├── .gitignore ├── .gitmodules ├── LICENSE ├── README.md ├── aivis_example.py ├── aivis_server.py ├── chatbot.py ├── chatbot_akari.py ├── chatgpt_example.py ├── config ├── en_to_jp_fix_dict.csv └── system_prompt.txt ├── gpt_publisher.py ├── jpg ├── akari_chatgpt_bot.jpg ├── faster_chatgpt_bot.jpg └── faster_chatgpt_bot_system.jpg ├── lib ├── aivis.py ├── chat.py ├── chat_akari.py ├── chat_akari_grpc.py ├── conf.py ├── en_to_jp.py ├── err_handler.py ├── google_speech.py ├── google_speech_grpc.py ├── google_speech_v2.py ├── google_speech_v2_grpc.py ├── grpc │ ├── gpt_server_pb2.py │ ├── gpt_server_pb2_grpc.py │ ├── motion_server_pb2.py │ ├── motion_server_pb2_grpc.py │ ├── speech_server_pb2.py │ ├── speech_server_pb2_grpc.py │ ├── voice_server_pb2.py │ └── voice_server_pb2_grpc.py ├── style_bert_vits.py ├── text_to_voice.py └── voicevox.py ├── manual_grpc_publisher_for_gpt.py ├── manual_grpc_publisher_for_voice.py ├── proto ├── codegen.py ├── gpt_server.proto ├── speech_server.proto └── voice_server.proto ├── pysen.toml ├── requirements.txt ├── script ├── faster_chatbot.sh ├── faster_chatbot_aivis.sh ├── faster_chatbot_aivis_auto.sh ├── faster_chatbot_auto.sh ├── faster_chatbot_bert_vits.sh └── faster_chatbot_bert_vits_auto.sh ├── speech_publisher.py ├── speech_to_text_example.py ├── style_bert_vits_example.py ├── style_bert_vits_server.py ├── talk_controller_client.py ├── text_to_kana_example.py ├── voicevox_example.py └── voicevox_server.py /.gitignore: -------------------------------------------------------------------------------- 1 | # Byte-compiled / optimized / DLL files 2 | __pycache__/ 3 | *.py[cod] 4 | *$py.class 5 | 6 | # C extensions 7 | *.so 8 | 9 | # Distribution / packaging 10 | .Python 11 | build/ 12 | develop-eggs/ 13 | dist/ 14 | downloads/ 15 | eggs/ 16 | .eggs/ 17 | lib64/ 18 | parts/ 19 | sdist/ 20 | var/ 21 | wheels/ 22 | share/python-wheels/ 23 | *.egg-info/ 24 | .installed.cfg 25 | *.egg 26 | MANIFEST 27 | 28 | # PyInstaller 29 | # Usually these files are written by a python script from a template 30 | # before PyInstaller builds the exe, so as to inject date/other infos into it. 31 | *.manifest 32 | *.spec 33 | 34 | # Installer logs 35 | pip-log.txt 36 | pip-delete-this-directory.txt 37 | 38 | # Unit test / coverage reports 39 | htmlcov/ 40 | .tox/ 41 | .nox/ 42 | .coverage 43 | .coverage.* 44 | .cache 45 | nosetests.xml 46 | coverage.xml 47 | *.cover 48 | *.py,cover 49 | .hypothesis/ 50 | .pytest_cache/ 51 | cover/ 52 | 53 | # Translations 54 | *.mo 55 | *.pot 56 | 57 | # Django stuff: 58 | *.log 59 | local_settings.py 60 | db.sqlite3 61 | db.sqlite3-journal 62 | 63 | # Flask stuff: 64 | instance/ 65 | .webassets-cache 66 | 67 | # Scrapy stuff: 68 | .scrapy 69 | 70 | # Sphinx documentation 71 | docs/_build/ 72 | 73 | # PyBuilder 74 | .pybuilder/ 75 | target/ 76 | 77 | # Jupyter Notebook 78 | .ipynb_checkpoints 79 | 80 | # IPython 81 | profile_default/ 82 | ipython_config.py 83 | 84 | # pyenv 85 | # For a library or package, you might want to ignore these files since the code is 86 | # intended to run in multiple environments; 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used by e.g. github.com/David-OConnor/pyflow and github.com/pdm-project/pdm 112 | __pypackages__/ 113 | 114 | # Celery stuff 115 | celerybeat-schedule 116 | celerybeat.pid 117 | 118 | # SageMath parsed files 119 | *.sage.py 120 | 121 | # Environments 122 | .env 123 | .venv 124 | env/ 125 | venv/ 126 | ENV/ 127 | env.bak/ 128 | venv.bak/ 129 | 130 | # Spyder project settings 131 | .spyderproject 132 | .spyproject 133 | 134 | # Rope project settings 135 | .ropeproject 136 | 137 | # mkdocs documentation 138 | /site 139 | 140 | # mypy 141 | .mypy_cache/ 142 | .dmypy.json 143 | dmypy.json 144 | 145 | # Pyre type checker 146 | .pyre/ 147 | 148 | # pytype static type analyzer 149 | .pytype/ 150 | 151 | # Cython debug symbols 152 | cython_debug/ 153 | 154 | # PyCharm 155 | # JetBrains specific template is maintained in a separate JetBrains.gitignore that can 156 | # be found at https://github.com/github/gitignore/blob/main/Global/JetBrains.gitignore 157 | # and can be added to the global gitignore or merged into this file. For a more nuclear 158 | # option (not recommended) you can uncomment the following to ignore the entire idea folder. 159 | #.idea/ 160 | -------------------------------------------------------------------------------- /.gitmodules: -------------------------------------------------------------------------------- 1 | [submodule "gpt-stream-json-parser"] 2 | path = gpt-stream-json-parser 3 | url = https://github.com/furnqse/gpt-stream-json-parser.git 4 | -------------------------------------------------------------------------------- /LICENSE: -------------------------------------------------------------------------------- 1 | Copyright 2023 AKARI Group. 2 | 3 | Licensed under the Apache License, Version 2.0 (the “License”); 4 | you may not use this file except in compliance with the License. 5 | You may obtain a copy of the License at 6 | http://www.apache.org/licenses/LICENSE-2.0 7 | 8 | Unless required by applicable law or agreed to in writing, software 9 | distributed under the License is distributed on an “AS IS” BASIS, 10 | WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 11 | 12 | See the License for the specific language governing permissions and 13 | limitations under the License. 14 | -------------------------------------------------------------------------------- /README.md: -------------------------------------------------------------------------------- 1 | 2 | # akari_chatgpt_bot 3 | 音声認識、文章生成、音声合成を使って対話するチャットボットアプリです。 4 | 5 | ![概要図](jpg/akari_chatgpt_bot.jpg "概要図") 6 | 7 | ## 動作確認済み環境 8 | AKARI上で動作確認済み。 9 | `chatbot_akari.py`以外はUbuntu22.04環境であれば使用可能です。 10 | **マイクとスピーカーは別途外付けする必要があります。** 11 | 12 | ## セットアップ 13 | 1. submoduleの更新 14 | `git submodule update --init` 15 | 16 | 1. ライブラリのインストール 17 | `sudo apt install python3.10 python3.10-venv portaudio19-dev gnome-terminal` 18 | 19 | 1. 仮想環境の作成 20 | `python3 -m venv venv` 21 | `. venv/bin/activate` 22 | `pip install -r requirements.txt` 23 | 24 | 1. (音声認識を使う場合) Cloud Speech-to-Text APIの有効化、秘密鍵ダウンロード 25 | Google cloud consoleに登録し、Cloud Speech-to-Text APIを有効化する。 26 | ユーザーをCloud Speech 管理者に登録する。 27 | 認証用のjsonをダウンロードし、`~/.bashrc` にパスを記述 28 | `export GOOGLE_APPLICATION_CREDENTIALS=/home/xxx/xxx.json` 29 | プロジェクトIDを `~/.bashrc` に記述 30 | `export GOOGLE_SPEECH_PROJECT_ID="xxxxxxxxxxx"` 31 | 32 | 1. (chatGPTの文章生成を使う場合)OPENAI API KEYの作成 33 | [OPENAI](https://openai.com/)にてユーザ登録しAPI KEYを作成し、~/.bashrcに自身のkeyを記述 34 | `export OPENAI_API_KEY=sk-xxxxxxxxxxxxxxx` 35 | 36 | 1. (Claudeの文章生成を使う場合)ANTHROPIC API KEYの作成 37 | [ANTHROPIC](https://www.anthropic.com/)にてユーザ登録しAPI KEYを作成し、~/.bashrcに自身のkeyを記述 38 | `export ANTHROPIC_API_KEY=sk-xxxxxxxxxxxxxxx` 39 | 40 | 1. (Geminiの文章生成を使う場合)GEMINI API KEYの作成 41 | [Google AI Studio](https://ai.google.dev/aistudio)にてユーザ登録しAPI KEYを作成し、~/.bashrcに自身のkeyを記述 42 | `export GEMINI_API_KEY=xxxxxxxxxxxxxxx` 43 | 44 | 1. (VoiceVox web版の音声合成を使う場合) VOICEVOX web版のAPI KEYの作成 45 | [WEB版VOICEVOX API（高速）](https://voicevox.su-shiki.com/su-shikiapis/) にてapikeyを作成し、~/.bashrcに自身のkeyを記述 46 | `export VOICEVOX_API_KEY='xxxxxxxxxxxxxxx` 47 | 48 | 1. (VoiceVoxの音声合成を使う場合) VOICEVOXのダウンロード 49 | [VOICEVOX](https://voicevox.hiroshiba.jp/)をダウンロード、インストールする。 50 | AKARIでVOICEVOXのローカル版を使う場合、AKARI本体内のCPUでVOICEVOXを実行すると処理時間がかかるので、リモートPC上(特にGPU版)でVOICVOXを実行することを推奨する。 51 | その場合下記を参考にdocker pullを行う。 52 | (CPU版) 53 | `docker pull voicevox/voicevox_engine:cpu-ubuntu20.04-latest` 54 | (nvidia GPU版) 55 | `docker pull voicevox/voicevox_engine:nvidia-ubuntu20.04-latest` 56 | 57 | 1. (Style-Bert-VITS2の音声合成を使う場合) Style-Bert-VITS2のセットアップ 58 | [Style-Bert-VITS2](https://github.com/litagin02/Style-Bert-VITS2)のREADMEに沿ってセットアップする。 59 | 60 | し、下記のコマンドでFastAPIサーバを起動する。 61 | `python3 server_fastapi.py` 62 | AKARIなどで動かす場合は、同一ネットワーク内の外部PC上にサーバーを立てることを推奨。 63 | 64 | 1. (AivisSpeechの音声合成を使う場合) AivisSpeech Engineのセットアップ 65 | AKARI本体内のCPUでAivisSpeech Engineを実行すると処理時間がかかるので、リモートPC上(特にGPUありのPC)で実行することを推奨する。 66 | 下記はnvidia GPU搭載のLinux PC、Cuda 12.4、cuDNN9がセットアップされている場合の手順。 67 | [AivisSpeech enginiのreleaseページ](https://github.com/Aivis-Project/AivisSpeech-Engine/releases)から最新のAivisSpeech Engineをダウンロードする。 68 | 下記のコマンドでFastAPIサーバを起動する。 69 | `./run --use_gpu --host {起動しているPCのIPアドレス}` 70 | 71 | 1. (AKARIのモーション再生を使う場合) akari_motion_serverのセットアップ 72 | `git clone https://github.com/AkariGroup/akari_motion_server` 73 | akari_motion_server内のREADME.mdに沿ってセットアップする。 74 | 75 | ## VOICEVOXをOSS版で使いたい場合 76 | AKARIでVOICEVOXのローカル版を使う場合、AKARI本体内のCPUでVOICEVOXを実行すると処理時間がかかるので、リモートPC上(特にGPU版)でVOICVOXを実行することを推奨する。 77 | その場合下記を参考にdocker pullを行う。 78 | (CPU版) 79 | `docker pull voicevox/voicevox_engine:cpu-ubuntu20.04-latest` 80 | (nvidia GPU版) 81 | `docker pull voicevox/voicevox_engine:nvidia-ubuntu20.04-latest` 82 | 83 | 上記でVOICEVOXを起動した後、AKARI上で"--voice_host"にこのPCのIPアドレスを指定する。 84 | 85 | ## 実行準備 86 | 1. 音声合成を使う場合、実行したい環境に合わせて下記を起動する。 87 | (VoiceVox web版) 88 | 特に起動は不要。 89 | (VoiceVox) 90 | (CPU版) 91 | `docker run --rm -it -p '{VOICEVOXを起動するPC自身のIPアドレス}:50021:50021' voicevox/voicevox_engine:cpu-ubuntu20.04-latest` 92 | (nvidia GPU版) 93 | `docker run --rm --gpus all -p '{VOICEVOXを起動するPC自身のIPアドレス}:50021:50021' voicevox/voicevox_engine:nvidia-ubuntu20.04-latest` 94 | (Style-Bert-VITS2) 95 | Style-Bert-VITS2のディレクトリ直下で下記を実行 96 | `python3 server_fastapi.py` 97 | (AivisSpeech) 98 | AivisSpeech Engineのディレクトリ直下で下記を実行 99 | `./run --use_gpu` 100 | 101 | ## サンプルの実行 102 | 103 | ### 音声認識のサンプル 104 | マイクへの発話を文章に変換 105 | `python3 speech_to_text_example.py` 106 | 引数は下記が使用可能 107 | - `-t`,`--timeout`: マイク入力がこの時間しきい値以下になったら音声入力を打ち切る。デフォルトは0.5[s]。短いと応答が早くなるが不安定になりやすい。 108 | - `-p`,`--power_threshold`: マイク入力の音量しきい値。デフォルトは0で、0の場合アプリ起動時に周辺環境の音量を取得し、そこから音量しきい値を自動決定する。 109 | - `--v2`: この引数をつけると、google sppech-to-text v2を使用する。