Refactor Docker configurations and update test mocks for development routers

Author: remsky

.gitignore

@@ -6,7 +6,6 @@ ui/data/*
 *.db
 *.pyc
 *.pth
-*.pt
 
 Kokoro-82M/*
 __pycache__/

api/src/routers/openai_compatible.py

@@ -93,6 +93,7 @@ async def create_speech(
                     "Content-Disposition": f"attachment; filename=speech.{request.response_format}",
                     "X-Accel-Buffering": "no",  # Disable proxy buffering
                     "Cache-Control": "no-cache",  # Prevent caching
+                    "Transfer-Encoding": "chunked",  # Enable chunked transfer encoding
                 },
             )
         else:

api/src/services/audio.py

@@ -104,7 +104,7 @@ def convert_audio(
                 # Raw 16-bit PCM samples, no header
                 buffer.write(normalized_audio.tobytes())
             elif output_format == "wav":
-                # Always use soundfile for WAV to ensure proper headers and normalization
+                # WAV format with headers
                 sf.write(
                     buffer,
                     normalized_audio,
@@ -113,14 +113,14 @@ def convert_audio(
                     subtype="PCM_16",
                 )
             elif output_format == "mp3":
-                # Use format settings or defaults
+                # MP3 format with proper framing
                 settings = format_settings.get("mp3", {}) if format_settings else {}
                 settings = {**AudioService.DEFAULT_SETTINGS["mp3"], **settings}
                 sf.write(
                     buffer, normalized_audio, sample_rate, format="MP3", **settings
                 )
-
             elif output_format == "opus":
+                # Opus format in OGG container
                 settings = format_settings.get("opus", {}) if format_settings else {}
                 settings = {**AudioService.DEFAULT_SETTINGS["opus"], **settings}
                 sf.write(
@@ -131,8 +131,8 @@ def convert_audio(
                     subtype="OPUS",
                     **settings,
                 )
-
             elif output_format == "flac":
+                # FLAC format with proper framing
                 if is_first_chunk:
                     logger.info("Starting FLAC stream...")
                 settings = format_settings.get("flac", {}) if format_settings else {}
@@ -145,15 +145,14 @@ def convert_audio(
                     subtype="PCM_16",
                     **settings,
                 )
+            elif output_format == "aac":
+                raise ValueError(
+                    "Format aac not currently supported. Supported formats are: wav, mp3, opus, flac, pcm."
+                )
             else:
-                if output_format == "aac":
-                    raise ValueError(
-                        "Format aac not supported. Supported formats are: wav, mp3, opus, flac, pcm."
-                    )
-                else:
-                    raise ValueError(
-                        f"Format {output_format} not supported. Supported formats are: wav, mp3, opus, flac, pcm."
-                    )
+                raise ValueError(
+                    f"Format {output_format} not supported. Supported formats are: wav, mp3, opus, flac, pcm, aac."
+                )
 
             buffer.seek(0)
             return buffer.getvalue()

api/src/services/tts_service.py

@@ -177,14 +177,15 @@ async def generate_audio_stream(
                     )
 
                     if chunk_audio is not None:
-                        # Convert chunk with proper header handling
+                        # Convert chunk with proper streaming header handling
                         chunk_bytes = AudioService.convert_audio(
                             chunk_audio,
                             24000,
                             output_format,
                             is_first_chunk=is_first,
                             normalizer=stream_normalizer,
                             is_last_chunk=(next_chunk is None),  # Last if no next chunk
+                            stream=True  # Ensure proper streaming format handling
                         )
 
