bugfix mp3_stream, add ogg_stream, bugfix max_level

diff --git a/timeside/analyzer/max_level.py b/timeside/analyzer/max_level.py
index 58ea322b1a1d70bc3ea7614be8bc1f181c70f830..b97c9e217f404687ab273d9a6e2786b9782c805c 100644 (file)
--- a/timeside/analyzer/max_level.py
+++ b/timeside/analyzer/max_level.py
@@ -32,7 +32,7 @@ class MaxLevel(Processor):
     @interfacedoc
     def setup(self, channels=None, samplerate=None, nframes=None):
         super(MaxLevel, self).setup(channels, samplerate, nframes)
-        self.value = 0
+        self.value = -120
 
     @staticmethod
     @interfacedoc

diff --git a/timeside/encoder/__init__.py b/timeside/encoder/__init__.py
index 28d1c4d96f4a23b2bb443d756795994e7531d812..e53e08800022539258e8ee9bb7eb38fb761aca3c 100644 (file)
--- a/timeside/encoder/__init__.py
+++ b/timeside/encoder/__init__.py
@@ -4,7 +4,8 @@ from core import *
 from ogg import *
 from wav import *
 from mp3 import *
-from mp3_stream import *
 from m4a import *
+from mp3_stream import *
+from ogg_stream import *
 
 #from timeside.encoder.flac import *

diff --git a/timeside/encoder/mp3_stream.py b/timeside/encoder/mp3_stream.py
index e34352082a5b0e893f9fd789c26361b4bc1440e7..8b3de51aba165ef83972c0006deeb4ee6c67c5e9 100644 (file)
--- a/timeside/encoder/mp3_stream.py
+++ b/timeside/encoder/mp3_stream.py
@@ -37,15 +37,13 @@ class Mp3EncoderStream(Processor):
     implements(IEncoder)
 
     def __init__(self, output=None):
-#        self.file = 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
-        
+        # the output data format we want        
         pipe = '''appsrc name=src ! audioconvert 
                   ! lame name=enc vbr=0 bitrate=256 ! id3v2mux 
                   '''
@@ -117,4 +115,3 @@ class Mp3EncoderStream(Processor):
         """ gstreamer buffer to numpy array conversion """
         buf = gst.Buffer(getbuffer(frames))
         return buf
-

diff --git a/timeside/encoder/ogg_stream.py b/timeside/encoder/ogg_stream.py
new file mode 100644 (file)
index 0000000..7978f46
--- /dev/null
+++ b/timeside/encoder/ogg_stream.py
@@ -0,0 +1,115 @@
+# -*- 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