Fix audio type conversion (#259)

* Fix conversion between audio dtypes

* Run Pytest in CI

* Add pytest tests path in pyproject.toml

* Fix usages

* Use other PR's test format (more or less)

* Support legacy arguments

* Fix pyproject.toml and test location

* Omit `test` arg in CI, given by pyproject.toml

---------

Co-authored-by: Freddy Boulton <alfonsoboulton@gmail.com>

.github/workflows/tests.yml

@@ -13,7 +13,7 @@ jobs:
       - name: Run linters
         run: |
           pip install ruff pyright
-          pip install -e .
+          pip install -e .[dev]
           ruff check .
           ruff format --check --diff .
           pyright
@@ -35,5 +35,5 @@ jobs:
         run: |
           python -m pip install -U pip
           pip install .[dev]
-          python -m pytest -s test
+          python -m pytest --capture=no
         shell: bash

backend/fastrtc/pause_detection/silero.py

@@ -8,7 +8,7 @@ import numpy as np
 from huggingface_hub import hf_hub_download
 from numpy.typing import NDArray
 
-from ..utils import AudioChunk
+from ..utils import AudioChunk, audio_to_float32
 from .protocol import PauseDetectionModel
 
 logger = logging.getLogger(__name__)
@@ -274,8 +274,7 @@ class SileroVADModel:
         sampling_rate, audio_ = audio
         logger.debug("VAD audio shape input: %s", audio_.shape)
         try:
-            if audio_.dtype != np.float32:
-                audio_ = audio_.astype(np.float32) / 32768.0
+            audio_ = audio_to_float32(audio_)
             sr = 16000
             if sr != sampling_rate:
                 try:

backend/fastrtc/reply_on_stopwords.py

@@ -161,7 +161,7 @@ class ReplyOnStopWords(ReplyOnPause):
 
         if duration >= self.algo_options.audio_chunk_duration:
             if not state.stop_word_detected:
-                audio_f32 = audio_to_float32((sampling_rate, audio))
+                audio_f32 = audio_to_float32(audio)
                 audio_rs = librosa.resample(
                     audio_f32, orig_sr=sampling_rate, target_sr=16000
                 )

backend/fastrtc/speech_to_text/stt_.py

@@ -32,8 +32,7 @@ class MoonshineSTT(STTModel):
 
     def stt(self, audio: tuple[int, NDArray[np.int16 | np.float32]]) -> str:
         sr, audio_np = audio  # type: ignore
-        if audio_np.dtype == np.int16:
-            audio_np = audio_to_float32(audio)
+        audio_np = audio_to_float32(audio_np)
         if sr != 16000:
             audio_np: NDArray[np.float32] = librosa.resample(
                 audio_np, orig_sr=sr, target_sr=16000

backend/fastrtc/utils.py

@@ -7,6 +7,7 @@ import json
 import logging
 import tempfile
 import traceback
+import warnings
 from collections.abc import Callable, Coroutine
 from contextvars import ContextVar
 from dataclasses import dataclass
@@ -211,7 +212,7 @@ async def player_worker_decode(
                 first_sample_rate = sample_rate
 
             if format == "s16":
-                audio_array = audio_to_float32((sample_rate, audio_array))
+                audio_array = audio_to_float32(audio_array)
 
             if first_sample_rate != sample_rate:
                 audio_array = librosa.resample(
@@ -319,17 +320,15 @@ def audio_to_file(audio: tuple[int, NDArray[np.int16 | np.float32]]) -> str:
 
 
 def audio_to_float32(
-    audio: tuple[int, NDArray[np.int16 | np.float32]],
+    audio: NDArray[np.int16 | np.float32] | tuple[int, NDArray[np.int16 | np.float32]],
 ) -> NDArray[np.float32]:
     """
     Convert an audio tuple containing sample rate (int16) and numpy array data to float32.
 