                         yield chunk_bytes

api/tests/conftest.py

@@ -181,7 +181,7 @@ def mock_tts_service(monkeypatch):
     # Mock TTSModel.generate_from_tokens since we call it directly
     mock_generate = Mock(return_value=np.zeros(48000))
     monkeypatch.setattr(
-        "api.src.routers.text_processing.TTSModel.generate_from_tokens", mock_generate
+        "api.src.routers.development.TTSModel.generate_from_tokens", mock_generate
     )
 
     return mock_service
@@ -192,5 +192,5 @@ def mock_audio_service(monkeypatch):
     """Mock AudioService"""
     mock_service = Mock()
     mock_service.convert_audio.return_value = b"mock audio data"
-    monkeypatch.setattr("api.src.routers.text_processing.AudioService", mock_service)
+    monkeypatch.setattr("api.src.routers.development.AudioService", mock_service)
     return mock_service

api/tests/test_audio_service.py

@@ -63,7 +63,7 @@ def test_convert_to_aac_raises_error(sample_audio):
     audio_data, sample_rate = sample_audio
     with pytest.raises(
         ValueError,
-        match="Format aac not supported. Supported formats are: wav, mp3, opus, flac, pcm.",
+        match="Failed to convert audio to aac: Format aac not currently supported. Supported formats are: wav, mp3, opus, flac, pcm.",
     ):
         AudioService.convert_audio(audio_data, sample_rate, "aac")
 

api/tests/test_text_processing.py

@@ -20,8 +20,8 @@ async def async_client():
 @pytest.mark.asyncio
 async def test_phonemize_endpoint(async_client):
     """Test phoneme generation endpoint"""
-    with patch("api.src.routers.text_processing.phonemize") as mock_phonemize, patch(
-        "api.src.routers.text_processing.tokenize"
+    with patch("api.src.routers.development.phonemize") as mock_phonemize, patch(
+        "api.src.routers.development.tokenize"
     ) as mock_tokenize:
         # Setup mocks
         mock_phonemize.return_value = "həlˈoʊ"
@@ -56,7 +56,7 @@ async def test_generate_from_phonemes(
 ):
     """Test audio generation from phonemes"""
     with patch(
-        "api.src.routers.text_processing.TTSService", return_value=mock_tts_service
+        "api.src.routers.development.TTSService", return_value=mock_tts_service
     ):
         response = await async_client.post(
             "/text/generate_from_phonemes",
@@ -76,7 +76,7 @@ async def test_generate_from_phonemes_invalid_voice(async_client, mock_tts_servi
     """Test audio generation with invalid voice"""
     mock_tts_service._get_voice_path.return_value = None
     with patch(
-        "api.src.routers.text_processing.TTSService", return_value=mock_tts_service
+        "api.src.routers.development.TTSService", return_value=mock_tts_service
     ):
         response = await async_client.post(
             "/text/generate_from_phonemes",
@@ -111,7 +111,7 @@ async def test_generate_from_phonemes_invalid_speed(async_client, monkeypatch):
 async def test_generate_from_phonemes_empty_phonemes(async_client, mock_tts_service):
     """Test audio generation with empty phonemes"""
     with patch(
-        "api.src.routers.text_processing.TTSService", return_value=mock_tts_service
+        "api.src.routers.development.TTSService", return_value=mock_tts_service
     ):
         response = await async_client.post(
             "/text/generate_from_phonemes",

docker-compose.cpu.yml

@@ -26,11 +26,11 @@ services:
       start_period: 1s
 
   kokoro-tts:
-    # image: ghcr.io/remsky/kokoro-fastapi:latest-cpu
-    # Uncomment below to build from source instead of using the released image
-    build:
-      context: .
-      dockerfile: Dockerfile.cpu
+    image: ghcr.io/remsky/kokoro-fastapi:latest-cpu
+    # Uncomment below (and comment out above) to build from source instead of using the released image
+    # build:
+      # context: .
+      # dockerfile: Dockerfile.cpu
     volumes:
       - ./api/src:/app/api/src
       - ./Kokoro-82M:/app/Kokoro-82M
@@ -52,8 +52,8 @@ services:
 
