diff --git a/.vscode/launch.json b/.vscode/launch.json
index 7774467..889dcfc 100644
--- a/.vscode/launch.json
+++ b/.vscode/launch.json
@@ -10,7 +10,8 @@
             "type": "debugpy",
             "request": "launch",
             "program": "${file}",
-            "console": "integratedTerminal"
+            "console": "integratedTerminal",
+            "justMyCode": true
         }
     ]
 }
\ No newline at end of file
diff --git a/.vscode/tasks.json b/.vscode/tasks.json
index fa6c685..43682fb 100644
--- a/.vscode/tasks.json
+++ b/.vscode/tasks.json
@@ -13,6 +13,11 @@
             "label": "pip install -e bumble",
             "type": "shell",
             "command": "./venv/bin/python -m pip install -e ../bumble --config-settings editable_mode=compat"
+        },
+        {
+            "label": "pip install -e auracast",
+            "type": "shell",
+            "command": "./venv/bin/python -m pip install -e ../bumble-auracast --config-settings editable_mode=compat"
         }
     ]
 }
\ No newline at end of file
diff --git a/multilang_translator/config.py b/multilang_translator/config.py
index 806dbef..e54ca61 100644
--- a/multilang_translator/config.py
+++ b/multilang_translator/config.py
@@ -7,8 +7,6 @@ FRAME_DUR_MS = 10
 SAMPLING_RATE_HZ = int(16e3)
 BITRATE_BPS = int(32e3)
 
-def mk_filename_lc3(lang=''):
-    return f"{ANNOUNCEMENT_DIR}/announcement_{lang}_{SAMPLING_RATE_HZ//1000}_{FRAME_DUR_MS}_{BITRATE_BPS//1000}.lc3"
 
 LANG_CONFIG = {
     "de": {
diff --git a/multilang_translator/main.py b/multilang_translator/main.py
index a7fecc7..a9adc42 100644
--- a/multilang_translator/main.py
+++ b/multilang_translator/main.py
@@ -32,34 +32,34 @@ def syntesize_resample(text, tts_model, file_wav, file_wav_resamp):
     return audio_dur
 
 
-def translate_from_german(text_de, model):
+async def announcement_from_german_text(
+        caster: multicast_control.Multicaster,
+        text_de
+        ):
+    TRANSLATOR_LLM = 'llama3.2:3b-instruct-q4_0'
+
     config = copy(LANG_CONFIG)
     base_lang = "de"
 
-    file = config[base_lang]["filepath_wav"]
-    file_resamp = config[base_lang]['filepath_wav_resamp']
-    tts_json = {}
-
-    for key, val in config.items():
+    for i, d in enumerate(config.items()):
+        key, val = d
         if key == base_lang:
             text = text_de
         else:
-            text = llm_translator.translate_de_to_x(text_de, key, model=model)
-        
+            text = llm_translator.translate_de_to_x(text_de, key, model=TRANSLATOR_LLM)
+
         log.info('%s', text)
-        file = val['filepath_wav']
-        file_resamp = val['filepath_wav_resamp']
-        tts_json[key] = syntesize_resample(text, val['tts'], file, file_resamp)
 
-    return tts_json
-
-
-async def announcement_from_german_text(caster:multicast_control.Multicaster, text_de):
-
-    tts_json = translate_from_german(text_de, model = 'llama3.2:3b-instruct-q4_0')
+        lc3_audio = text_to_speech.synthesize(
+            text,
+            SAMPLING_RATE_HZ,
+            'piper',
+            val['tts'],
+            return_lc3=True
+            )
+        caster.big_conf[i].audio_source = lc3_audio
 
     start = time.time()
-    await caster.init_audio()
     caster.start_streaming()
 
     log.info("Starting all broadcasts took %s s", round(time.time() - start, 3))
@@ -74,7 +74,7 @@ async def command_line_ui(caster: multicast_control.Multicaster):
         prompt += "\n".join([f"{i}: {s}" for i,s in enumerate(sentence_list)])
         prompt += "\n"
         command = await aioconsole.ainput(prompt)
-    
+
         if command.strip().lower() == "quit":
             print("👋 Exiting...")
             if caster.device:
@@ -103,25 +103,22 @@ async def main():
     #global_conf.transport='serial:/dev/serial/by-id/usb-SEGGER_J-Link_001057705357-if02,1000000,rtscts' # transport for nrf54l15dk
     global_conf.transport='serial:/dev/serial/by-id/usb-ZEPHYR_Zephyr_HCI_UART_sample_81BD14B8D71B5662-if00,115200,rtscts' #nrf52dongle hci_uart usb cdc
 
