Fish Audio API — 完全チュートリアル
Fish Audioは、音声生成、音声クローン、オーディオストーリーテリングのための最先端AIプラットフォームです。超低遅延ストリーミングと感情コントロールで30以上の言語をサポートします。
学習内容
- セットアップ&認証
- テキスト音声変換(TTS) — 基本から応用まで
- 感情と表現のコントロール
- 音声クローン — カスタム音声モデルの作成
- ストリーミングオーディオ
- バッチ処理&エラーハンドリング
前提条件: pip install fish-audio-sdk python-dotenv requests ormsgpack
APIキーの取得: fish.audio → Settings → API Keys
1. セットアップ&環境
# Install dependencies (run once)
# !pip install fish-audio-sdk python-dotenv requests ormsgpackimport os
from dotenv import load_dotenv
# Load API keys from .env file
# Create a .env file with: FISH_AUDIO_API_KEY=your_key_here
load_dotenv()
FISH_API_KEY = os.getenv("FISH_AUDIO_API_KEY")
if not FISH_API_KEY:
raise ValueError("FISH_AUDIO_API_KEY not found in .env file. Please add it.")
print(f"✓ API key loaded: {FISH_API_KEY[:8]}...")from fishaudio import FishAudio
from fishaudio.utils import save
import os
# Initialize the Fish Audio client
client = FishAudio(api_key=FISH_API_KEY)
# Create output directory
os.makedirs("audio_output", exist_ok=True)
print("✓ Fish Audio client initialized")2. 基本的なテキスト音声変換
Fish Audio Discoveryページの事前作成済み音声を使って音声を生成します。そこから音声モデルIDをコピーしてください。
利用可能なモデル
| モデル | 品質 | 速度 | ユースケース |
|---|---|---|---|
s1 |
優秀 | 高速 | 最新機能、最高の自然さ |
speech-1.6 |
非常に良好 | 高速 | 安定したプロダクション用 |
speech-1.5 |
良好 | 最速 | レガシーサポート |
# A popular public voice model from Fish Audio Discovery
# Replace with any model ID from https://fish.audio/discovery
VOICE_MODEL_ID = "7f92f8afb8ec43bf81429cc1c9199cb1"
# Basic TTS — simplest usage
audio = client.tts.convert(
text="Hello! Welcome to the Fish Audio tutorial. This is your first AI-generated voice.",
reference_id=VOICE_MODEL_ID
)
save(audio, "audio_output/01_basic_tts.mp3")
print("✓ Saved: audio_output/01_basic_tts.mp3")# TTS with advanced options
audio = client.tts.convert(
text="This audio uses advanced options: MP3 format at 192 kbps, balanced latency mode.",
reference_id=VOICE_MODEL_ID,
format="mp3", # Options: "mp3", "wav", "pcm", "opus"
mp3_bitrate=192, # 64, 128, or 192 kbps
chunk_length=200, # 100–300 characters per processing chunk
latency="balanced" # "normal" (quality) or "balanced" (~300ms latency)
)
save(audio, "audio_output/02_advanced_tts.mp3")
print("✓ Saved: audio_output/02_advanced_tts.mp3")3. 感情と表現のコントロール
音声の聞こえ方をコントロールするために、文の先頭にかっこで感情タグを追加します。
重要ルール: 感情タグは文の先頭に置いてください。途中には置かないでください。
使用可能な感情
| カテゴリ | タグ |
|---|---|
| 基本 | (happy) (sad) (angry) (excited) (calm) |
| トーン | (whispering) (shouting) (soft tone) |
| エフェクト | (laughing) (sighing) (crying) (panting) |
# Emotion control — tags at START of sentences
emotional_text = """(excited) I just got accepted into my dream school!
(happy) This is the best day of my life.
(sad) But I will miss all my friends here.
(calm) Everything will work out in the end."""
audio = client.tts.convert(
text=emotional_text,
reference_id=VOICE_MODEL_ID
)
save(audio, "audio_output/03_emotions.mp3")
print("✓ Saved: audio_output/03_emotions.mp3")# Storytelling with multiple emotion transitions
story_text = """(narrator) Chapter One: The Discovery.
(mysterious)(whispering) Something was hidden beneath the old floorboards.
(excited) She pulled up the wooden plank and found a glowing box!
(scared)(shouting) Suddenly, a loud bang echoed through the house.