     Parameters
     ----------
-    audio : tuple[int, np.ndarray]
-        A tuple containing:
-            - sample_rate (int): The audio sample rate in Hz
-            - data (np.ndarray): The audio data as a numpy array
+    audio : np.ndarray
+        The audio data as a numpy array
 
     Returns
     -------
@@ -338,26 +337,39 @@ def audio_to_float32(
 
     Example
     -------
-    >>> sample_rate = 44100
     >>> audio_data = np.array([0.1, -0.2, 0.3])  # Example audio samples
-    >>> audio_tuple = (sample_rate, audio_data)
-    >>> audio_float32 = audio_to_float32(audio_tuple)
+    >>> audio_float32 = audio_to_float32(audio_data)
     """
-    return audio[1].astype(np.float32) / 32768.0
+    if isinstance(audio, tuple):
+        warnings.warn(
+            UserWarning(
+                "Passing a (sr, audio) tuple to audio_to_float32() is deprecated "
+                "and will be removed in a future release. Pass only the audio array."
+            ),
+            stacklevel=2,  # So that the warning points to the user's code
+        )
+        _sr, audio = audio
+
+    if audio.dtype == np.int16:
+        # Divide by 32768.0 so that the values are in the range [-1.0, 1.0).
+        # 1.0 can actually never be reached because the int16 range is [-32768, 32767].
+        return audio.astype(np.float32) / 32768.0
+    elif audio.dtype == np.float32:
+        return audio  # type: ignore
+    else:
+        raise TypeError(f"Unsupported audio data type: {audio.dtype}")
 
 
 def audio_to_int16(
-    audio: tuple[int, NDArray[np.int16 | np.float32]],
+    audio: NDArray[np.int16 | np.float32] | tuple[int, NDArray[np.int16 | np.float32]],
 ) -> NDArray[np.int16]:
     """
     Convert an audio tuple containing sample rate and numpy array data to int16.
 
     Parameters
     ----------
-    audio : tuple[int, np.ndarray]
-        A tuple containing:
-            - sample_rate (int): The audio sample rate in Hz
-            - data (np.ndarray): The audio data as a numpy array
+    audio : np.ndarray
+        The audio data as a numpy array
 
     Returns
     -------
@@ -366,18 +378,27 @@ def audio_to_int16(
 
     Example
     -------
-    >>> sample_rate = 44100
     >>> audio_data = np.array([0.1, -0.2, 0.3], dtype=np.float32)  # Example audio samples
-    >>> audio_tuple = (sample_rate, audio_data)
-    >>> audio_int16 = audio_to_int16(audio_tuple)
+    >>> audio_int16 = audio_to_int16(audio_data)
     """
-    if audio[1].dtype == np.int16:
-        return audio[1]  # type: ignore
-    elif audio[1].dtype == np.float32:
-        # Convert float32 to int16 by scaling to the int16 range
-        return (audio[1] * 32767.0).astype(np.int16)
+    if isinstance(audio, tuple):
+        warnings.warn(
+            UserWarning(
+                "Passing a (sr, audio) tuple to audio_to_float32() is deprecated "
+                "and will be removed in a future release. Pass only the audio array."
+            ),
+            stacklevel=2,  # So that the warning points to the user's code
+        )
+        _sr, audio = audio
+
+    if audio.dtype == np.int16:
+        return audio  # type: ignore
+    elif audio.dtype == np.float32:
+        # Convert float32 to int16 by scaling to the int16 range.
+        # Multiply by 32767 and not 32768 so that int16 doesn't overflow.
+        return (audio * 32767.0).astype(np.int16)
     else:
-        raise TypeError(f"Unsupported audio data type: {audio[1].dtype}")
+        raise TypeError(f"Unsupported audio data type: {audio.dtype}")
 
 
 def aggregate_bytes_to_16bit(chunks_iterator):

backend/fastrtc/websocket.py

@@ -12,7 +12,14 @@ from fastapi import WebSocket
 from fastapi.websockets import WebSocketDisconnect, WebSocketState
 
 from .tracks import AsyncStreamHandler, StreamHandlerImpl
-from .utils import AdditionalOutputs, CloseStream, DataChannel, split_output
+from .utils import (
+    AdditionalOutputs,
+    CloseStream,
+    DataChannel,
+    audio_to_float32,
+    audio_to_int16,
+    split_output,
+)
 
 
 class WebSocketDataChannel(DataChannel):
@@ -31,14 +38,12 @@ def convert_to_mulaw(
     audio_data: np.ndarray, original_rate: int, target_rate: int
 ) -> bytes:
     """Convert audio data to 8kHz mu-law format"""
-
-    if audio_data.dtype != np.float32:
-        audio_data = audio_data.astype(np.float32) / 32768.0
+    audio_data = audio_to_float32(audio_data)
 
     if original_rate != target_rate:
         audio_data = librosa.resample(audio_data, orig_sr=original_rate, target_sr=8000)
 
