diff --git a/src/whisper_streaming/backends.py b/src/whisper_streaming/backends.py
index 682cfc5..99ba762 100644
--- a/src/whisper_streaming/backends.py
+++ b/src/whisper_streaming/backends.py
@@ -164,7 +164,7 @@ class MLXWhisper(ASRBase):
     Significantly faster than faster-whisper (without CUDA) on Apple M1.
     """
 
-    sep = " "
+    sep = "" # In my experience in french it should also be no space.
 
     def load_model(self, modelsize=None, cache_dir=None, model_dir=None):
         """
diff --git a/src/whisper_streaming/online_asr.py b/src/whisper_streaming/online_asr.py
index dc34fd8..c9e4258 100644
--- a/src/whisper_streaming/online_asr.py
+++ b/src/whisper_streaming/online_asr.py
@@ -166,7 +166,13 @@ class OnlineASRProcessor:
             if (
                 len(self.audio_buffer) / self.SAMPLING_RATE > self.buffer_trimming_sec
             ):  # longer than this
+                
+                logger.debug("chunking sentence")
                 self.chunk_completed_sentence()
+                
+
+            else:
+                logger.debug("not enough audio to trim as a sentence")
 
         if self.buffer_trimming_way == "segment":
             s = self.buffer_trimming_sec  # trim the completed segments longer than s,
@@ -194,7 +200,10 @@ class OnlineASRProcessor:
     def chunk_completed_sentence(self):
         if self.commited == []:
             return
-        logger.debug("COMPLETED SENTENCE: ", [s[2] for s in self.commited])
+        
+        import pdb; pdb.set_trace()
+        raw_text = self.asr.sep.join([s[2] for s in self.commited]) 
+        logger.debug(f"COMPLETED SENTENCE: {raw_text}")
         sents = self.words_to_sentences(self.commited)
         for s in sents:
             logger.debug(f"\t\tSENT: {s}")
@@ -243,7 +252,7 @@ class OnlineASRProcessor:
         """
 
         cwords = [w for w in words]
-        t = " ".join(o[2] for o in cwords)
+        t = self.asr.sep.join(o[2] for o in cwords)
         s = self.tokenize(t)
         out = []
         while s:
@@ -269,7 +278,7 @@ class OnlineASRProcessor:
         """
         o = self.transcript_buffer.complete()
         f = self.to_flush(o)
-        logger.debug(f"last, noncommited: {f}")
+        logger.debug(f"last, noncommited: {f[0]*1000:.0f}-{f[1]*1000:.0f}: {f[2]}")
         self.buffer_time_offset += len(self.audio_buffer) / 16000
         return f