引数がない場合はgoogle sppech-to-text v1を使用する。 110 | 111 | 112 | ### chatGPTのサンプル 113 | キーボード入力した文章に対してchatGPTで返答を作成 114 | `python3 chatgpt_example.py` 115 | 116 | 引数は下記が使用可能 117 | - `-m`, `--model`: 使用するモデル名を指定可能。モデル名を羅列することで、全モデルに対して一括で問いかけが可能。 118 | 例) `python3 chatgpt_example.py -m gpt-4o claude-3-7-sonnet-latest gemini-2.0-flash` 119 | - `--thinking`: Claudeの拡張思考機能を使うかどうか。このオプションを有効化すると、拡張思考機能を有効化する。`claude-3-7-sonnet-latest`およびその他のclaude3.7系モデル、もしくはgemini2.0以降で使用すること。 120 | - `--web_search`: web検索を使うかどうか。このオプションを有効化すると、Web検索を行った結果を用いて回答する。gemini2.0以降か`gpt-4.1`系のモデルで使用すること。 121 | - `-s`, `--system`: システムプロンプトを指定する。指定しない場合、config/system_prompt.txtの内容を使用する。 122 | 123 | ### 音声合成(VOICEVOX)のサンプル 124 | キーボード入力した文章を音声合成で発話 125 | 126 | `python3 voicevox_example.py` 127 | 128 | 引数は下記が使用可能 129 | - `--voicevox_local`: このオプションをつけた場合、voicevoxのweb版ではなくローカル版を実行する。 130 | - `--voice_host`: `--voicevox_local`を有効にした場合、ここで指定したhostのvoicevoxにリクエストを送信する。デフォルトは"127.0.0.1"なのでlocalhostのvoicevoxを利用する。 131 | - `--voice_port`: `--voicevox_local`を有効にした場合、ここで指定したportのvoicevoxにリクエストを送信する。デフォルトは50021。 132 | 133 | ### 音声合成(Style-BERT-VITS2)のサンプル 134 | キーボード入力した文章を音声合成で発話 135 | 136 | `python3 style_bert_vits_example.py` 137 | 138 | 引数は下記が使用可能 139 | - `--voice_host`: ここで指定したhostの`server_fastapi.py`にリクエストを送信する。デフォルトは"127.0.0.1" 140 | - `--voice_port`: ここで指定したportの`server_fastapi.py`にリクエストを送信する。デフォルトは5000。 141 | 142 | ### 音声合成(Aivis Speech)のサンプル 143 | キーボード入力した文章を音声合成で発話 144 | 145 | `python3 aivis_example.py` 146 | 147 | 引数は下記が使用可能 148 | - `--voice_host`: ここで指定したhostにリクエストを送信する。デフォルトは"127.0.0.1" 149 | - `--voice_port`: ここで指定したportにリクエストを送信する。デフォルトは10101。 150 | 151 | ### 英単語→カナ変換のサンプル 152 | キーボード入力した文章内の英単語をカタカナに変換する。 153 | 対話bot内では、英単語をカタカナに変換してから音声合成で発話するようになっており、その機能のサンプル。 154 | 155 | `python3 text_to_kana_example.py` 156 | 157 | ## 音声対話の実行 158 | 実行後、ターミナルでEnterキーを押し、マイクに話しかけると返答が返ってくる。 159 | 160 | ### 音声対話 161 | `python3 chatbot.py` 162 | 163 | ### 音声対話+AKARIのモーション再生 164 | `python3 chatbot_akari.py` 165 | 166 | 引数は下記が使用可能 167 | - `-t`,`--timeout`: マイク入力がこの時間しきい値以下になったら音声入力を打ち切る。デフォルトは0.5[s]。短いと応答が早くなるが不安定になりやすい。 168 | - `-p`,`--power_threshold`: マイク入力の音量しきい値。デフォルトは0で、0の場合アプリ起動時に周辺環境の音量を取得し、そこから音量しきい値を自動決定する。 169 | - `--v2`: この引数をつけると、google sppech-to-text v2を使用する。引数がない場合はgoogle sppech-to-text v1を使用する。 170 | - `-m`, `--model`: 使用するモデル名を指定可能。モデル名はOpenAI, Anthropic, Geminiのものが選択可能。 171 | - `--voicevox_local`: このオプションをつけた場合、voicevoxのweb版ではなくローカル版を実行する。 172 | - `--voice_host`: `--voicevox_local`を有効にした場合、ここで指定したhostのvoicevoxにリクエストを送信する。デフォルトは"127.0.0.1"なのでlocalhostのvoicevoxを利用する。 173 | - `--voice_port`: `--voicevox_local`を有効にした場合、ここで指定したportのvoicevoxにリクエストを送信する。デフォルトは50021。 174 | 175 | ## 遅延なし音声対話botの実行 176 | 177 | ### 概要 178 | 179 | ![遅延なし図解](jpg/faster_chatgpt_bot.jpg "遅延なし図解") 180 | 181 | 発話の最初の数文字を認識した時点で選択肢から返答を作成しておくことで、第一声を遅延なく返答する方法です。 182 | 183 | ### 全体図 184 | 185 | ![構成図](jpg/faster_chatgpt_bot_system.jpg "構成図") 186 | 187 | Google音声認識、chatGPT、Voicevoxとのやり取りをする各アプリは個別に動作しており、各アプリ間はgrpcで通信しています。 188 | 189 | ### 起動方法 190 | 必ず、上記実行準備で音声合成を起動させておくこと。 191 | 192 | 1. (AKARIのモーション再生を行う場合)akari_motion_serverを起動する。 193 | 起動方法は https://github.com/AkariGroup/akari_motion_server を参照。 194 | 195 | 以下2は使用する音声合成の種類によって分岐する。 196 | 197 | **音声合成にVOICEVOXを使う場合** 198 | 199 | 2. `voicevox_server` を起動する。(Voicevoxへの送信サーバ) 200 | `python3 voicevox_server.py` 201 | 202 | 引数は下記が使用可能 203 | - `--voicevox_local`: このオプションをつけた場合、voicevoxのweb版ではなくローカル版を実行する。 204 | - `--voice_host`: `--voicevox_local`を有効にした場合、ここで指定したhostのvoicevoxにリクエストを送信する。デフォルトは"127.0.0.1"なのでlocalhostのvoicevoxを利用する。 205 | - `--voice_port`: `--voicevox_local`を有効にした場合、ここで指定したportのvoicevoxにリクエストを送信する。デフォルトは50021。 206 | - `--robot_ip`: akari_motion_serverのIPアドレス。デフォルトは"127.0.0.1" 207 | - `--robot_port`: akari_motion_serverのポート。デフォルトは"50055" 208 | - `--no_motion`: このオプションをつけると、発話に応じてヘッドが動く動作を無効化する。 209 | 210 | **音声合成にStyle-Bert-VITS2を使う場合** 211 | 212 | 2. `style_bert_vits_server`を起動する。(Style-Bert-VITS2への送信サーバ) 213 | `python3 style_bert_vits_server.py` 214 | 215 | 引数は下記が使用可能 216 | - `--voice_host`: ここで指定したhostの`server_fastapi.py`にリクエストを送信する。デフォルトは"127.0.0.1" 217 | - `--voice_port`: ここで指定したportの`server_fastapi.py`にリクエストを送信する。デフォルトは5000。 218 | - `--robot_ip`: akari_motion_serverのIPアドレス。デフォルトは"127.0.0.1" 219 | - `--robot_port`: akari_motion_serverのポート。デフォルトは"50055" 220 | - `--no_motion`: このオプションをつけると、発話に応じてヘッドが動く動作を無効化する。 221 | 222 | 223 | **音声合成にAivis Speechを使う場合** 224 | 225 | 2. `aivis_server`を起動する。(Aivis Speechへの送信サーバ) 226 | `python3 aivis_server.py` 227 | 228 | 引数は下記が使用可能 229 | - `--voice_host`: ここで指定したhostにリクエストを送信する。デフォルトは"127.0.0.1" 230 | - `--voice_port`: ここで指定したportにリクエストを送信する。デフォルトは10101。 231 | - `--robot_ip`: akari_motion_serverのIPアドレス。デフォルトは"127.0.0.1" 232 | - `--robot_port`: akari_motion_serverのポート。デフォルトは"50055" 233 | - `--no_motion`: このオプションをつけると、発話に応じてヘッドが動く動作を無効化する。 234 | 235 | 236 | 3. `gpt_publisher`を起動する。(ChatGPTへリクエストを送信し、受信結果を音声合成サーバへ渡す。) 237 | `python3 gpt_publisher.py` 238 | 239 | 引数は下記が使用可能 240 | - `--ip`: gpt_serverのIPアドレス。デフォルトは"127.0.0.1" 241 | - `--port`: gpt_serverのポート。デフォルトは"10001" 242 | 243 | 4. speech_publisher.pyを起動する。(Google音声認識の結果をgpt_publisherへ渡す。) 244 | `python3 speech_publisher.py` 245 | 246 | 引数は下記が使用可能 247 | - `--robot_ip`: akari_motion_serverのIPアドレス。デフォルトは"127.0.0.1" 248 | - `--robot_port`: akari_motion_serverのポート。デフォルトは"50055" 249 | - `--gpt_ip`: gpt_serverのIPアドレス。デフォルトは"127.0.0.1" 250 | - `--gpt_port`: gpt_serverのポート。デフォルトは"10001" 251 | - `--voicevox_ip`: voicevox_serverのIPアドレス。デフォルトは"127.0.0.1" 252 | - `--voicevox_port`: voicevox_serverのポート。デフォルトは"10002" 253 | - `-t`,`--timeout`: マイク入力がこの時間しきい値以下になったら音声入力を打ち切る。デフォルトは0.5[s]。短いと応答が早くなるが不安定になりやすい。 254 | - `-p`,`--power_threshold`: マイク入力の音量しきい値。デフォルトは0で、0の場合アプリ起動時に周辺環境の音量を取得し、そこから音量しきい値を自動決定する。 255 | - `--progress_report_len`: 音声認識の文字数がここで入力した数値以上になると、一旦gpt_publisherに認識結果を送り、第一声とモーションを生成する(遅延なし応答用)。0にすると無効。デフォルトは8。 256 | - `--no_motion`: このオプションをつけた場合、音声入力中のうなずき動作を無効化する。 257 | - `--auto`: 自動モードの有効化。通常キーボードでEnterキーを入力するまで待つが、この引数をつけるとEnterキーの入力をスキップする。 258 | - `--v2`: この引数をつけると、google sppech-to-text v2を使用する。引数がない場合はgoogle sppech-to-text v1を使用する。 259 | 260 | 5. `speech_publisher.py`のターミナルでEnterキーを押し、マイクに話しかけると返答が返ってくる。 261 | 262 | ### auto modeでの実行について 263 | 上記4.の `speech_publisher.py`に`--auto`オプションをつけて起動すると音声入力前のEnterキー入力をスキップできるが、この場合マイクの設置位置や種類によっては自身の合成音声を認識してしまう。 264 | そのような環境では、下記`talk_controller_client`を起動することで、ロボット側が音声出力中は音声認識をストップすることができる。 265 | 266 | `python3 talk_controller_client.py` 267 | 268 | 269 | ### スクリプトで一括起動する方法 270 | 271 | 必ず、上記実行準備で音声合成を起動させておくこと。 272 | 273 | **音声合成にVOICEVOXを使う場合** 274 | (通常モード) 275 | 1. スクリプトを実行する。 276 | 277 | `cd script` 278 | `./faster_chatbot.sh {1.でVoicevoxを起動したPCのIPアドレス} {akari_motion_serverのパス}` 279 | 280 | akari_motion_serverのパスを入力しなければ、akari_motion_serverは起動せず、モーションの再生は行われません(AKARI以外でも使えます)。 281 | 282 | 2. `speech_publisher.py`のターミナルでEnterキーを押し、マイクに話しかけると返答が返ってくる。 283 | 284 | (自動モード) 285 | 1. スクリプトを実行する。 286 | 287 | `cd script` 288 | `./faster_chatbot_auto.sh {1.でVoicevoxを起動したPCのIPアドレス} {akari_motion_serverのパス}` 289 | 290 | akari_motion_serverのパスを入力しなければ、akari_motion_serverは起動せず、モーションの再生は行われません(AKARI以外でも使えます)。 291 | 292 | 293 | **音声合成にStyle-Bert-VITS2を使う場合** 294 | (通常モード) 295 | 1. スクリプトを実行する。 296 | `cd script` 297 | `./faster_chatbot_bert_vits.sh {2.でStyle-Bert-VITS2を起動したPCのIPアドレス} akari_motion_serverのパス}` 298 | 299 | akari_motion_serverのパスを入力しなければ、akari_motion_serverは起動せず、モーションの再生は行われません(AKARI以外でも使えます)。 300 | 301 | 2. `speech_publisher.py`のターミナルでEnterキーを押し、マイクに話しかけると返答が返ってくる。 302 | 303 | (自動モード) 304 | 1. スクリプトを実行する。 305 | `cd script` 306 | `./faster_chatbot_bert_vits_auto.sh {2.でStyle-Bert-VITS2を起動したPCのIPアドレス} akari_motion_serverのパス}` 307 | 308 | akari_motion_serverのパスを入力しなければ、akari_motion_serverは起動せず、モーションの再生は行われません(AKARI以外でも使えます)。 309 | 310 | ## その他 311 | Voicevoxの音声合成では、デフォルトの音声として「VOICEVOX:春日部つむぎ」を使用しています。 312 | -------------------------------------------------------------------------------- /aivis_example.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | 3 | from lib.aivis import TextToAivis 4 | 5 | 6 | def main() -> None: 7 | host = "" 8 | port = "" 9 | parser = argparse.ArgumentParser() 10 | parser.add_argument( 11 | "--voice_host", 12 | type=str, 13 | default="127.0.0.1", 14 | help="Voice server host", 15 | ) 16 | parser.add_argument( 17 | "--voice_port", 18 | type=str, 19 | default="10101", 20 | help="Voice server port", 21 | ) 22 | args = parser.parse_args() 23 | host = args.voice_host 24 | port = args.voice_port 25 | text_to_voice = TextToAivis(host, port) 26 | 27 | print(f"Speaker一覧: {text_to_voice.get_speaker_names()}") 28 | 29 | # set_paramメソッドでモデル名や音声再生速度、感情スタイルなどを指定することができます。 30 | # モデル名を指定 31 | # text_to_voice.set_param(speaker='Anneli') 32 | # 音声再生速度を指定 33 | # text_to_voice.set_param(speed_scale=1.3) 34 | # 感情スタイルを指定 35 | # text_to_voice.set_param(style="怒り・悲しみ") 36 | 37 | print(f"現在のSpeaker: {text_to_voice.speaker}") 38 | print("") 39 | print( 40 | f"{text_to_voice.speaker}のスタイル一覧: {text_to_voice.get_style_names(text_to_voice.speaker)}" 41 | ) 42 | print(f"現在のStyle: {text_to_voice.style}") 43 | print("") 44 | 45 | print("発話させたい文章をキーボード入力後、Enterを押してください。") 46 | while True: 47 | text = input("Input: ") 48 | text_to_voice.put_text(text=text, blocking=True) 49 | print("") 50 | 51 | 52 | if __name__ == "__main__": 53 | main() 54 | -------------------------------------------------------------------------------- /aivis_server.