emit('pull-buffer')
- if eod: self.src.emit('end-of-stream')
- return appbuffer, eod
-
- def numpy_array_to_gst_buffer(self, frames):
- """ gstreamer buffer to numpy array conversion """
- buf = gst.Buffer(getbuffer(frames))
- return buf
""" gstreamer-based vorbis encoder """
implements(IEncoder)
- def __init__(self, output):
- self.file = None
+ def __init__(self, output, streaming=False):
if isinstance(output, basestring):
self.filename = output
else:
- raise Exception("Streaming not supported")
+ self.filename = None
+ self.streaming = streaming
+
+ if not self.filename and self.streaming:
+ raise Exception('Must give an output')
@interfacedoc
def setup(self, channels=None, samplerate=None, nframes=None):
super(VorbisEncoder, self).setup(channels, samplerate, nframes)
# TODO open file for writing
- # the output data format we want
- pipeline = gst.parse_launch(''' appsrc name=src
- ! audioconvert
- ! vorbisenc
- ! oggmux
- ! filesink location=%s ''' % self.filename)
+ # the output data format we want
+ pipe = ''' appsrc name=src max-bytes=32768 block=true
+ ! audioconvert
+ ! vorbisenc
+ ! oggmux
+ '''
+ if self.filename and self.streaming:
+ pipe += '''
+ ! queue2 name=q0 ! tee name=tee
+ tee. ! queue name=q1 ! appsink name=app
+ tee. ! queue name=q2 ! filesink location=%s
+ ''' % self.filename
+
+ elif self.filename :
+ pipe += '! filesink location=%s' % self.filename
+ else:
+ pipe += '! appsink name=app'
+
+ self.pipeline = gst.parse_launch(pipe)
+ # store a pointer to appsrc in our encoder object
+ self.src = self.pipeline.get_by_name('src')
# store a pointer to appsink in our encoder object
- self.src = pipeline.get_by_name('src')
+ self.app = self.pipeline.get_by_name('app')
+
srccaps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
self.src.set_property("caps", srccaps)
# start pipeline
- pipeline.set_state(gst.STATE_PLAYING)
- self.pipeline = pipeline
+ self.pipeline.set_state(gst.STATE_PLAYING)
@staticmethod
@interfacedoc
@interfacedoc
def set_metadata(self, metadata):
- #TODO
+ #TODO:
pass
@interfacedoc
def process(self, frames, eod=False):
buf = self.numpy_array_to_gst_buffer(frames)
self.src.emit('push-buffer', buf)
+ if self.streaming:
+ frames = self.app.emit('pull-buffer')
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
-
-
-
+++ /dev/null
-# -*- coding: utf-8 -*-
-#
-# Copyright (c) 2010 Paul Brossier <piem@piem.org>
-# Copyright (c) 2010 Guillaume Pellerin <yomguy@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/>.
-
-
-from timeside.core import Processor, implements, interfacedoc
-from timeside.api import IEncoder
-from numpy import array, frombuffer, getbuffer, float32
-
-import pygst
-pygst.require('0.10')
-import gst
-import gobject
-gobject.threads_init()
-
-
-class VorbisEncoderStream(Processor):
- """ gstreamer-based vorbis encoder """
- implements(IEncoder)
-
- def __init__(self, output=None):
- self.filename = output
-
- @interfacedoc
- def setup(self, channels=None, samplerate=None, nframes=None):
- super(VorbisEncoderStream, self).setup(channels, samplerate, nframes)
- # TODO open file for writing
- # the output data format we want
- pipe = '''appsrc name=src ! audioconvert
- ! vorbisenc ! oggmux
- '''
- if self.filename:
- pipe += '''
- ! queue2 name=q0 ! tee name=tee
- tee. ! queue name=q1 ! appsink name=app
- tee. ! queue name=q2 ! filesink location=%s
- ''' % self.filename
- else:
- pipe += '! appsink name=app'
-
- pipeline = gst.parse_launch(pipe)
- # store a pointer to appsrc in our encoder object
- self.src = pipeline.get_by_name('src')
- # store a pointer to appsink in our encoder object
- self.app = pipeline.get_by_name('app')
-
- 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 "gst_vorbis_enc_stream"
-
- @staticmethod
- @interfacedoc
- def description():
- return "Vorbis GStreamer based encoder and streamer"
-
- @staticmethod
- @interfacedoc
- def format():
- return "OGG"
-
- @staticmethod
- @interfacedoc
- def file_extension():
- return "ogg"
-
- @staticmethod
- @interfacedoc
- def mime_type():
- return "application/ogg"
-
- @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)
- appbuffer = self.app.emit('pull-buffer')
- if eod: self.src.emit('end-of-stream')
- return appbuffer, eod
-
- def numpy_array_to_gst_buffer(self, frames):
- """ gstreamer buffer to numpy array conversion """
- buf = gst.Buffer(getbuffer(frames))
- return buf
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
+ self.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')
+ self.src = self.pipeline.get_by_name('src')
srccaps = gst.Caps("""audio/x-raw-float,
endianness=(int)1234,
channels=(int)%s,
self.src.set_property("caps", srccaps)
# start pipeline
- pipeline.set_state(gst.STATE_PLAYING)
- self.pipeline = pipeline
+ self.pipeline.set_state(gst.STATE_PLAYING)
@staticmethod
@interfacedoc
from timeside.decoder import *
from timeside.encoder import *
+import os.path
-source = "../samples/sweep.wav"
-dest = "../results/sweep_wav.mp3"
+source = os.path.join(os.path.dirname(__file__), "../samples/sweep.wav")
+dest = os.path.join(os.path.dirname(__file__), "../results/sweep_wav.mp3")
decoder = FileDecoder(source)
encoder = Mp3Encoder(dest)
from timeside.decoder import *
from timeside.encoder import *
+import os.path
-source = "../samples/sweep.wav"
-dest = "../results/sweep_wav.ogg"
+source = os.path.join(os.path.dirname(__file__), "../samples/sweep.wav")
+dest = os.path.join(os.path.dirname(__file__), "../results/sweep_wav.ogg")
decoder = FileDecoder(source)
encoder = VorbisEncoder(dest)
projects / timeside.git / commitdiff