-    audio_data = (audio_data * 32768).astype(np.int16)
+    audio_data = audio_to_int16(audio_data)
 
     return audioop.lin2ulaw(audio_data, 2)  # type: ignore
 
@@ -122,14 +127,13 @@ class WebSocketHandler:
                     )
 
                     if self.stream_handler.input_sample_rate != 8000:
-                        audio_array = audio_array.astype(np.float32) / 32768.0
+                        audio_array = audio_to_float32(audio_array)
                         audio_array = librosa.resample(
                             audio_array,
                             orig_sr=8000,
                             target_sr=self.stream_handler.input_sample_rate,
                         )
-                        audio_array = (audio_array * 32768).astype(np.int16)
-
+                        audio_array = audio_to_int16(audio_array)
                     try:
                         if isinstance(self.stream_handler, AsyncStreamHandler):
                             await self.stream_handler.receive(

demo/nextjs_voice_chat/backend/server.py

@@ -1,6 +1,6 @@
 import fastapi
 from fastrtc import ReplyOnPause, Stream, AlgoOptions, SileroVadOptions
-from fastrtc.utils import audio_to_bytes
+from fastrtc.utils import audio_to_bytes, audio_to_float32
 from openai import OpenAI
 import logging
 import time
@@ -78,8 +78,8 @@ def echo(audio):
     )
 
     for audio_chunk in audio_stream:
-        audio_array = (
-            np.frombuffer(audio_chunk, dtype=np.int16).astype(np.float32) / 32768.0
+        audio_array = audio_to_float32(
+            np.frombuffer(audio_chunk, dtype=np.int16)
         )
         yield (24000, audio_array)
 

pyproject.toml

@@ -83,6 +83,7 @@ artifacts = ["/backend/fastrtc/templates", "*.pyi"]
 packages = ["/backend/fastrtc"]
 
 [tool.pytest.ini_options]
+testpaths = ["test/"]
 asyncio_mode = "auto"
 asyncio_default_fixture_loop_scope = "function"
 

test/test_utils.py

@@ -0,0 +1,61 @@
+import numpy as np
+import pytest
+from fastrtc.utils import audio_to_float32, audio_to_int16
+
+
+def test_audio_to_float32_valid_int16():
+    audio = np.array([-32768, 0, 32767], dtype=np.int16)
+    expected = np.array([-1.0, 0.0, 32767 / 32768.0], dtype=np.float32)
+    result = audio_to_float32(audio)
+    np.testing.assert_array_almost_equal(result, expected)
+
+
+def test_audio_to_float32_valid_float32():
+    audio = np.array([-1.0, 0.0, 1.0], dtype=np.float32)
+    result = audio_to_float32(audio)
+    np.testing.assert_array_equal(result, audio)
+
+
+def test_audio_to_float32_empty_array():
+    audio = np.array([], dtype=np.int16)
+    result = audio_to_float32(audio)
+    np.testing.assert_array_equal(result, np.array([], dtype=np.float32))
+
+
+def test_audio_to_float32_invalid_dtype():
+    audio = np.array([1, 2, 3], dtype=np.int32)
+    with pytest.raises(TypeError, match="Unsupported audio data type"):
+        audio_to_float32(audio)  # type: ignore
+
+
+def test_audio_to_int16_valid_float32():
+    audio = np.array([-1.0, 0.0, 1.0], dtype=np.float32)
+    expected = np.array([-32767, 0, 32767], dtype=np.int16)
+    result = audio_to_int16(audio)
+    np.testing.assert_array_equal(result, expected)
+
+
+def test_audio_to_int16_valid_int16():
+    audio = np.array([-32768, 0, 32767], dtype=np.int16)
+    result = audio_to_int16(audio)
+    np.testing.assert_array_equal(result, audio)
+
+
+def test_audio_to_int16_empty_array():
+    audio = np.array([], dtype=np.float32)
+    result = audio_to_int16(audio)
+    np.testing.assert_array_equal(result, np.array([], dtype=np.int16))
+
+
+def test_audio_to_int16_invalid_dtype():
+    audio = np.array([1, 2, 3], dtype=np.int32)
+    with pytest.raises(TypeError, match="Unsupported audio data type"):
+        audio_to_int16(audio)  # type: ignore
+
+
+def test_legacy_arguments():
+    result = audio_to_float32((16000, np.zeros(10, dtype=np.int16)))
+    np.testing.assert_array_equal(result, np.zeros(10, dtype=np.float32))
+
+    result = audio_to_int16((16000, np.zeros(10, dtype=np.float32)))
+    np.testing.assert_array_equal(result, np.zeros(10, dtype=np.int16))