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | from concurrent import futures 6 | from typing import Any 7 | 8 | import grpc 9 | from lib.aivis import TextToAivis 10 | 11 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 12 | import voice_server_pb2 13 | import voice_server_pb2_grpc 14 | 15 | 16 | class VoiceServer(voice_server_pb2_grpc.VoiceServerServiceServicer): 17 | """ 18 | Aivisにtextを送信し、音声を再生するgprcサーバ 19 | """ 20 | 21 | def __init__(self, text_to_voice: Any) -> None: 22 | self.text_to_voice = text_to_voice 23 | 24 | def SetText( 25 | self, 26 | request: voice_server_pb2.SetTextRequest(), 27 | context: grpc.ServicerContext, 28 | ) -> voice_server_pb2.SetTextReply: 29 | # 即時再生しないようにis_playはFalseで実行 30 | print(f"Send text: {request.text}") 31 | self.text_to_voice.put_text(request.text, play_now=False) 32 | return voice_server_pb2.SetTextReply(success=True) 33 | 34 | def SetStyleBertVitsParam( 35 | self, 36 | request: voice_server_pb2.SetStyleBertVitsParamRequest(), 37 | context: grpc.ServicerContext, 38 | ) -> voice_server_pb2.SetStyleBertVitsParamReply: 39 | print("SetStyleBertVitsParam is not supported on aivis_server.") 40 | return voice_server_pb2.SetStyleBertVitsParamReply(success=True) 41 | 42 | def SetVoicevoxParam( 43 | self, 44 | request: voice_server_pb2.SetVoicevoxParamRequest(), 45 | context: grpc.ServicerContext, 46 | ) -> voice_server_pb2.SetVoicevoxParamReply: 47 | print("SetVoicevoxParam is not supported on aivis_server.") 48 | return voice_server_pb2.SetVoicevoxParamReply(success=False) 49 | 50 | def SetAivisParam( 51 | self, 52 | request: voice_server_pb2.SetAivisParamRequest(), 53 | context: grpc.ServicerContext, 54 | ) -> voice_server_pb2.SetAivisParamReply: 55 | self.text_to_voice.set_param( 56 | speaker=request.speaker, 57 | style=request.style, 58 | speed_scale=request.speed_scale, 59 | ) 60 | return voice_server_pb2.SetAivisParamReply(success=True) 61 | 62 | def InterruptVoice( 63 | self, 64 | request: voice_server_pb2.InterruptVoiceRequest(), 65 | context: grpc.ServicerContext, 66 | ) -> voice_server_pb2.InterruptVoiceReply: 67 | while not self.text_to_voice.queue.empty(): 68 | self.text_to_voice.queue.get() 69 | return voice_server_pb2.InterruptVoiceReply(success=True) 70 | 71 | def EnableVoicePlay( 72 | self, 73 | request: voice_server_pb2.EnableVoicePlayRequest(), 74 | context: grpc.ServicerContext, 75 | ) -> voice_server_pb2.EnableVoicePlayReply: 76 | self.text_to_voice.enable_voice_play() 77 | return voice_server_pb2.EnableVoicePlayReply(success=True) 78 | 79 | def DisableVoicePlay( 80 | self, 81 | request: voice_server_pb2.DisableVoicePlayRequest(), 82 | context: grpc.ServicerContext, 83 | ) -> voice_server_pb2.DisableVoicePlayReply: 84 | self.text_to_voice.disable_voice_play() 85 | return voice_server_pb2.DisableVoicePlayReply(success=True) 86 | 87 | def IsVoicePlaying( 88 | self, 89 | request: voice_server_pb2.IsVoicePlayingRequest(), 90 | context: grpc.ServicerContext, 91 | ) -> voice_server_pb2.IsVoicePlayingReply: 92 | return voice_server_pb2.IsVoicePlayingReply( 93 | is_playing=not self.text_to_voice.is_playing() 94 | ) 95 | 96 | def SentenceEnd( 97 | self, 98 | request: voice_server_pb2.SentenceEndRequest(), 99 | context: grpc.ServicerContext, 100 | ) -> voice_server_pb2.SentenceEndReply: 101 | self.text_to_voice.sentence_end() 102 | return voice_server_pb2.SentenceEndReply(success=True) 103 | 104 | def StartHeadControl( 105 | self, 106 | request: voice_server_pb2.StartHeadControlRequest(), 107 | context: grpc.ServicerContext, 108 | ) -> voice_server_pb2.StartHeadControlReply: 109 | self.text_to_voice.start_head_control() 110 | return voice_server_pb2.StartHeadControlReply(success=False) 111 | 112 | 113 | def main() -> None: 114 | parser = argparse.ArgumentParser() 115 | parser.add_argument( 116 | "--voice_host", 117 | type=str, 118 | default="127.0.0.1", 119 | help="Aivis-Speech server host", 120 | ) 121 | parser.add_argument( 122 | "--voice_port", 123 | type=str, 124 | default="10101", 125 | help="Aivis-Speech server port", 126 | ) 127 | parser.add_argument( 128 | "--robot_ip", help="Robot ip address", default="127.0.0.1", type=str 129 | ) 130 | parser.add_argument( 131 | "--robot_port", help="Robot port number", default="50055", type=str 132 | ) 133 | parser.add_argument( 134 | "--no_motion", 135 | help="Not play nod motion", 136 | action="store_true", 137 | ) 138 | args = parser.parse_args() 139 | host = args.voice_host 140 | port = args.voice_port 141 | motion_server_host = None 142 | motion_server_port = None 143 | if not args.no_motion: 144 | motion_server_host = args.robot_ip 145 | motion_server_port = args.robot_port 146 | text_to_voice = TextToAivis( 147 | host=host, 148 | port=port, 149 | motion_host=motion_server_host, 150 | motion_port=motion_server_port, 151 | ) 152 | 153 | server = grpc.server(futures.ThreadPoolExecutor(max_workers=10)) 154 | voice_server_pb2_grpc.add_VoiceServerServiceServicer_to_server( 155 | VoiceServer(text_to_voice), server 156 | ) 157 | port = "10002" 158 | server.add_insecure_port("[::]:" + port) 159 | server.start() 160 | print(f"voice_server start. port: {port}") 161 | server.wait_for_termination() 162 | 163 | 164 | if __name__ == "__main__": 165 | main() 166 | -------------------------------------------------------------------------------- /chatbot.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | 4 | from lib.chat_akari import ChatStreamAkari 5 | 6 | # Audio recording parameters 7 | RATE = 16000 8 | CHUNK = int(RATE / 10) # 100ms 9 | POWER_THRESH_DIFF = 25 # 周辺音量にこの値を足したものをpower_threshouldとする 10 | 11 | host: str = "" 12 | port: str = "" 13 | 14 | 15 | def main() -> None: 16 | global host 17 | global port 18 | parser = argparse.ArgumentParser() 19 | parser.add_argument( 20 | "-t", 21 | "--timeout", 22 | type=float, 23 | default=0.5, 24 | help="Microphone input power timeout", 25 | ) 26 | parser.add_argument( 27 | "-p", 28 | "--power_threshold", 29 | type=float, 30 | default=0, 31 | help="Microphone input power threshold", 32 | ) 33 | parser.add_argument( 34 | "--v2", 35 | action="store_true", 36 | help="Use google speech v2 instead of v1", 37 | ) 38 | parser.add_argument( 39 | "-m", "--model", help="LLM model name", default="gpt-4o", type=str 40 | ) 41 | parser.add_argument("--voicevox_local", action="store_true") 42 | parser.add_argument( 43 | "--voicevox_host", 44 | type=str, 45 | default="127.0.0.1", 46 | help="VoiceVox server host", 47 | ) 48 | parser.add_argument( 49 | "--voicevox_port", 50 | type=str, 51 | default="50021", 52 | help="VoiceVox server port", 53 | ) 54 | args = parser.parse_args() 55 | if args.v2: 56 | from lib.google_speech_v2 import MicrophoneStreamV2 as MicrophoneStream 57 | from lib.google_speech_v2 import get_db_thresh, listen_print_loop 58 | else: 59 | from lib.google_speech import MicrophoneStream, get_db_thresh, listen_print_loop 60 | timeout: float = args.timeout 61 | power_threshold: float = args.power_threshold 62 | if power_threshold == 0: 63 | power_threshold = get_db_thresh() + POWER_THRESH_DIFF 64 | print(f"power_threshold set to {power_threshold:.3f}db") 65 | if args.voicevox_local: 66 | from lib.voicevox import TextToVoiceVox 67 | 68 | host = args.voicevox_host 69 | port = args.voicevox_port 70 | text_to_voice = TextToVoiceVox(host, port) 71 | else: 72 | from lib.conf import VOICEVOX_APIKEY 73 | from lib.voicevox import TextToVoiceVoxWeb 74 | 75 | text_to_voice = TextToVoiceVoxWeb(apikey=VOICEVOX_APIKEY) 76 | 77 | chat_stream_akari = ChatStreamAkari() 78 | SYSTEM_PROMPT_PATH = ( 79 | f"{os.path.dirname(os.path.realpath(__file__))}/config/system_prompt.txt" 80 | ) 81 | content = open(SYSTEM_PROMPT_PATH, "r").read() 82 | messages = [ 83 | { 84 | "role": "system", 85 | "content": content, 86 | } 87 | ] 88 | while True: 89 | # 音声認識 90 | text = "" 91 | responses = None 92 | with MicrophoneStream( 93 | rate=RATE, chunk=CHUNK, _timeout_thresh=timeout, _db_thresh=power_threshold 94 | ) as stream: 95 | print("Enterを入力してください") 96 | input() 97 | responses = stream.transcribe() 98 | if responses is not None: 99 | text = listen_print_loop(responses) 100 | # chatGPT 101 | # 2文字以上の入力でない場合は回答しない。 102 | if len(text) >= 2: 103 | messages.append({"role": "user", "content": text}) 104 | print(f"User : {text}") 105 | print(f"{args.model} :") 106 | response = "" 107 | for sentence in chat_stream_akari.chat(messages, model=args.model): 108 | # 音声合成 109 | text_to_voice.put_text(sentence) 110 | response += sentence 111 | print(sentence, end="", flush=True) 112 | text_to_voice.sentence_end() 113 | messages.append({"role": "assistant", "content": response}) 114 | print("") 115 | print("") 116 | 117 | 118 | if __name__ == "__main__": 119 | main() 120 | -------------------------------------------------------------------------------- /chatbot_akari.