   # Gradio UI service [Comment out everything below if you don't need it]
   gradio-ui:
-    # image: ghcr.io/remsky/kokoro-fastapi:latest-ui
-    # Uncomment below to build from source instead of using the released image
+    image: ghcr.io/remsky/kokoro-fastapi:latest-ui
+    # Uncomment below (and comment out above) to build from source instead of using the released image
     build:
       context: ./ui
     ports:
@@ -63,3 +63,4 @@ services:
       - ./ui/app.py:/app/app.py  # Mount app.py for hot reload
     environment:
       - GRADIO_WATCH=True  # Enable hot reloading
+      - PYTHONUNBUFFERED=1  # Ensure Python output is not buffered

docker-compose.yml

@@ -32,10 +32,10 @@ services:
       start_period: 1s
 
   kokoro-tts:
-    # image: ghcr.io/remsky/kokoro-fastapi:latest
-    # Uncomment below to build from source instead of using the released image
-    build:
-      context: .
+    image: ghcr.io/remsky/kokoro-fastapi:latest
+    # Uncomment below (and comment out above) to build from source instead of using the released image
+    # build:
+      # context: .
     volumes:
       - ./api/src:/app/api/src
       - ./Kokoro-82M:/app/Kokoro-82M
@@ -50,20 +50,30 @@ services:
             - driver: nvidia
               count: 1
               capabilities: [gpu]
+    healthcheck:
+      test: ["CMD", "curl", "-f", "http://localhost:8880/v1/audio/voices"]
+      interval: 10s
+      timeout: 5s
+      retries: 30
+      start_period: 30s
     depends_on:
       model-fetcher:
         condition: service_healthy
 
   # Gradio UI service [Comment out everything below if you don't need it]
   gradio-ui:
-    # image: ghcr.io/remsky/kokoro-fastapi:latest-ui
-    # Uncomment below to build from source instead of using the released image
-    build:
-      context: ./ui
+    image: ghcr.io/remsky/kokoro-fastapi:latest-ui
+    # Uncomment below (and comment out above) to build from source instead of using the released image
+    # build:
+      # context: ./ui
     ports:
       - "7860:7860"
     volumes:
       - ./ui/data:/app/ui/data
       - ./ui/app.py:/app/app.py  # Mount app.py for hot reload
     environment:
       - GRADIO_WATCH=True  # Enable hot reloading
+      - PYTHONUNBUFFERED=1  # Ensure Python output is not buffered
+    depends_on:
+      kokoro-tts:
+        condition: service_healthy

examples/openai_streaming_audio.py

@@ -1,23 +1,25 @@
 #!/usr/bin/env rye run python
-# %%
 import time
 from pathlib import Path
 
 from openai import OpenAI
 
 # gets OPENAI_API_KEY from your environment variables
-openai = OpenAI(base_url="http://localhost:8000/v1", api_key="not-needed-for-local")
+openai = OpenAI(base_url="http://localhost:8880/v1", api_key="not-needed-for-local")
 
 speech_file_path = Path(__file__).parent / "speech.mp3"
 
 
-
-
-
-
 def main() -> None:
     stream_to_speakers()
 
+    # Create text-to-speech audio file
+    with openai.audio.speech.with_streaming_response.create(
+        model="kokoro",
+        voice="af_bella",
+        input="the quick brown fox jumped over the lazy dogs",
+    ) as response:
+        response.stream_to_file(speech_file_path)
 
 
 def stream_to_speakers() -> None:
@@ -31,9 +33,12 @@ def stream_to_speakers() -> None:
 
     with openai.audio.speech.with_streaming_response.create(
         model="kokoro",
-        voice=VOICE,
-        response_format="mp3",  # similar to WAV, but without a header chunk at the start.
-        input="""My dear sir, that is just where you are wrong. That is just where the whole world has gone wrong. We are always getting away from the present moment. Our mental existences, which are immaterial and have no dimensions, are passing along the Time-Dimension""",
+        voice="af_bella",
+        response_format="pcm",  # similar to WAV, but without a header chunk at the start.
+        input="""I see skies of blue and clouds of white
+                The bright blessed days, the dark sacred nights
+                And I think to myself
+                What a wonderful world""",
     ) as response:
         print(f"Time to first byte: {int((time.time() - start_time) * 1000)}ms")
         for chunk in response.iter_bytes(chunk_size=1024):
@@ -44,5 +49,3 @@ def stream_to_speakers() -> None:
 
 if __name__ == "__main__":
     main()
-
-# %%