try to get encoders working but some are blocking : the output buffer size of the...

DON'T use encoders for now :(

diff --git a/timeside/decoder/core.py b/timeside/decoder/core.py
index b5cb9c937e7ec9bc30fe8cf91f784f7080a1abfd..66540b01cfdbd206e8f8d0e8dda58e5b695cd3ca 100644 (file)
--- a/timeside/decoder/core.py
+++ b/timeside/decoder/core.py
@@ -61,8 +61,9 @@ class FileDecoder(Processor):
             ! appsink name=sink sync=False ''' % (self.uri, caps))
         # store a pointer to appsink in our decoder object
         self.sink = self.pipeline.get_by_name('sink')
+        self.sink.set_property('emit-signals', True)
         # adjust length of emitted buffers
-        # self.sink.set_property('blocksize', 0x10000)
+        self.sink.set_property('blocksize', 0x8000)
         # start pipeline
         self.pipeline.set_state(gst.STATE_PLAYING)
 

diff --git a/timeside/encoder/flac.py b/timeside/encoder/flac.py
index dce1caa702c118c64037bcd319357ba627ea5a6f..0f177a12855b3e17dc4e4d7c243856253548a645 100644 (file)
--- a/timeside/encoder/flac.py
+++ b/timeside/encoder/flac.py
@@ -60,7 +60,7 @@ class FlacEncoder(Processor):
         elif self.filename :
             self.pipe += '! filesink location=%s ' % self.filename
         else:
-            self.pipe += '! appsink name=sink blocksize=16384 sync=False '
+            self.pipe += '! appsink name=sink sync=False '
             
         self.pipeline = gst.parse_launch(self.pipe)
         # store a pointer to appsrc in our encoder object

diff --git a/timeside/encoder/mp3.py b/timeside/encoder/mp3.py
index bbe97a6fe7b4af7e58c5d4909fa3f1aa1b884ff7..919174976d0a0b3a1af897183dbbb8b0efbbd192 100644 (file)
--- a/timeside/encoder/mp3.py
+++ b/timeside/encoder/mp3.py
@@ -31,6 +31,7 @@ import pygst
 pygst.require('0.10')
 import gst
 import gobject
+import Queue
 gobject.threads_init()
 
 
@@ -48,8 +49,10 @@ class Mp3Encoder(Processor):
         if not self.filename and not self.streaming:
             raise Exception('Must give an output')
         
-        self.max_bytes = 65536
-        self.size_adapter = self.max_bytes
+        self.eod = False
+        self.buffer_size = 8192
+        self.max_bytes = 4096
+        self.adapter_size = self.max_bytes
         
     @interfacedoc
     def setup(self, channels=None, samplerate=None, nframes=None):
@@ -57,12 +60,12 @@ class Mp3Encoder(Processor):
         super(Mp3Encoder, self).setup(channels, samplerate, nframes)
         #TODO: open file for writing
         # the output data format we want        
-        self.pipe = '''appsrc name=src max-bytes=%s
+        self.pipe = '''appsrc name=src
                   ! audioconvert 
                   ! lamemp3enc bitrate=256 ! id3v2mux
-                  ''' % self.max_bytes
+                  '''
         if self.filename and self.streaming:
-            self.pipe += ''' ! queue ! tee name=t
+            self.pipe += ''' ! tee name=t
             ! queue ! filesink location=%s
             t. ! queue ! appsink name=app sync=False
             ''' % self.filename
@@ -87,9 +90,16 @@ class Mp3Encoder(Processor):
 
         # start pipeline
         if self.streaming:
+            self.queue = Queue.Queue(self.buffer_size)
             self.app.set_property('emit-signals', True)
+            self.app.connect("new-buffer", self.buffer)
+            
         self.pipeline.set_state(gst.STATE_PLAYING)
-        self.adapter = FixedSizeInputAdapter(self.size_adapter, channels, pad=True)
+        self.adapter = FixedSizeInputAdapter(self.adapter_size, channels, pad=True)
+    
+    def buffer(self, appsink):
+        data = appsink.props.last_buffer.data
+        self.queue.put_nowait(data)
         
     @staticmethod
     @interfacedoc
@@ -123,22 +133,21 @@ class Mp3Encoder(Processor):
 
     @interfacedoc
     def process(self, frames, eod=False):
+        self.eod = eod
         print frames.shape
-        emit = 0
-        for samples, end in self.adapter.process(frames, eod):
-            print 'push: ', samples.shape
-            buf = self.numpy_array_to_gst_buffer(samples)
-            self.src.emit('push-buffer', buf)
-            if self.streaming:
-                pull = self.app.emit('pull-buffer')
-                emit = 1
+        
+        buf = self.numpy_array_to_gst_buffer(frames)
+        self.src.emit('push-buffer', buf)
+
+#        for samples, end in self.adapter.process(frames, eod):
+#            print 'push: ', samples.shape
+#            buf = self.numpy_array_to_gst_buffer(samples)
+#            self.src.emit('push-buffer', buf)
+        
         if self.streaming:
-            if emit:
-                return pull, eod
-            else:
-                return None, eod
-        else:
-            return frames, eod
+            self.chunk = self.queue.get(self.buffer_size)
+            
+        return frames, eod
         
     def numpy_array_to_gst_buffer(self, frames):
         """ gstreamer buffer to numpy array conversion """
@@ -146,8 +155,8 @@ class Mp3Encoder(Processor):
         return buf
 
 
-class Mp3EncoderOld:
-    """Defines methods to encode to MP3"""
+class Mp3EncoderSubprocess(object):
+    """MP3 encoder in a subprocess pipe"""
 