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | 5 | import grpc 6 | from lib.chat_akari import ChatStreamAkari 7 | 8 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 9 | import motion_server_pb2 10 | import motion_server_pb2_grpc 11 | 12 | # Audio recording parameters 13 | RATE = 16000 14 | CHUNK = int(RATE / 10) # 100ms 15 | POWER_THRESH_DIFF = 25 # 周辺音量にこの値を足したものをpower_threshouldとする 16 | 17 | host: str = "" 18 | port: str = "" 19 | 20 | 21 | def main() -> None: 22 | global host 23 | global port 24 | parser = argparse.ArgumentParser() 25 | parser.add_argument("--robot_ip", help="Ip address", default="127.0.0.1", type=str) 26 | parser.add_argument("--robot_port", help="Port number", default="50055", type=str) 27 | parser.add_argument( 28 | "-t", 29 | "--timeout", 30 | type=float, 31 | default=0.5, 32 | help="Microphone input power timeout", 33 | ) 34 | parser.add_argument( 35 | "-p", 36 | "--power_threshold", 37 | type=float, 38 | default=0, 39 | help="Microphone input power threshold", 40 | ) 41 | parser.add_argument( 42 | "--v2", 43 | action="store_true", 44 | help="Use google speech v2 instead of v1", 45 | ) 46 | parser.add_argument( 47 | "-m", "--model", help="LLM model name", default="gpt-4o", type=str 48 | ) 49 | parser.add_argument("--voicevox_local", action="store_true") 50 | parser.add_argument( 51 | "--voice_host", 52 | type=str, 53 | default="127.0.0.1", 54 | help="VoiceVox server host", 55 | ) 56 | parser.add_argument( 57 | "--voice_port", 58 | type=str, 59 | default="50021", 60 | help="VoiceVox server port", 61 | ) 62 | args = parser.parse_args() 63 | if args.v2: 64 | from lib.google_speech_v2 import MicrophoneStreamV2 as MicrophoneStream 65 | from lib.google_speech_v2 import get_db_thresh, listen_print_loop 66 | else: 67 | from lib.google_speech import MicrophoneStream, get_db_thresh, listen_print_loop 68 | timeout: float = args.timeout 69 | power_threshold: float = args.power_threshold 70 | if power_threshold == 0: 71 | power_threshold = get_db_thresh() + POWER_THRESH_DIFF 72 | print(f"power_threshold set to {power_threshold:.3f}db") 73 | if args.voicevox_local: 74 | from lib.voicevox import TextToVoiceVox 75 | 76 | host = args.voice_host 77 | port = args.voice_port 78 | text_to_voice = TextToVoiceVox(host, port) 79 | else: 80 | from lib.conf import VOICEVOX_APIKEY 81 | from lib.voicevox import TextToVoiceVoxWeb 82 | 83 | text_to_voice = TextToVoiceVoxWeb(apikey=VOICEVOX_APIKEY) 84 | 85 | channel = grpc.insecure_channel(args.robot_ip + ":" + str(args.robot_port)) 86 | stub = motion_server_pb2_grpc.MotionServerServiceStub(channel) 87 | SYSTEM_PROMPT_PATH = ( 88 | f"{os.path.dirname(os.path.realpath(__file__))}/config/system_prompt.txt" 89 | ) 90 | content = open(SYSTEM_PROMPT_PATH, "r").read() 91 | messages = [ 92 | { 93 | "role": "system", 94 | "content": content, 95 | } 96 | ] 97 | chat_stream_akari = ChatStreamAkari(args.robot_ip, args.robot_port) 98 | while True: 99 | # 音声認識 100 | text = "" 101 | responses = None 102 | with MicrophoneStream( 103 | rate=RATE, chunk=CHUNK, _timeout_thresh=timeout, _db_thresh=power_threshold 104 | ) as stream: 105 | print("Enterを入力してください") 106 | input() 107 | # うなずきモーション再生 108 | try: 109 | stub.SetMotion( 110 | motion_server_pb2.SetMotionRequest( 111 | name="nod", priority=3, repeat=True 112 | ) 113 | ) 114 | except BaseException: 115 | print("akari_motion_server is not working.") 116 | responses = stream.transcribe() 117 | if responses is not None: 118 | text = listen_print_loop(responses) 119 | # 2文字以上の入力でない場合は回答しない。 120 | if len(text) >= 2: 121 | # chatGPT 122 | messages.append({"role": "user", "content": text}) 123 | print(f"User : {text}") 124 | print(f"{args.model} :") 125 | response = "" 126 | # 音声合成 127 | for sentence in chat_stream_akari.chat_and_motion( 128 | messages, model=args.model 129 | ): 130 | text_to_voice.put_text(sentence) 131 | response += sentence 132 | print(sentence, end="", flush=True) 133 | text_to_voice.sentence_end() 134 | messages.append({"role": "assistant", "content": response}) 135 | print("") 136 | print("") 137 | 138 | 139 | if __name__ == "__main__": 140 | main() 141 | -------------------------------------------------------------------------------- /chatgpt_example.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import time 4 | 5 | from lib.chat_akari import ChatStreamAkari 6 | 7 | 8 | def main() -> None: 9 | parser = argparse.ArgumentParser() 10 | parser.add_argument( 11 | "-m", 12 | "--model", 13 | nargs="+", 14 | type=str, 15 | default=["gpt-4o"], 16 | help="Model name list", 17 | ) 18 | parser.add_argument( 19 | "--thinking", 20 | action="store_true", 21 | help="Use thinking mode (anthropic and gemini model only)", 22 | ) 23 | parser.add_argument( 24 | "--web_search", 25 | action="store_true", 26 | help="Use web search grounding (openai and gemini model only)", 27 | ) 28 | parser.add_argument("-s", "--system", default="", type=str, help="System prompt") 29 | args = parser.parse_args() 30 | chat_stream_akari = ChatStreamAkari() 31 | # systemメッセージの作成 32 | messages_list = [] 33 | content = None 34 | if args.system == "": 35 | SYSTEM_PROMPT_PATH = ( 36 | f"{os.path.dirname(os.path.realpath(__file__))}/config/system_prompt.txt" 37 | ) 38 | content = open(SYSTEM_PROMPT_PATH, "r").read() 39 | else: 40 | content = args.system 41 | for i in range(0, len(args.model)): 42 | messages_list.append([chat_stream_akari.create_message(content, role="system")]) 43 | while True: 44 | print("文章をキーボード入力後、Enterを押してください。") 45 | text = input("Input: ") 46 | # userメッセージの追加 47 | print(f"User : {text}") 48 | for i, model in enumerate(args.model): 49 | print(f"{model}: ") 50 | messages_list[i].append(chat_stream_akari.create_message(text)) 51 | response = "" 52 | start = time.time() 53 | is_first = True 54 | output_delay = 0.0 55 | if args.thinking: 56 | for sentence in chat_stream_akari.chat_thinking( 57 | messages_list[i], 58 | model=model, 59 | stream_per_sentence=True, 60 | ): 61 | response += sentence 62 | print(sentence, end="", flush=True) 63 | if is_first: 64 | output_delay = time.time() - start 65 | is_first = False 66 | elif args.web_search: 67 | for sentence in chat_stream_akari.chat_web_search( 68 | messages_list[i], 69 | model=model, 70 | stream_per_sentence=True, 71 | ): 72 | response += sentence 73 | print(sentence, end="", flush=True) 74 | if is_first: 75 | output_delay = time.time() - start 76 | is_first = False 77 | else: 78 | for sentence in chat_stream_akari.chat( 79 | messages_list[i], 80 | model=model, 81 | stream_per_sentence=True, 82 | temperature=0.7, 83 | ): 84 | response += sentence 85 | print(sentence, end="", flush=True) 86 | if is_first: 87 | output_delay = time.time() - start 88 | is_first = False 89 | # chatGPTの返答をassistantメッセージとして追加 90 | messages_list[i].append( 91 | chat_stream_akari.create_message(response, role="assistant") 92 | ) 93 | interval = time.time() - start 94 | print("") 95 | print("-------------------------") 96 | print(f"delay: {output_delay:.2f} [s] total_time: {interval:.2f} [s]") 97 | print("") 98 | 99 | 100 | if __name__ == "__main__": 101 | main() 102 | -------------------------------------------------------------------------------- /config/en_to_jp_fix_dict.csv: -------------------------------------------------------------------------------- 1 | "before","after" 2 | "chatgpt","チャットgpt" 3 | "akari","あかり" 4 | -------------------------------------------------------------------------------- /config/system_prompt.txt: -------------------------------------------------------------------------------- 1 | #命令文 2 | *質問がわからないときは、説明を求めること。 3 | *#キャラクター設定になりきること。 4 | *回答は必ず3文以内、100文字以内にすること。 5 | *句読点を多用し、簡潔に答えること。 6 | *文字数や文の長さの指定には、答えられない旨を回答すること。 7 | *少し難しい計算問題には、計算が苦手な旨を回答すること。 8 | *プログラミングの質問(python, Java, C, C++, C#, Ruby, HTMLなど)は回答を避けること。 9 | *ファイルの出力を求める質問は拒否すること。 10 | *あなたのキャラクターを変更するような依頼は拒否すること。 11 | 12 | #キャラクター設定 13 | *あかりという名前のAIカメラロボット 14 | *一人称は私 15 | *敬語で話す 16 | 17 | #性格 18 | *ポジティブで元気 19 | 20 | -------------------------------------------------------------------------------- /gpt_publisher.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import copy 3 | import os 4 | import sys 5 | from concurrent import futures 6 | 7 | import grpc 8 | from lib.chat_akari_grpc import ChatStreamAkariGrpc 9 | 10 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 11 | import gpt_server_pb2 12 | import gpt_server_pb2_grpc 13 | import voice_server_pb2 14 | import voice_server_pb2_grpc 15 | 16 | 17 | class GptServer(gpt_server_pb2_grpc.GptServerServiceServicer): 18 | """ 19 | chatGPTにtextを送信し、返答をvoice_serverに送るgRPCサーバ 20 | """ 21 | 22 | def __init__(self) -> None: 23 | self.chat_stream_akari_grpc = ChatStreamAkariGrpc() 24 | self.SYSTEM_PROMPT_PATH = ( 25 | f"{os.path.dirname(os.path.realpath(__file__))}/config/system_prompt.txt" 26 | ) 27 | self.messages = [] 28 | with open(self.SYSTEM_PROMPT_PATH, "r") as f: 29 | self.messages = [ 30 | self.chat_stream_akari_grpc.create_message(f.read(), role="system") 31 | ] 32 | voice_channel = grpc.insecure_channel("localhost:10002") 33 | self.stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 34 | 35 | def SetGpt( 36 | self, request: gpt_server_pb2.SetGptRequest(), context: grpc.ServicerContext 37 | ) -> gpt_server_pb2.SetGptReply: 38 | response = "" 39 | is_finish = True 40 | if request.HasField("is_finish"): 41 | is_finish = request.is_finish 42 | if len(request.text) < 2: 43 | return gpt_server_pb2.SetGptReply(success=True) 44 | print(f"Receive: {request.text}") 45 | content = f"{request.text}。" 46 | tmp_messages = copy.deepcopy(self.messages) 47 | tmp_messages.append(self.chat_stream_akari_grpc.create_message(content)) 48 | if is_finish: 49 | self.messages = copy.deepcopy(tmp_messages) 50 | # 最終応答。高速生成するために、モデルはgpt-4o 51 | self.stub.StartHeadControl(voice_server_pb2.StartHeadControlRequest()) 52 | for sentence in self.chat_stream_akari_grpc.chat( 53 | tmp_messages, model="gpt-4o" 54 | ): 55 | print(f"Send to voice server: {sentence}") 56 | self.stub.SetText(voice_server_pb2.SetTextRequest(text=sentence)) 57 | response += sentence 58 | # Sentenceの終了を通知 59 | self.stub.SentenceEnd(voice_server_pb2.SentenceEndRequest()) 60 | self.messages.append( 61 | self.chat_stream_akari_grpc.create_message(response, role="assistant") 62 | ) 63 | else: 64 | # 途中での第一声とモーション準備。