(relieved)(sighing) It was just the cat knocking over a lamp.
(laughing) Ha ha ha, what a scare that was!"""
audio = client.tts.convert(
text=story_text,
reference_id=VOICE_MODEL_ID
)
save(audio, "audio_output/04_storytelling.mp3")
print("✓ Saved: audio_output/04_storytelling.mp3")4. 参照音声の使用(リアルタイム音声クローン)
保存済みモデルの代わりに、音声ファイルを直接音声参照として提供できます。永続的なモデルを作成せずに素早く音声マッチングするのに便利です。
from fishaudio.types import ReferenceAudio
# Use reference audio on-the-fly
# Replace "voice_sample.wav" with your own audio file path
REFERENCE_AUDIO_PATH = "voice_sample.wav" # your 10+ second audio sample
REFERENCE_AUDIO_TEXT = "Hello, this is a sample of my voice speaking naturally."
if os.path.exists(REFERENCE_AUDIO_PATH):
with open(REFERENCE_AUDIO_PATH, "rb") as f:
audio = client.tts.convert(
text="This voice was cloned from a reference audio file provided on the fly.",
references=[
ReferenceAudio(
audio=f.read(),
text=REFERENCE_AUDIO_TEXT # transcript of the reference audio
)
]
)
save(audio, "audio_output/05_reference_audio.mp3")
print("✓ Saved: audio_output/05_reference_audio.mp3")
else:
print(f"⚠ Reference file not found: {REFERENCE_AUDIO_PATH}")
print(" Provide a .wav/.mp3 file to use this feature.")5. 音声モデルの作成(音声クローンAPI)
音声サンプルから永続的なカスタム音声モデルを作成します。作成したら、今後のすべてのTTSコールにそのモデルIDを使用します。
最良の結果のための録音ヒント
- 最小: サンプルごとに10秒(15〜20秒推奨)
- 環境: 静かな部屋 — 寝室、駐車した車、またはオフィス
- 話者: 一人の話者、一定の音量
- 含める: 文の間の自然なポーズ
- 避ける: バックグラウンドミュージック、複数の話者、突然の音量変化
import requests
def create_voice_model(
api_key: str,
title: str,
audio_files: list,
transcripts: list = None,
description: str = "",
visibility: str = "private",
enhance_audio: bool = True
) -> dict:
"""
Create a voice model from audio samples.
Args:
api_key: Fish Audio API key
title: Name for your voice model
audio_files: List of file paths (.mp3, .wav, .m4a, .opus)
transcripts: Optional list of transcripts (must match number of audio files)
description: Optional description
visibility: 'private', 'public', or 'unlist'
enhance_audio: Remove background noise
Returns:
dict with model info including 'id'
"""
files = [("voices", open(f, "rb")) for f in audio_files]
data = [
("title", title),
("description", description),
("visibility", visibility),
("type", "tts"),
("train_mode", "fast"),
("enhance_audio_quality", str(enhance_audio).lower())
]
# Transcripts must match count of audio files exactly
if transcripts:
assert len(transcripts) == len(audio_files), \
f"Transcripts count ({len(transcripts)}) must match audio files ({len(audio_files)})"
for t in transcripts:
data.append(("texts", t))
response = requests.post(
"https://api.fish.audio/model",
files=files,
data=data,
headers={"Authorization": f"Bearer {api_key}"}
)
response.raise_for_status()
# Close file handles
for _, f in files:
f.close()
return response.json()
print("✓ create_voice_model() function defined")# Example: Create a voice model (requires actual audio files)
SAMPLE_FILES = ["voice_sample.wav"] # Replace with your audio file paths
SAMPLE_TRANSCRIPTS = [
"Hello, my name is Alex, and I enjoy reading books about technology and science."
]
if all(os.path.exists(f) for f in SAMPLE_FILES):
result = create_voice_model(
api_key=FISH_API_KEY,
title="My Tutorial Voice",
audio_files=SAMPLE_FILES,
transcripts=SAMPLE_TRANSCRIPTS,
description="Voice model created in Fish Audio tutorial",
visibility="private",
enhance_audio=True
)
MY_MODEL_ID = result.get("_id") or result.get("id")
print(f"✓ Voice model created!")
print(f" Model ID: {MY_MODEL_ID}")
print(f" State: {result.get('state')}")
else:
print("⚠ Audio sample files not found.")
print(" Provide .wav/.mp3 files to create a voice model.")