 #    implements(IEncoder)
     
@@ -172,26 +181,42 @@ class Mp3EncoderOld:
                              'publisher': 'tc', #comment
                              'date': 'ty', #year
                              }
-
+    
+    @interfacedoc
+    def setup(self, channels=None, samplerate=None, nframes=None):
+        self.channels = channels
+        super(Mp3EncoderSubprocess, self).setup(channels, samplerate, nframes)
+        
     @staticmethod
+    @interfacedoc
     def id():
-        return "mp3enc"
+        return "subprocess_mp3_enc"
 
-    def format(self):
-        return 'MP3'
+    @staticmethod
+    @interfacedoc
+    def description():
+        return "MP3 subprocess based encoder"
 
-    def file_extension(self):
-        return 'mp3'
+    @staticmethod
+    @interfacedoc
+    def format():
+        return "MP3"
 
-    def mime_type(self):
-        return 'audio/mpeg'
+    @staticmethod
+    @interfacedoc
+    def file_extension():
+        return "mp3"
 
-    def description(self):
-        return """
-        MPEG-1 Audio Layer 3, more commonly referred to as MP3, is a patented
-        digital audio encoding format using a form of lossy data compression.
-        """
+    @staticmethod
+    @interfacedoc
+    def mime_type():
+        return "audio/mpeg"
 
+    @interfacedoc
+    def set_metadata(self, metadata):
+        #TODO: 
+        pass
+  
     def get_file_info(self):
         try:
             file_out1, file_out2 = os.popen4('mp3info "'+self.dest+'"')

diff --git a/timeside/encoder/ogg.py b/timeside/encoder/ogg.py
index 981bbcb82151879b124e1df6ef7289ddc35a6372..3da76c6622fb6a1b3b36b59f46322e925386a658 100644 (file)
--- a/timeside/encoder/ogg.py
+++ b/timeside/encoder/ogg.py
@@ -118,7 +118,7 @@ class VorbisEncoder(Processor):
         self.src.emit('push-buffer', buf)
         if self.streaming:
             pull = self.app.emit('pull-buffer')
-        if eod: self.src.emit('end-of-stream')
+#        if eod: self.src.emit('end-of-stream')
         if not self.streaming:
             return frames, eod
         else:

diff --git a/timeside/tests/api/test_lolevel_streaming.py b/timeside/tests/api/test_lolevel_streaming.py
index 8bc444fed90d531fa512062699be520b8cd6a1b5..1bca2500508db835f9214545c80d87d35b7bc293 100644 (file)
--- a/timeside/tests/api/test_lolevel_streaming.py
+++ b/timeside/tests/api/test_lolevel_streaming.py
@@ -11,7 +11,7 @@ if len(sys.argv) > 1:
     source = sys.argv[1]
 else:
     import os.path
-    source= os.path.join (os.path.dirname(__file__),  "../samples/sweep.wav")
+    source= os.path.join (os.path.dirname(__file__),  "../samples/sweep_source.wav")
 
 decoder = FileDecoder(source)
 print "Creating decoder with id=%s for: %s" % (decoder.id(), source)
@@ -21,19 +21,19 @@ print 'channels :', channels
 samplerate = decoder.samplerate()
 nframes = decoder.nframes()
 
-dest1 = "/tmp/test.mp3"
-dest2 = "/tmp/test2.mp3"
+dest1 = "/tmp/test_filesink.mp3"
+dest2 = "/tmp/test_appsink.mp3"
 f = open(dest2,'w')
 
 streaming=True
-encoder = Mp3Encoder(dest1, streaming=streaming)
+encoder = Mp3Encoder(dest1, streaming=True)
 encoder.setup(channels=channels, samplerate=samplerate)
 
 while True:
-    buf, eod = encoder.process(*decoder.process())
-    if streaming and buf:
-        f.write(buf)
-    if eod :
+    encoder.process(*decoder.process())
+    if streaming:
+        f.write(encoder.chunk)
+    if encoder.eod :
         break
 
 f.close()

diff --git a/timeside/tests/api/test_lolevel_streaming_bad.py b/timeside/tests/api/test_lolevel_streaming_bad.py
new file mode 100644 (file)
index 0000000..e414ead
--- /dev/null
+++ b/timeside/tests/api/test_lolevel_streaming_bad.py
@@ -0,0 +1,42 @@
+# -*- coding: utf-8 -*-
+
+from timeside.core import *
+from timeside.decoder import *
+from timeside.analyzer import *
+from timeside.encoder import *
+from timeside.api import *
+
+import sys
+if len(sys.argv) > 1:
+    source = sys.argv[1]
+else:
+    import os.path
+    source= os.path.join (os.path.dirname(__file__),  "../samples/sweep.wav")
+
+decoder = FileDecoder(source)
+print "Creating decoder with id=%s for: %s" % (decoder.id(), source)
+decoder.setup()
+channels  = decoder.channels()
+print 'channels :', channels
+samplerate = decoder.samplerate()
+nframes = decoder.nframes()
+
+dest1 = "/tmp/test_filesink.mp3"
+dest2 = "/tmp/test_appsink.mp3"
+f = open(dest2,'w')
+
+streaming=True
+encoder = Mp3Encoder(dest1, streaming=True)
+encoder.setup(channels=channels, samplerate=samplerate)
+
+print encoder.pipe
+
+while True:
+    encoder.process(*decoder.process())
+    if streaming:
+        f.write(encoder.chunk)
+    if encoder.eod :
+        break
+
+f.close()
+print encoder.pipe