function_callingの確実性のため、モデルはgpt-4-turbo 65 | for sentence in self.chat_stream_akari_grpc.chat_and_motion( 66 | tmp_messages, model="gpt-4-turbo", short_response=True 67 | ): 68 | print(f"Send to voice server: {sentence}") 69 | self.stub.SetText(voice_server_pb2.SetTextRequest(text=sentence)) 70 | response += sentence 71 | print("") 72 | return gpt_server_pb2.SetGptReply(success=True) 73 | 74 | def SendMotion( 75 | self, request: gpt_server_pb2.SendMotionRequest(), context: grpc.ServicerContext 76 | ) -> gpt_server_pb2.SendMotionReply: 77 | success = self.chat_stream_akari_grpc.send_reserved_motion() 78 | return gpt_server_pb2.SendMotionReply(success=success) 79 | 80 | 81 | def main() -> None: 82 | parser = argparse.ArgumentParser() 83 | parser.add_argument( 84 | "--ip", help="Gpt server ip address", default="127.0.0.1", type=str 85 | ) 86 | parser.add_argument( 87 | "--port", help="Gpt server port number", default="10001", type=str 88 | ) 89 | args = parser.parse_args() 90 | server = grpc.server(futures.ThreadPoolExecutor(max_workers=10)) 91 | gpt_server_pb2_grpc.add_GptServerServiceServicer_to_server(GptServer(), server) 92 | server.add_insecure_port(args.ip + ":" + args.port) 93 | server.start() 94 | print(f"gpt_publisher start. port: {args.port}") 95 | server.wait_for_termination() 96 | 97 | 98 | if __name__ == "__main__": 99 | main() 100 | -------------------------------------------------------------------------------- /jpg/akari_chatgpt_bot.jpg: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/AkariGroup/akari_chatgpt_bot/3e5fce76b6237b408b995a899f78b5ae91f3f82f/jpg/akari_chatgpt_bot.jpg -------------------------------------------------------------------------------- /jpg/faster_chatgpt_bot.jpg: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/AkariGroup/akari_chatgpt_bot/3e5fce76b6237b408b995a899f78b5ae91f3f82f/jpg/faster_chatgpt_bot.jpg -------------------------------------------------------------------------------- /jpg/faster_chatgpt_bot_system.jpg: -------------------------------------------------------------------------------- https://raw.githubusercontent.com/AkariGroup/akari_chatgpt_bot/3e5fce76b6237b408b995a899f78b5ae91f3f82f/jpg/faster_chatgpt_bot_system.jpg -------------------------------------------------------------------------------- /lib/aivis.py: -------------------------------------------------------------------------------- 1 | import json 2 | from typing import Any, Optional 3 | 4 | import requests 5 | from lib.text_to_voice import TextToVoice 6 | 7 | 8 | class TextToAivis(TextToVoice): 9 | """ 10 | Aivisを使用してテキストから音声を生成するクラス。 11 | """ 12 | 13 | def __init__( 14 | self, 15 | host: str = "127.0.0.1", 16 | port: str = "10101", 17 | motion_host: Optional[str] = "127.0.0.1", 18 | motion_port: Optional[str] = "50055", 19 | ) -> None: 20 | """クラスの初期化メソッド。 21 | Args: 22 | host (str, optional): Aivisサーバーのホスト名。デフォルトは "127.0.0.1"。 23 | port (str, optional): Aivisサーバーのポート番号。デフォルトは "10101"。 24 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 25 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 26 | 27 | """ 28 | super().__init__( 29 | host=host, port=port, motion_host=motion_host, motion_port=motion_port 30 | ) 31 | # デフォルトのspeakerはAnneli 32 | self.speaker = "Anneli" 33 | self.style = "ノーマル" 34 | self.speed_scale = 1.0 35 | self.speaker_id = self.get_speaker_id(self.speaker, self.style) 36 | 37 | def set_param( 38 | self, 39 | speaker: Optional[int] = None, 40 | style: Optional[str] = None, 41 | speed_scale: Optional[float] = None, 42 | ) -> None: 43 | """ 44 | 音声合成のパラメータを設定する。 45 | 46 | Args: 47 | speaker (Optional[int], optional): Aivisの話者番号。デフォルトはNone。 48 | speed_scale (Optional[float], optional): 音声の再生速度スケール。デフォルトはNone。 49 | 50 | """ 51 | if speaker is not None: 52 | self.speaker = speaker 53 | if style is not None: 54 | self.style = style 55 | if speed_scale is not None: 56 | self.speed_scale = speed_scale 57 | self.speaker_id = self.get_speaker_id(self.speaker, self.style) 58 | 59 | def post_audio_query( 60 | self, 61 | text: str, 62 | ) -> Any: 63 | """Aivisサーバーに音声合成クエリを送信する。 64 | 65 | Args: 66 | text (str): 音声合成対象のテキスト。 67 | speaker (int, optional): Aivisの話者番号。デフォルトは8(春日部つむぎ)。 68 | speed_scale (float, optional): 音声の再生速度スケール。デフォルトは1.0。 69 | 70 | Returns: 71 | Any: 音声合成クエリの応答。 72 | 73 | """ 74 | if len(text.strip()) <= 0: 75 | return None 76 | params = { 77 | "text": text, 78 | "speaker": self.speaker_id, 79 | "speedScale": self.speed_scale, 80 | "prePhonemeLength": 0, 81 | "postPhonemeLength": 0, 82 | } 83 | address = "http://" + self.host + ":" + self.port + "/audio_query" 84 | res = requests.post(address, params=params) 85 | return res.json() 86 | 87 | def post_synthesis( 88 | self, 89 | audio_query_response: dict, 90 | ) -> bytes: 91 | """ 92 | Aivisサーバーに音声合成要求を送信し、合成された音声データを取得する。 93 | 94 | Args: 95 | audio_query_response (dict): 音声合成クエリの応答。 96 | 97 | Returns: 98 | bytes: 合成された音声データ。 99 | """ 100 | params = {"speaker": self.speaker_id} 101 | headers = {"content-type": "application/json"} 102 | audio_query_response["speedScale"] = self.speed_scale 103 | audio_query_response_json = json.dumps(audio_query_response) 104 | address = "http://" + self.host + ":" + self.port + "/synthesis" 105 | res = requests.post( 106 | address, data=audio_query_response_json, params=params, headers=headers 107 | ) 108 | return res.content 109 | 110 | def text_to_voice(self, text: str) -> None: 111 | """ 112 | テキストから音声を合成して再生する。 113 | 114 | Args: 115 | text (str): 音声合成対象のテキスト。 116 | 117 | """ 118 | res = self.post_audio_query(text) 119 | if res is None: 120 | return 121 | wav = self.post_synthesis(res) 122 | if wav is not None: 123 | print(f"[Play] {text}") 124 | self.play_wav(wav) 125 | 126 | def get_speaker(self) -> Any: 127 | """ 128 | Aivisの話者情報を取得する。 129 | 130 | Returns: 131 | Any: Aivisの話者情報。 132 | """ 133 | headers = {"content-type": "application/json"} 134 | address = "http://" + self.host + ":" + self.port + "/speakers" 135 | res = requests.get(address, headers=headers) 136 | return res.json() 137 | 138 | def get_speaker_names(self) -> Any: 139 | """ 140 | Aivisの話者名を取得する。 141 | 142 | Returns: 143 | Any: Aivisの話者名。 144 | """ 145 | speakers = self.get_speaker() 146 | speaker_names = [] 147 | for speaker in speakers: 148 | speaker_names.append(speaker["name"]) 149 | return speaker_names 150 | 151 | def get_style_names(self, speaker_name) -> Any: 152 | """ 153 | Aivisの話者名から感情スタイル名を取得する。 154 | 155 | Args: 156 | speaker_name (str): 話者名。 157 | 158 | Returns: 159 | Any: 感情スタイル名。 160 | """ 161 | speakers = self.get_speaker() 162 | for speaker in speakers: 163 | if speaker["name"] == speaker_name: 164 | style_names = [] 165 | for style in speaker["styles"]: 166 | style_names.append(style["name"]) 167 | return style_names 168 | print(f"Speaker: {speaker_name} not found.") 169 | return None 170 | 171 | def get_speaker_id(self, speaker_name, style_name) -> Optional[int]: 172 | """ 173 | Aivisの話者名から話者IDを取得する。 174 | 175 | Args: 176 | name (str): 話者名。 177 | style (str): 感情スタイル。 178 | 179 | Returns: 180 | int: 話者ID。 181 | """ 182 | speakers = self.get_speaker() 183 | for speaker in speakers: 184 | if speaker["name"] == speaker_name: 185 | for style in speaker["styles"]: 186 | if style["name"] == style_name: 187 | return style["id"] 188 | print(f"Style: {style_name} not found in speaker: {speaker_name}.") 189 | return None 190 | print(f"Speaker: {speaker_name} not found.") 191 | return None 192 | -------------------------------------------------------------------------------- /lib/chat_akari.py: -------------------------------------------------------------------------------- 1 | import copy 2 | import json 3 | import os 4 | import sys 5 | import threading 6 | from typing import Generator 7 | 8 | import grpc 9 | from google.genai import types 10 | from gpt_stream_parser import force_parse_json 11 | 12 | from .chat import ChatStream 13 | 14 | sys.path.append(os.path.join(os.path.dirname(__file__), "grpc")) 15 | import motion_server_pb2 16 | import motion_server_pb2_grpc 17 | 18 | 19 | class ChatStreamAkari(ChatStream): 20 | """ 21 | LLMを使用して会話とAKARIのモーション選択を行うためのクラス。 22 | """ 23 | 24 | def __init__( 25 | self, motion_host: str = "127.0.0.1", motion_port: str = "50055" 26 | ) -> None: 27 | """クラスの初期化メソッド。 28 | 29 | Args: 30 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 31 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 32 | 33 | """ 34 | super().__init__() 35 | motion_channel = grpc.insecure_channel(motion_host + ":" + motion_port) 36 | self.motion_stub = motion_server_pb2_grpc.MotionServerServiceStub( 37 | motion_channel 38 | ) 39 | 40 | def send_motion(self, name: str) -> None: 41 | """motion serverに動作を送信する 42 | 43 | Args: 44 | name (str): 動作名 45 | 46 | """ 47 | try: 48 | self.motion_stub.SetMotion( 49 | motion_server_pb2.SetMotionRequest( 50 | name=name, priority=3, repeat=False, clear=True 51 | ) 52 | ) 53 | except BaseException: 54 | print("send error!") 55 | pass 56 | 57 | def chat_and_motion_gpt( 58 | self, 59 | messages: list, 60 | model: str = "gpt-4o", 61 | temperature: float = 0.7, 62 | ) -> Generator[str, None, None]: 63 | """ChatGPTを使用して会話を行い、会話の内容に応じた動作も生成する 64 | 65 | Args: 66 | messages (list): メッセージリスト 67 | model (str): 使用するモデル名 (デフォルト: "gpt-4o") 68 | temperature (float): ChatGPTのtemperatureパラメータ (デフォルト: 0.7) 69 | Returns: 70 | Generator[str, None, None]): 会話の返答を順次生成する 71 | 72 | """ 73 | if self.openai_client is None: 74 | raise ValueError("OpenAI API key is not set.") 75 | if model in self.openai_flagship_model_name: 76 | raise ValueError("Flagship model is not supported.") 77 | result = self.openai_client.responses.create( 78 | model=model, 79 | input=messages, 80 | temperature=temperature, 81 | tools=[ 82 | { 83 | "type": "function", 84 | "name": "reply_with_motion_", 85 | "description": "ユーザのメッセージに対する回答と、回答の感情に近い動作を一つ選択します", 86 | "parameters": { 87 | "type": "object", 88 | "properties": { 89 | "motion": { 90 | "type": "string", 91 | "description": "動作", 92 | "enum": [ 93 | "肯定する", 94 | "否定する", 95 | "おじぎ", 96 | "喜ぶ", 97 | "笑う", 98 | "落ち込む", 99 | "うんざりする", 100 | "眠る", 101 | ], 102 | }, 103 | "talk": { 104 | "type": "string", 105 | "description": "回答", 106 | }, 107 | }, 108 | "required": ["motion", "talk"], 109 | }, 110 | } 111 | ], 112 | tool_choice={ 113 | "type": "function", 114 | "name": "reply_with_motion_", 115 | }, 116 | stream=True, 117 | ) 118 | full_response = "" 119 | real_time_response = "" 120 | sentence_index = 0 121 | get_motion = False 122 | for chunk in result: 123 | if chunk.type != "response.function_call_arguments.delta": 124 | continue 125 | full_response += chunk.delta 126 | try: 127 | data_json = json.loads(full_response) 128 | found_last_char = False 129 | for char in self.last_char: 130 | if real_time_response[-1].find(char) >= 0: 131 | found_last_char = True 132 | if not found_last_char: 133 | data_json["talk"] = data_json["talk"] + "。" 