MY_MODEL_ID = VOICE_MODEL_ID # Fall back to public model6. ストリーミングオーディオ
リアルタイムで音声チャンクをストリーミング — ライブアプリケーション、チャットボット、インタラクティブシステムに最適です。バランスモードでは最初の音声チャンクが約300ms以内に届きます。
import time
def stream_tts(client, text: str, model_id: str, output_path: str) -> dict:
"""
Stream TTS audio and save to file, reporting timing stats.
Returns:
dict with 'chunks', 'total_bytes', 'duration_sec'
"""
start = time.time()
first_chunk_time = None
chunk_count = 0
total_bytes = 0
audio_stream = client.tts.stream(
text=text,
reference_id=model_id,
latency="balanced" # lower latency for streaming
)
with open(output_path, "wb") as f:
for chunk in audio_stream:
if first_chunk_time is None:
first_chunk_time = time.time() - start
f.write(chunk)
chunk_count += 1
total_bytes += len(chunk)
duration = time.time() - start
stats = {
"chunks": chunk_count,
"total_bytes": total_bytes,
"first_chunk_sec": round(first_chunk_time, 3),
"total_sec": round(duration, 3)
}
return stats
stats = stream_tts(
client=client,
text="Streaming audio is perfect for real-time applications like chatbots and voice assistants.",
model_id=VOICE_MODEL_ID,
output_path="audio_output/06_streaming.mp3"
)
print("✓ Streaming complete!")
print(f" First chunk: {stats['first_chunk_sec']}s")
print(f" Total time: {stats['total_sec']}s")
print(f" Chunks: {stats['chunks']}")
print(f" Total size: {stats['total_bytes']:,} bytes")7. ormsgpackを使った直接API
最大限のコントロールのために、MessagePackシリアライゼーションを使ってREST APIを直接呼び出します。細かいコントロールが必要な場合や、Python以外の統合を構築する場合に便利です。
import httpx
import ormsgpack
def tts_direct_api(
api_key: str,
text: str,
reference_id: str,
output_path: str,
format: str = "mp3",
model: str = "s1"
) -> int:
"""
Call Fish Audio TTS API directly with MessagePack.
Returns size of saved audio in bytes.
"""
request_data = {
"text": text,
"reference_id": reference_id,
"format": format,
"chunk_length": 200,
"latency": "normal"
}
with httpx.Client(timeout=30) as http_client:
response = http_client.post(
"https://api.fish.audio/v1/tts",
content=ormsgpack.packb(request_data),
headers={
"authorization": f"Bearer {api_key}",
"content-type": "application/msgpack",
"model": model
}
)
response.raise_for_status()
with open(output_path, "wb") as f:
f.write(response.content)
return len(response.content)
size = tts_direct_api(
api_key=FISH_API_KEY,
text="This was generated using the direct REST API with MessagePack serialization.",
reference_id=VOICE_MODEL_ID,
output_path="audio_output/07_direct_api.mp3"
)
print(f"✓ Direct API call successful: {size:,} bytes saved")8. バッチ処理
複数の音声ファイルを効率的に生成 — オーディオブック、アナウンス、マルチシーンスクリプトに便利です。
import time
from fishaudio.exceptions import FishAudioError
def generate_with_retry(client, text: str, model_id: str, max_retries: int = 3):
"""Generate TTS with exponential backoff retry."""
for attempt in range(max_retries):
try:
return client.tts.convert(text=text, reference_id=model_id)
except FishAudioError as e:
if attempt < max_retries - 1:
wait = 2 ** attempt
print(f" Retry {attempt + 1}/{max_retries - 1} in {wait}s: {e}")
time.sleep(wait)
else:
raise
def batch_tts(client, texts: list, model_id: str, output_dir: str, prefix: str = "batch") -> list:
"""
Generate TTS for a list of texts and save to individual files.