-    big_conf = [ 
+    big_conf = [
         auracast_config.broadcast_de,
         auracast_config.broadcast_en,
         auracast_config.broadcast_fr,
         #auracast_config.broadcast_es,
         #auracast_config.broadcast_it,
         ]
-    files = [v['filepath_wav_resamp'] for v in LANG_CONFIG.values()]
     for i, conf in enumerate(big_conf):
-        conf.loop_wav = False
-        conf.audio_source = f'file:{files[i]}'
+        conf.loop = False
 
     caster = multicast_control.Multicaster(global_conf, big_conf)
     await caster.init_broadcast()
-    
-    #await announcement_from_german_text(caster, test_content.TESTSENTENCE_DE_HELLO)
 
-    await command_line_ui(caster)
+    #await announcement_from_german_text(caster, test_content.TESTSENTENCE.DE_HELLO)
     #await asyncio.wait([caster.streamer.task])
+    await command_line_ui(caster)
 
 if __name__ == '__main__':
     asyncio.run(main())
diff --git a/multilang_translator/text_to_speech/encode_lc3.py b/multilang_translator/text_to_speech/encode_lc3.py
new file mode 100644
index 0000000..c120771
--- /dev/null
+++ b/multilang_translator/text_to_speech/encode_lc3.py
@@ -0,0 +1,32 @@
+import numpy as np
+import lc3
+
+def encode(
+        audio: np.array,
+        output_sample_rate_hz,
+        octets_per_frame,
+        frame_duration_us=10000,
+        pcm_bit_depth = 16
+        ):
+
+    encoder = lc3.Encoder(
+        frame_duration_us=frame_duration_us,
+        sample_rate_hz=output_sample_rate_hz,
+        num_channels=1,
+        #input_sample_rate_hz=input_sample_rate,
+    )
+
+    lc3_frame_samples = encoder.get_frame_samples() # number of the pcm samples per lc3 frame
+
+    # reshape array into slices of lc3_frame_samples and padd with zeros
+    pad_width = (lc3_frame_samples - len(audio) % lc3_frame_samples) % lc3_frame_samples  # Compute padding length
+    arr_padded = np.pad(audio, (0, pad_width), mode='constant', constant_values=0)
+    reshaped_arr = arr_padded.reshape(-1, lc3_frame_samples)
+
+    lc3_bytes = b''
+    for pcm_frame in reshaped_arr:
+        lc3_bytes += encoder.encode(
+                pcm_frame, num_bytes=octets_per_frame, bit_depth=pcm_bit_depth
+            )
+
+    return lc3_bytes
\ No newline at end of file
diff --git a/multilang_translator/text_to_speech/text_to_speech.py b/multilang_translator/text_to_speech/text_to_speech.py
index d5895ba..5ce3dde 100644
--- a/multilang_translator/text_to_speech/text_to_speech.py
+++ b/multilang_translator/text_to_speech/text_to_speech.py
@@ -3,40 +3,80 @@ import subprocess
 import time
 import json
 import logging as log
+import numpy as np
 from multilang_translator import config
+from multilang_translator.utils.resample import resample_array
+from multilang_translator.text_to_speech import encode_lc3
 
 TTS_DIR = os.path.join(os.path.dirname(__file__))
 
-def synthesize(text, model="en_US-lessac-medium", output_file="out.wav"):
-
+def synth_piper(text, model="en_US-lessac-medium",):
     pwd = os.getcwd()
     os.chdir(TTS_DIR)
     start = time.time()
     