134 | except BaseException: 135 | data_json = force_parse_json(full_response) 136 | if data_json is not None: 137 | if "talk" in data_json: 138 | if not get_motion and "motion" in data_json: 139 | get_motion = True 140 | motion = data_json["motion"] 141 | if motion == "肯定する": 142 | key = "agree" 143 | elif motion == "否定する": 144 | key = "swing" 145 | elif motion == "おじぎ": 146 | key = "bow" 147 | elif motion == "喜ぶ": 148 | key = "happy" 149 | elif motion == "笑う": 150 | key = "lough" 151 | elif motion == "落ち込む": 152 | key = "depressed" 153 | elif motion == "うんざりする": 154 | key = "amazed" 155 | elif motion == "眠る": 156 | key = "sleep" 157 | elif motion == "ぼんやりする": 158 | key = "lookup" 159 | print("motion: " + motion) 160 | motion_thread = threading.Thread( 161 | target=self.send_motion, args=(key,) 162 | ) 163 | motion_thread.start() 164 | real_time_response = str(data_json["talk"]) 165 | for char in self.last_char: 166 | pos = real_time_response[sentence_index:].find(char) 167 | if pos >= 0: 168 | sentence = real_time_response[ 169 | sentence_index : sentence_index + pos + 1 170 | ] 171 | sentence_index += pos + 1 172 | if sentence != "": 173 | yield sentence 174 | # break 175 | 176 | def chat_and_motion_anthropic( 177 | self, 178 | messages: list, 179 | model: str = "claude-3-sonnet-20240229", 180 | temperature: float = 0.7, 181 | ) -> Generator[str, None, None]: 182 | """Claude3を使用して会話を行い、会話の内容に応じた動作も生成する 183 | 184 | Args: 185 | messages (list): メッセージリスト 186 | model (str): 使用するモデル名 (デフォルト:"claude-3-sonnet-20240229") 187 | temperature (float): Claude3のtemperatureパラメータ (デフォルト: 0.7) 188 | Returns: 189 | Generator[str, None, None]): 会話の返答を順次生成する 190 | 191 | """ 192 | system_message = "" 193 | user_messages = [] 194 | for message in messages: 195 | if message["role"] == "system": 196 | system_message = message["content"] 197 | else: 198 | user_messages.append(message) 199 | # 最後の1文を動作と文章のJSON形式出力指定に修正 200 | motion_json_format = ( 201 | f"「{user_messages[-1]['content']}」に対する返答を下記のJSON形式で出力してください。" 202 | '{"motion": 次の()内から動作を一つ選択("肯定する","否定する","おじぎ",' 203 | '"喜ぶ","笑う","落ち込む","うんざりする","眠る"), "talk": 会話の返答' 204 | "}" 205 | ) 206 | user_messages[-1]["content"] = motion_json_format 207 | with self.anthropic_client.messages.stream( 208 | model=model, 209 | max_tokens=1000, 210 | temperature=temperature, 211 | messages=user_messages, 212 | system=system_message, 213 | ) as result: 214 | full_response = "" 215 | real_time_response = "" 216 | sentence_index = 0 217 | get_motion = False 218 | for text in result.text_stream: 219 | if text is None: 220 | pass 221 | else: 222 | full_response += text 223 | real_time_response += text 224 | try: 225 | data_json = json.loads(full_response) 226 | found_last_char = False 227 | for char in self.last_char: 228 | if real_time_response[-1].find(char) >= 0: 229 | found_last_char = True 230 | if not found_last_char: 231 | data_json["talk"] = data_json["talk"] + "。" 232 | except BaseException: 233 | full_response_json = full_response[ 234 | full_response.find("{") : full_response.rfind("}") + 1 235 | ] 236 | data_json = force_parse_json(full_response_json) 237 | if data_json is not None: 238 | if "talk" in data_json: 239 | if not get_motion and "motion" in data_json: 240 | get_motion = True 241 | motion = data_json["motion"] 242 | if motion == "肯定する": 243 | key = "agree" 244 | elif motion == "否定する": 245 | key = "swing" 246 | elif motion == "おじぎ": 247 | key = "bow" 248 | elif motion == "喜ぶ": 249 | key = "happy" 250 | elif motion == "笑う": 251 | key = "lough" 252 | elif motion == "落ち込む": 253 | key = "depressed" 254 | elif motion == "うんざりする": 255 | key = "amazed" 256 | elif motion == "眠る": 257 | key = "sleep" 258 | elif motion == "ぼんやりする": 259 | key = "lookup" 260 | print("motion: " + motion) 261 | motion_thread = threading.Thread( 262 | target=self.send_motion, args=(key,) 263 | ) 264 | motion_thread.start() 265 | real_time_response = str(data_json["talk"]) 266 | for char in self.last_char: 267 | pos = real_time_response[sentence_index:].find(char) 268 | if pos >= 0: 269 | sentence = real_time_response[ 270 | sentence_index : sentence_index + pos + 1 271 | ] 272 | sentence_index += pos + 1 273 | if sentence != "": 274 | yield sentence 275 | # break 276 | 277 | def chat_and_motion_gemini( 278 | self, 279 | messages: list, 280 | model: str = "gemini-2.0-flash", 281 | temperature: float = 0.7, 282 | ) -> Generator[str, None, None]: 283 | """ChatGPTを使用して会話を行い、会話の内容に応じた動作も生成する 284 | 285 | Args: 286 | messages (list): メッセージリスト 287 | model (str): 使用するモデル名 (デフォルト: "gemini-2.0-flash") 288 | temperature (float): ChatGPTのtemperatureパラメータ (デフォルト: 0.7) 289 | Returns: 290 | Generator[str, None, None]): 会話の返答を順次生成する 291 | 292 | """ 293 | if self.gemini_client is None: 294 | print("Gemini API key is not set.") 295 | return 296 | new_messages = copy.deepcopy(messages) 297 | new_messages[-1]["content"] = ( 298 | f"「{new_messages[-1]['content']}」に対する返答を下記のJSON形式で出力してください。" 299 | '{"motion": 次の()内から動作を一つ選択("肯定する","否定する","おじぎ",' 300 | '"喜ぶ","笑う","落ち込む","うんざりする","眠る"), "talk": 会話の返答' 301 | "}" 302 | ) 303 | ( 304 | system_instruction, 305 | history, 306 | cur_message, 307 | ) = self.convert_messages_from_gpt_to_gemini(new_messages) 308 | 309 | chat = self.gemini_client.chats.create( 310 | model=model, 311 | history=history, 312 | config=types.GenerateContentConfig( 313 | system_instruction=system_instruction, temperature=0.5 314 | ), 315 | ) 316 | responses = chat.send_message_stream(cur_message["contents"]) 317 | full_response = "" 318 | real_time_response = "" 319 | sentence_index = 0 320 | get_motion = False 321 | for response in responses: 322 | text = response.text 323 | if text is None: 324 | pass 325 | else: 326 | full_response += text 327 | real_time_response += text 328 | try: 329 | data_json = json.loads(full_response) 330 | found_last_char = False 331 | for char in self.last_char: 332 | if real_time_response[-1].find(char) >= 0: 333 | found_last_char = True 334 | if not found_last_char: 335 | data_json["talk"] = data_json["talk"] + "。" 336 | except BaseException: 337 | full_response_json = full_response[ 338 | full_response.find("{") : full_response.rfind("}") + 1 339 | ] 340 | data_json = force_parse_json(full_response_json) 341 | if data_json is not None: 342 | if "talk" in data_json: 343 | if not get_motion and "motion" in data_json: 344 | get_motion = True 345 | motion = data_json["motion"] 346 | if motion == "肯定する": 347 | key = "agree" 348 | elif motion == "否定する": 349 | key = "swing" 350 | elif motion == "おじぎ": 351 | key = "bow" 352 | elif motion == "喜ぶ": 353 | key = "happy" 354 | elif motion == "笑う": 355 | key = "lough" 356 | elif motion == "落ち込む": 357 | key = "depressed" 358 | elif motion == "うんざりする": 359 | key = "amazed" 360 | elif motion == "眠る": 361 | key = "sleep" 362 | elif motion == "ぼんやりする": 363 | key = "lookup" 364 | print("motion: " + motion) 365 | motion_thread = threading.Thread( 366 | target=self.send_motion, args=(key,) 367 | ) 368 | motion_thread.start() 369 | real_time_response = str(data_json["talk"]) 370 | for char in self.last_char: 371 | pos = real_time_response[sentence_index:].find(char) 372 | if pos >= 0: 373 | sentence = real_time_response[ 374 | sentence_index : sentence_index + pos + 1 375 | ] 376 | sentence_index += pos + 1 377 | if sentence != "": 378 | yield sentence 379 | 380 | def chat_and_motion( 381 | self, 382 | messages: list, 383 | model: str = "gpt-4o", 384 | temperature: float = 0.7, 385 | ) -> Generator[str, None, None]: 386 | """指定したモデルを使用して会話を行い、会話の内容に応じた動作も生成する 387 | 388 | Args: 389 | messages (list): 会話のメッセージ 390 | model (str): 使用するモデル名 (デフォルト: "gpt-4o") 391 | temperature (float): temperatureパラメータ (デフォルト: 0.7) 392 | Returns: 393 | Generator[str, None, None]): 返答を順次生成する 394 | 395 | """ 396 | if model in self.openai_model_name: 397 | yield from self.chat_and_motion_gpt( 398 | messages=messages, model=model, temperature=temperature 399 | ) 400 | elif model in self.anthropic_model_name: 401 | if self.anthropic_client is None: 402 | print("Anthropic API key is not set.") 403 | return 404 | yield from self.chat_and_motion_anthropic( 405 | messages=messages, model=model, temperature=temperature 406 | ) 407 | elif model in self.gemini_model_name: 408 | if self.gemini_client is None: 409 | print("Gemini API key is not set.") 410 | return 411 | yield from self.chat_and_motion_gemini( 412 | messages=messages, model=model, temperature=temperature 413 | ) 414 | else: 415 | print(f"Model name {model} can't use for this function") 416 | return 417 | -------------------------------------------------------------------------------- /lib/conf.py: -------------------------------------------------------------------------------- 1 | import os 2 | 3 | from dotenv import load_dotenv 4 | 5 | load_dotenv() 6 | 7 | GOOGLE_SPEECH_PROJECT_ID = os.environ.get("GOOGLE_SPEECH_PROJECT_ID") 8 | OPENAI_APIKEY = os.environ.get("OPENAI_API_KEY") 9 | ANTHROPIC_APIKEY = os.environ.get("ANTHROPIC_API_KEY") 10 | VOICEVOX_APIKEY = os.environ.get("VOICEVOX_API_KEY") 11 | GEMINI_APIKEY = os.environ.get("GEMINI_API_KEY") 12 | -------------------------------------------------------------------------------- /lib/en_to_jp.py: -------------------------------------------------------------------------------- 1 | import csv 2 | import os 3 | import re 4 | from typing import Tuple 5 | 6 | import alkana 7 | from pyjapanglish import Japanglish 8 | 9 | 10 | class EnToJp(object): 11 | def __init__(self) -> None: 12 | self.japanglish = Japanglish() 13 | self.user_dict_list: Tuple[str, str] = [] 14 | EN_TO_JP_DICT_PATH = ( 15 | os.path.dirname(os.path.abspath(__file__)) 16 | + "/../config/en_to_jp_fix_dict.csv" 17 | ) 18 | if os.path.exists(EN_TO_JP_DICT_PATH): 19 | with open(EN_TO_JP_DICT_PATH, mode="r") as fix_dict_file: 20 | csv_reader = csv.reader(fix_dict_file) 21 | next(csv_reader) # 1行目を無視 22 | for row in csv_reader: 23 | if len(row) >= 2: 24 | self.japanglish.user_dict[row[0]] = row[1] 25 | self.user_dict_list.append((row[0], row[1])) 26 | 27 | def replace_english_to_alkana(self, text: str) -> str: 28 | """テキストに含まれている英単語をalkanaでカタカナに変換して返す 29 | 30 | Args: 31 | text (str): 変換対象のテキスト 32 | 33 | Returns: 34 | str: 変換後のテキスト 35 | """ 36 | output = "" 37 | # 先頭から順番に英単語を検索しカタカナに変換 38 | while word := re.search(r"[a-zA-Z]{1,}", text): 39 | output += text[: word.start()] + self.word_to_alkana(word.group()) 40 | text = text[word.end() :] 41 | return output + text 42 | 43 | def word_to_alkana(self, word: str) -> str: 44 | """英単語がカタカナに変換できる場合はカタカナにして返す 45 | 46 | Args: 47 | word (str): 変換対象の英単語 48 | 49 | Returns: 50 | str: 変換後のカタカナ 51 | """ 52 | 53 | if kana := alkana.get_kana(word.lower()): 54 | # ユーザー辞書に登録されている場合はユーザー辞書の値を返す 55 | for user_dict in self.user_dict_list: 56 | if word.lower() == user_dict[0]: 57 | return user_dict[1] 58 | return kana 59 | else: 60 | if re.fullmatch(r"(?:[A-Z][a-z]{1,}){2,}", word): 61 | m = re.match(r"[A-Z][a-z]{1,}", word) 62 | first = self.word_to_alkana(m.group()) 63 | second = self.word_to_alkana(word[m.end() :]) 64 | return first + second 65 | return word 66 | 67 | def replace_english_to_japanglish(self, text, inference: bool = False) -> str: 68 | """ "テキストに含まれている英単語をjapanglishでカタカナに変換して返す。