Returns:
List of output file paths
"""
os.makedirs(output_dir, exist_ok=True)
output_files = []
for i, text in enumerate(texts):
output_path = os.path.join(output_dir, f"{prefix}_{i+1:02d}.mp3")
print(f" [{i+1}/{len(texts)}] Generating: {text[:50]}...")
audio = generate_with_retry(client, text, model_id)
save(audio, output_path)
output_files.append(output_path)
return output_files
# Example: Generate announcements
announcements = [
"Welcome to the annual technology conference. Please find your seats.",
"The keynote presentation will begin in five minutes.",
"Refreshments are available in the lobby throughout the day.",
]
print("Generating batch audio...")
files = batch_tts(
client=client,
texts=announcements,
model_id=VOICE_MODEL_ID,
output_dir="audio_output/batch",
prefix="announcement"
)
print(f"\n✓ Batch complete: {len(files)} files generated")
for f in files:
print(f" {f}")9. 実践例 — マルチキャラクターダイアログ
感情コントロールとバッチ処理を組み合わせて、各キャラクターが異なる感情表現を使うマルチキャラクターストーリーを生成します。
# Multi-character dialogue script
# In a real project, each character would use a different reference_id (voice model)
dialogue = [
{"character": "narrator", "text": "(calm) It was a dark and stormy night at the old library."},
{"character": "alice", "text": "(excited) I found it! The ancient manuscript is right here!"},
{"character": "librarian", "text": "(angry)(shouting) Quiet! This is a library!"},
{"character": "alice", "text": "(scared)(whispering) Sorry... but look at what it says."},
{"character": "librarian", "text": "(curious) Let me see. (sighing) Oh my. This changes everything."},
{"character": "narrator", "text": "(mysterious) And so began the greatest adventure of their lives."},
]
print("Generating multi-character dialogue...")
os.makedirs("audio_output/dialogue", exist_ok=True)
for i, line in enumerate(dialogue):
output_path = f"audio_output/dialogue/line_{i+1:02d}_{line['character']}.mp3"
# In production: map character -> unique voice model ID
# Here we use the same model for demo purposes
audio = client.tts.convert(
text=line["text"],
reference_id=VOICE_MODEL_ID
)
save(audio, output_path)
print(f" [{i+1}] {line['character']}: {line['text'][:50]}")
print("\n✓ All dialogue lines generated!")10. 音声モデルの一覧
import requests
def list_my_models(api_key: str, page_size: int = 10) -> list:
"""Retrieve your voice models from Fish Audio."""
response = requests.get(
"https://api.fish.audio/model",
params={"page_size": page_size, "self": "true"},
headers={"Authorization": f"Bearer {api_key}"}
)
response.raise_for_status()
data = response.json()
return data.get("items", [])
models = list_my_models(FISH_API_KEY)
if models:
print(f"Your voice models ({len(models)} found):")
for m in models:
model_id = m.get('_id') or m.get('id')
print(f" • {m.get('title', 'Untitled')} — ID: {model_id} — Visibility: {m.get('visibility')}")
else:
print("No voice models found. Create one in Section 5!")11. まとめ&ベストプラクティス
生成されたファイル
すべての音声ファイルはaudio_output/に保存されます。
重要なポイント
TTS品質のヒント:
- 品質/速度のバランスには
chunk_length=200を使用 - リアルタイムアプリには
latency="balanced"、最高品質には"normal"を使用 - 自然なポーズのために適切な句読点を追加
感情コントロール:
- タグは文の先頭のみ:
(happy) Text here.✓ - 途中には置かない:
Text (happy) here.✗ - 文ごとに感情を1つ使用し、使いすぎを避ける
音声クローン:
- 音声サンプルごとに最低10秒
- 一人の話者がいる静かな環境で録音
- バックグラウンドノイズを除去するために
enhance_audio_quality=Trueを有効化 - 精度向上のために音声とともにトランスクリプトを提供
- 使用許可のある音声のみクローンする
プロダクションのヒント:
- モデルIDをキャッシュ — 毎回の呼び出しで調べない
- 指数バックオフのリトライを実装(セクション8参照)
- リアルタイム/低遅延アプリケーションにはストリーミングを使用
- APIキーは
.envに保存し、ハードコードしない
リソース
- 📖 完全APIドキュメント
- 🎙️ 音声ディスカバリー
- 💬 Discordコミュニティ
# Final summary of all generated files
import glob
all_files = sorted(glob.glob("audio_output/**/*.mp3", recursive=True))
total_size = sum(os.path.getsize(f) for f in all_files)
print(f"Tutorial complete! Generated {len(all_files)} audio files ({total_size/1024:.1f} KB total)\n")
for f in all_files:
size = os.path.getsize(f)
print(f" {f:55s} {size/1024:6.1f} KB")