-    ret = subprocess.run( # TODO: wrap this whole thing in a class and open a permanent instance of the model 
-        [config.PIPER_EXE_PATH, '--model', model, '--output_file', output_file], 
+    ret = subprocess.run( # TODO: wrap this whole thing in a class and open a permanent pipe to the model 
+        [config.PIPER_EXE_PATH, 
+         '--cuda',
+         '--model', model, 
+         '--output-raw'], 
         input=text.encode('utf-8'),
         capture_output=True
         )
-    log.info('%s', ret.stdout)
-    log.info('%s', ret.stderr)
 
+    log.warning('Piper stderr:\n%s', ret.stderr)
     assert ret.returncode == 0, 'Piper returncode was not 0.'
+    audio = ret.stdout
 
     log.info("Running piper for model %s took %s s", model, round(time.time() - start, 3))
 
-    with open (f'{model}.onnx.json') as f: # TODO: wrap everything into a class, store the json permanentl
+    with open (f'{model}.onnx.json') as f: # TODO: wrap everything into a class, store the json permanently
         model_json = json.load(f)
 
     os.chdir(pwd)
-    return model_json
+    return model_json, audio
+
+
+# TODO: framework should probably be a dataclass that holds all the relevant informations, also model
+# TODO: make a common repo that hold the configuration dataclasses ?
+def synthesize(text, target_sample_rate, framework, model="en_US-lessac-medium", return_lc3=True):
+
+    if framework == 'piper':
+        model_json, audio_raw = synth_piper(text, model)
+        tts_sample_rate = model_json['audio']['sample_rate']
+        audio_np = np.frombuffer(audio_raw, dtype=np.dtype('<i2')).astype(np.float32) /(2**15-1)# convert to float fraction
+        audio = resample_array(audio_np, tts_sample_rate, target_sample_rate)
+
+    elif framework == 'koro':
+        pass
+    elif framework == 'xtts':
+        pass
+    elif framework == 'zonos':
+        pass
+    else: raise NotImplementedError('unknown framework')
+
+    if return_lc3:
+        audio_pcm = (audio_np * 2**15-1).astype(np.int16)
+        lc3 = encode_lc3.encode(audio_pcm, target_sample_rate, 40) # TODO: octetts per frame should be parameter
+
+        return lc3
+    else:
+        return audio
+
 
 if __name__ == '__main__':
     import logging
+    import soundfile as sf
 
     logging.basicConfig(
         level=logging.INFO,
         format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
     )
+    target_rate=16000
 
-    synthesize('Hello World')
+    audio = synthesize('Hello World', target_rate, 'piper', model= 'de_DE-kerstin-low', encode_lc3=False)
+
+    sf.write('hello.wav', audio, target_rate)
+
+
+    print('Done.')
diff --git a/multilang_translator/utils/resample.py b/multilang_translator/utils/resample.py
index 2d06f6b..c0b5cce 100644
--- a/multilang_translator/utils/resample.py
+++ b/multilang_translator/utils/resample.py
@@ -23,5 +23,26 @@ def resample_file(filename, out_filename, target_rate=int(24e3)):
     
     log.info("Resampling of %s took  %s s", os.path.basename(filename), round(time.time() - start, 3))
 
+
+def resample_array(audio, rate, target_rate=int(24e3)):
+    start=time.time()
+    # Load the original audio file
+
+    if rate == target_rate: # Nothing to do 
+        log.info('audio already at target rate, skipping resample')
+        return audio
+
+    # Convert the sample rate to target rate
+    resampled_audio = librosa.resample(audio, orig_sr=rate, target_sr=target_rate)
+
+    # Save the resampled audio as a new .wav file
+    
+    log.info("Resampling took  %s s", round(time.time() - start, 3))
+    return resampled_audio
+
+
 if __name__ == "__main__":
-    resample_file('text_to_speech/welcome.wav', 'text_to_speech/welcome_resampled.wav')
+    import os
+    os.chdir(os.path.dirname(__file__))
+    file_dir = '../text_to_speech/'
+    resample_file(f'{file_dir}/welcome.wav', f'{file_dir}/welcome_resampled.wav')