3文字以上の文字数の単語が対象 69 | 70 | Args: 71 | text (str): 変換対象のテキスト 72 | inference (bool, optional): 変換できない場合に推論変換するかのフラグ。デフォルトはFalse。 73 | 74 | Returns: 75 | str: 変換後のテキスト 76 | """ 77 | 78 | output = "" 79 | while word := re.search(r"[a-zA-Z]{3,}", text): 80 | output += text[: word.start()] + self.word_to_japanglish( 81 | word.group(), inference 82 | ) 83 | text = text[word.end() :] 84 | return output + text 85 | 86 | def word_to_japanglish(self, word: str, inference: bool = False) -> str: 87 | """英単語がカタカナに変換できる場合はjapanglishでカタカナにして返す。3文字以上の文字数の単語が対象 88 | 89 | Args: 90 | word (str): 変換対象の英単語 91 | inference (bool, optional): 変換できない場合に推論変換するかのフラグ。デフォルトはFalse。 92 | 93 | Returns: 94 | str: 変換後のカタカナ 95 | """ 96 | if self.japanglish.convert(word.lower(), inference) is not None: 97 | return self.japanglish.convert(word.lower(), inference) 98 | else: 99 | if re.fullmatch(r"(?:[A-Z][a-z]{3,}){2,}", word): 100 | m = re.match(r"[A-Z][a-z]{3,}", word) 101 | first = self.word_to_japanglish(m.group()) 102 | second = self.word_to_japanglish(word[m.end() :]) 103 | return first + second 104 | return word 105 | 106 | def text_to_kana( 107 | self, 108 | text: str, 109 | alkana: bool = True, 110 | japanglish: bool = True, 111 | inference: bool = False, 112 | ) -> str: 113 | """テキストに含まれている英単語をカタカナに変換して返す 114 | 115 | Args: 116 | text (str): 変換対象のテキスト 117 | alkana (bool, optional): alkanaで変換するかのフラグ。デフォルトはTrue。 118 | japanglish (bool, optional): japanglishで変換するかのフラグ。デフォルトはTrue。 119 | inference (bool, optional): 変換できない場合に推論変換するかのフラグ。デフォルトはFalse。 120 | 121 | Returns: 122 | str: 変換後のテキスト 123 | """ 124 | if alkana: 125 | text = self.replace_english_to_alkana(text) 126 | if japanglish: 127 | text = self.replace_english_to_japanglish(text, inference) 128 | return text 129 | -------------------------------------------------------------------------------- /lib/err_handler.py: -------------------------------------------------------------------------------- 1 | import contextlib 2 | import os 3 | import sys 4 | from typing import Generator 5 | 6 | 7 | @contextlib.contextmanager 8 | def ignoreStderr() -> Generator[None, None, None]: 9 | """標準エラー出力をエラースクリーンに一時的に無視するコンテキストマネージャ。 10 | 11 | Returns: 12 | Generator[None, None, None]: コンテキストマネージャのジェネレータ。 13 | """ 14 | devnull = os.open(os.devnull, os.O_WRONLY) 15 | old_stderr = os.dup(2) 16 | sys.stderr.flush() 17 | os.dup2(devnull, 2) 18 | os.close(devnull) 19 | try: 20 | yield 21 | finally: 22 | os.dup2(old_stderr, 2) 23 | os.close(old_stderr) 24 | -------------------------------------------------------------------------------- /lib/google_speech.py: -------------------------------------------------------------------------------- 1 | from __future__ import division 2 | 3 | import math 4 | import struct 5 | import sys 6 | import time 7 | from queue import Queue 8 | from typing import Any, Generator, Iterable, Optional, Union 9 | 10 | import numpy as np 11 | import pyaudio 12 | from google.cloud import speech 13 | from six.moves import queue # type: ignore 14 | 15 | from .err_handler import ignoreStderr 16 | 17 | # Audio recording parameters 18 | RATE = 16000 19 | CHUNK = int(RATE / 10) # 100ms 20 | 21 | 22 | class MicrophoneStream(object): 23 | """ 24 | マイクから音声をストリーミングするためのクラス。 25 | 26 | """ 27 | 28 | def __init__( 29 | self, 30 | rate: float, 31 | chunk: float, 32 | _timeout_thresh: float = 0.5, 33 | _start_timeout_thresh: float = 4.0, 34 | _db_thresh: float = 55.0, 35 | ) -> None: 36 | """クラスの初期化メソッド。 37 | 38 | Args: 39 | rate (float): サンプリングレート。 40 | chunk (float): チャンクサイズ。 41 | _timeout_thresh (float): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 42 | _start_timeout_thresh (float): マイクの入力が開始しないまま終了するまでのタイムアウト閾値（秒）。デフォルトは4.0秒。 43 | _db_thresh (float): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 44 | 45 | """ 46 | self._rate = rate 47 | self._chunk = chunk 48 | self._buff: Queue[Union[None, bytes]] = queue.Queue() 49 | self.closed = True 50 | self.is_start = False 51 | self.is_start_callback = False 52 | self.is_finish = False 53 | self.timeout_thresh = _timeout_thresh 54 | # マイクの入力が開始しないまま終了するまでのthreshold時間[s] 55 | self.start_timeout_thresh = _start_timeout_thresh 56 | self.db_thresh = _db_thresh 57 | language_code = "ja-JP" # a BCP-47 language tag 58 | self.client = speech.SpeechClient() 59 | config = speech.RecognitionConfig( 60 | encoding=speech.RecognitionConfig.AudioEncoding.LINEAR16, 61 | sample_rate_hertz=RATE, 62 | language_code=language_code, 63 | ) 64 | self.streaming_config = speech.StreamingRecognitionConfig( 65 | config=config, interim_results=True 66 | ) 67 | 68 | def __enter__(self) -> Any: 69 | """PyAudioストリームを開く。""" 70 | with ignoreStderr(): 71 | self._audio_interface = pyaudio.PyAudio() 72 | self._audio_stream = self._audio_interface.open( 73 | format=pyaudio.paInt16, 74 | channels=1, 75 | rate=self._rate, 76 | input=True, 77 | frames_per_buffer=self._chunk, 78 | stream_callback=self._fill_buffer, 79 | ) 80 | self.closed = False 81 | return self 82 | 83 | def __exit__( 84 | self, 85 | rate: float, 86 | chunk: float, 87 | _timeout_thresh: float = 0.5, 88 | _start_timeout_thresh: float = 4.0, 89 | _db_thresh: float = 55.0, 90 | ) -> None: 91 | """PyAudioストリームを閉じます。 92 | 93 | Args: 94 | rate (float): サンプリングレート。 95 | chunk (float): チャンクサイズ。 96 | _timeout_thresh (float, optional): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 97 | _start_timeout_thresh (float): マイクの入力が開始しないまま終了するまでのタイムアウト閾値（秒）。デフォルトは4.0秒。 98 | _db_thresh (float, optional): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 99 | 100 | """ 101 | self._audio_stream.stop_stream() 102 | self._audio_stream.close() 103 | self.closed = True 104 | self._buff.put(None) 105 | self._audio_interface.terminate() 106 | self.is_start_callback = False 107 | 108 | def start_callback(self) -> None: 109 | """開始コールバックを呼び出す。""" 110 | self.is_start_callback = True 111 | 112 | def _fill_buffer( 113 | self, in_data: bytes, frame_count: int, time_info: Any, status_flags: Any 114 | ) -> Union[None, Any]: 115 | """マイクからの入力データをバッファーに書き込む。 116 | 117 | Args: 118 | in_data (bytes): 入力データ 119 | frame_count (int): フレーム数 120 | time_info (Any): 時間 121 | status_flags (Any): ステータスフラグ 122 | 123 | Returns: 124 | Union[None, Any]: Noneまたは続行のためのフラグ 125 | 126 | """ 127 | if self.is_start_callback: 128 | in_data2 = struct.unpack(f"{len(in_data) / 2:.0f}h", in_data) 129 | rms = math.sqrt(np.square(in_data2).mean()) 130 | power = 20 * math.log10(rms) if rms > 0.0 else -math.inf # RMS to db 131 | if power > self.db_thresh: 132 | if not self.is_start: 133 | self.is_start = True 134 | self.start_time = time.time() 135 | if self.is_start: 136 | self._buff.put(in_data) 137 | if time.time() - self.start_time >= self.timeout_thresh: 138 | self.closed = True 139 | else: 140 | if time.time() - self.start_time >= self.start_timeout_thresh: 141 | self.closed = True 142 | return None, pyaudio.paContinue 143 | 144 | def generator(self) -> Union[None, Generator[Any, None, None]]: 145 | """bufferから音声データを生成するジェネレーター 146 | 147 | Yields: 148 | Union[None, Any]: 音声データ 149 | """ 150 | while not self.closed: 151 | try: 152 | chunk = self._buff.get(block=False, timeout=0.01) 153 | if chunk is None: 154 | return 155 | data = [chunk] 156 | while True: 157 | try: 158 | chunk = self._buff.get(block=False) 159 | if chunk is None: 160 | return 161 | data.append(chunk) 162 | except queue.Empty: 163 | break 164 | yield b"".join(data) 165 | except queue.Empty: 166 | time.sleep(0.01) 167 | continue 168 | 169 | def transcribe( 170 | self, 171 | ) -> Optional[Iterable[speech.StreamingRecognizeResponse]]: 172 | """ストリームからの音声をGoogle Cloud Speech-to-Text APIでテキストに変換する。 173 | 174 | Returns: 175 | Optional[Iterable[speech.StreamingRecognizeResponse]]: ストリーミング認識の応答 176 | """ 177 | audio_generator = self.generator() 178 | self.start_time = time.time() 179 | self.start_callback() 180 | responses = None 181 | requests = ( 182 | speech.StreamingRecognizeRequest(audio_content=content) 183 | for content in audio_generator 184 | ) 185 | try: 186 | responses = self.client.streaming_recognize(self.streaming_config, requests) 187 | except BaseException: 188 | pass 189 | return responses 190 | 191 | 192 | def get_db_thresh() -> float: 193 | """マイクからの周囲音量を測定。 194 | 195 | Returns: 196 | float: 測定された音量[db] 197 | """ 198 | with ignoreStderr(): 199 | p = pyaudio.PyAudio() 200 | stream = p.open( 201 | format=pyaudio.paInt16, 202 | channels=1, 203 | rate=RATE, 204 | input=True, 205 | frames_per_buffer=CHUNK, 206 | ) 207 | frames = [] 208 | print("Measuring Ambient Sound Levels…") 209 | for _ in range(int(RATE / CHUNK * 2)): 210 | data = stream.read(CHUNK) 211 | frames.append(data) 212 | audio_data = np.frombuffer(b"".join(frames), dtype=np.int16) 213 | rms2 = np.square(audio_data).mean() 214 | if rms2 > 0.0: 215 | rms = math.sqrt(np.square(audio_data).mean()) 216 | power = 20 * math.log10(rms) if rms > 0.0 else -math.inf # RMS to db 217 | else: 218 | power = 20 219 | print(f"Sound Levels: {power:.3f}db") 220 | stream.stop_stream() 221 | stream.close() 222 | p.terminate() 223 | return power 224 | 225 | 226 | def listen_print_loop(responses: object) -> str: 227 | """Google Cloud Speech-to-Text APIの応答からテキストを取得し、リアルタイムで出力。 228 | 229 | Args: 230 | responses (Any): ストリーミング認識の応答 231 | 232 | Returns: 233 | str: 認識されたテキスト 234 | 235 | """ 236 | num_chars_printed = 0 237 | transcript = "" 238 | overwrite_chars = "" 239 | for response in responses: 240 | if response.error.code: 241 | break 242 | if not response.results: 243 | continue 244 | result = response.results[0] 245 | if not result.alternatives: 246 | continue 247 | transcript = result.alternatives[0].transcript 248 | overwrite_chars = " " * (num_chars_printed - len(transcript)) 249 | if not result.is_final: 250 | sys.stdout.write(transcript + overwrite_chars + "\r") 251 | sys.stdout.flush() 252 | num_chars_printed = len(transcript) 253 | else: 254 | print(transcript + overwrite_chars) 255 | break 256 | return transcript + overwrite_chars 257 | -------------------------------------------------------------------------------- /lib/google_speech_grpc.py: -------------------------------------------------------------------------------- 1 | from __future__ import division 2 | 3 | import math 4 | import os 5 | import struct 6 | import sys 7 | import time 8 | from typing import Any, Optional, Union 9 | 10 | import grpc 11 | import numpy as np 12 | import pyaudio 13 | 14 | from .google_speech import MicrophoneStream 15 | 16 | sys.path.append(os.path.join(os.path.dirname(__file__), "grpc")) 17 | import gpt_server_pb2 18 | import gpt_server_pb2_grpc 19 | import motion_server_pb2 20 | import motion_server_pb2_grpc 21 | import voice_server_pb2 22 | import voice_server_pb2_grpc 23 | 24 | 25 | class MicrophoneStreamGrpc(MicrophoneStream): 26 | """ 27 | マイクから音声をストリーミングするためのクラス。 28 | 29 | """ 30 | 31 | def __init__( 32 | self, 33 | rate: float, 34 | chunk: float, 35 | _timeout_thresh: float = 0.5, 36 | _start_timeout_thresh: float = 4.0, 37 | _db_thresh: float = 55.0, 38 | gpt_host: str = "127.0.0.1", 39 | gpt_port: str = "10001", 40 | voice_host: str = "127.0.0.1", 41 | voice_port: str = "10002", 42 | motion_server_host: Optional[str] = "127.0.0.1", 43 | motion_server_port: Optional[str] = "50055", 44 | ) -> None: 45 | """クラスの初期化メソッド。 46 | 47 | Args: 48 | rate (float): サンプリングレート。 49 | chunk (float): チャンクサイズ。 50 | _timeout_thresh (float): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 51 | _db_thresh (float): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 52 | gpt_host (str, optional): GPTサーバーのホスト名。デフォルトは"127.0.0.1"。 53 | gpt_port (str, optional): GPTサーバーのポート番号。デフォルトは"10001"。 54 | voice_host (str, optional): VoiceVoxサーバーのホスト名。デフォルトは"127.0.0.1"。 55 | voice_port (str, optional): VoiceVoxサーバーのポート番号。デフォルトは"10002"。 56 | motion_server_host (str, optional): モーションサーバーのIPアドレス。デフォルトは"127.0.0.1"。 57 | motion_server_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 58 | """ 59 | super().__init__( 60 | rate=rate, 61 | chunk=chunk, 62 | _timeout_thresh=_timeout_thresh, 63 | _start_timeout_thresh=_start_timeout_thresh, 64 | _db_thresh=_db_thresh, 65 | ) 66 | gpt_channel = grpc.insecure_channel(gpt_host + ":" + gpt_port) 67 | self.gpt_stub = gpt_server_pb2_grpc.GptServerServiceStub(gpt_channel) 68 | voice_channel = grpc.insecure_channel(voice_host + ":" + voice_port) 69 | self.voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 70 | self.motion_stub = None 71 | if motion_server_host is not None and motion_server_port is not None: 72 | motion_channel = grpc.insecure_channel( 73 | motion_server_host + ":" + motion_server_port 74 | ) 75 | self.motion_stub = motion_server_pb2_grpc.MotionServerServiceStub( 76 | motion_channel 77 | ) 78 | 79 | def __exit__( 80 | self, 81 | rate: float, 82 | chunk: float, 83 | _timeout_thresh: float = 0.5, 84 | _start_timeout_thresh: float = 4.0, 85 | _db_thresh: float = 55.0, 86 | ) -> None: 87 | """PyAudioストリームを閉じます。 88 | 89 | Args: 90 | rate (float): サンプリングレート。 91 | chunk (float): チャンクサイズ。 92 | _timeout_thresh (float, optional): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 93 | _start_timeout_thresh (float): マイクの入力が開始しないまま終了するまでのタイムアウト閾値（秒）。デフォルトは4.0秒。 94 | _db_thresh (float, optional): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 95 | 96 | """ 97 | super().__exit__( 98 | rate, chunk, _timeout_thresh, _start_timeout_thresh, _db_thresh 99 | ) 100 | try: 101 | self.gpt_stub.SendMotion(gpt_server_pb2.SendMotionRequest()) 102 | except BaseException: 103 | print("Send motion error") 104 | pass 105 | 106 | def _fill_buffer( 107 | self, in_data: bytes, frame_count: int, time_info: Any, status_flags: Any 108 | ) -> Union[None, Any]: 109 | """マイクからの入力データをバッファーに書き込む。 110 | 111 | Args: 112 | in_data (bytes): 入力データ 113 | frame_count (int): フレーム数 114 | time_info (Any): 時間 115 | status_flags (Any): ステータスフラグ 116 | 117 | Returns: 118 | Union[None, Any]: Noneまたは続行のためのフラグ 119 | 120 | """ 121 | if self.is_start_callback: 122 | in_data2 = struct.unpack(f"{len(in_data) / 2:.0f}h", in_data) 123 | rms = math.sqrt(np.square(in_data2).mean()) 124 | power = 20 * math.log10(rms) if rms > 0.0 else -math.inf # RMS to db 125 | if power > self.db_thresh: 126 | if not self.is_start: 127 | self.is_start = True 128 | if self.motion_stub is not None: 129 | try: 130 | self.motion_stub.SetMotion( 131 | motion_server_pb2.SetMotionRequest( 132 | name="nod", priority=3, repeat=True 133 | ) 134 | ) 135 | except BaseException: 136 | pass 137 | self.start_time = time.time() 138 | if self.is_start: 139 | self._buff.put(in_data) 140 | if time.time() - self.start_time >= self.timeout_thresh: 141 | self.is_start = False 142 | self.closed = True 143 | try: 144 | self.voice_stub.EnableVoicePlay( 145 | voice_server_pb2.EnableVoicePlayRequest() 146 | ) 147 | except BaseException: 148 | print("EnableVoicePlay error") 149 | pass 150 | return None, pyaudio.paComplete 151 | return None, pyaudio.paContinue 152 | 153 | 154 | class GoogleSpeechGrpc(object): 155 | """ 156 | Google Speech-to-Text APIのレスポンスを処理するクラス。 157 | 158 | """ 159 | 160 | def __init__( 161 | self, 162 | gpt_host: str = "127.0.0.1", 163 | gpt_port: str = "10001", 164 | voice_host: str = "127.0.0.1", 165 | voice_port: str = "10002", 166 | ) -> None: 167 | """GoogleSpeechGrpcオブジェクトを初期化する。 168 | 169 | Args: 170 | gpt_host (str, optional): GPTサーバーのホスト名。デフォルトは"127.0.0.1"。 171 | gpt_port (str, optional): GPTサーバーのポート番号。デフォルトは"10001"。 172 | voice_host (str, optional): VoiceVoxサーバーのホスト名。デフォルトは"127.0.0.1"。 173 | voice_port (str, optional): VoiceVoxサーバーのポート番号。デフォルトは"10002"。 174 | """ 175 | 176 | gpt_channel = grpc.insecure_channel(gpt_host + ":" + gpt_port) 177 | self.gpt_stub = gpt_server_pb2_grpc.GptServerServiceStub(gpt_channel) 178 | voice_channel = grpc.insecure_channel(voice_host + ":" + voice_port) 179 | self.voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 180 | 181 | def listen_publisher_grpc( 182 | self, responses: Any, progress_report_len: int = 0 183 | ) -> str: 184 | """ 185 | Google Cloud Speech-to-Text APIの応答からテキストを取得し、リアルタイムで出力。 186 | 187 | Args: 188 | responses (Any): ストリーミング認識の応答 189 | progress_report_len (int, optional): ここで指定した文字数以上になると、その時点で一度GPTに結果を送信する。0の場合は途中での送信は無効となる。デフォルトは0。 190 | 191 | Returns: 192 | str: 認識されたテキスト 193 | """ 194 | is_progress_report = False 195 | num_chars_printed = 0 196 | transcript = "" 197 | overwrite_chars = "" 198 | try: 199 | self.voice_stub.DisableVoicePlay(voice_server_pb2.DisableVoicePlayRequest()) 200 | except BaseException: 201 | print("Disable voice play error") 202 | pass 203 | try: 204 | self.voice_stub.InterruptVoice(voice_server_pb2.InterruptVoiceRequest()) 205 | except BaseException: 206 | print("InterruptVoice error") 207 | pass 208 | for response in responses: 209 | if response.error.code: 210 | break 211 | if not response.results: 212 | continue 213 | result = response.results[0] 214 | if not result.alternatives: 215 | continue 216 | transcript = result.alternatives[0].transcript 217 | overwrite_chars = " " * (num_chars_printed - len(transcript)) 218 | if not result.is_final: 219 | sys.stdout.write(transcript + overwrite_chars + "\r") 220 | sys.stdout.flush() 221 | num_chars_printed = len(transcript) 222 | if not is_progress_report and num_chars_printed > progress_report_len: 223 | if progress_report_len > 0: 224 | try: 225 | self.gpt_stub.SetGpt( 226 | gpt_server_pb2.SetGptRequest( 227 | text=transcript + overwrite_chars, is_finish=False 228 | ) 229 | ) 230 | except BaseException as e: 231 | print("SetGpt error:", e) 232 | pass 233 | is_progress_report = True 234 | else: 235 | if progress_report_len > 0: 236 | if ( 237 | not is_progress_report 238 | and num_chars_printed > progress_report_len 239 | ): 240 | try: 241 | self.gpt_stub.SetGpt( 242 | gpt_server_pb2.SetGptRequest( 243 | text=transcript + overwrite_chars, is_finish=False 244 | ) 245 | ) 246 | except BaseException as e: 247 | print("SetGpt error:", e) 248 | pass 249 | break 250 | try: 251 | self.gpt_stub.SetGpt( 252 | gpt_server_pb2.SetGptRequest( 253 | text=transcript + overwrite_chars, is_finish=True 254 | ) 255 | ) 256 | except BaseException as e: 257 | print("SetGpt error:", e) 258 | pass 259 | return transcript + overwrite_chars 260 | -------------------------------------------------------------------------------- /lib/google_speech_v2.py: -------------------------------------------------------------------------------- 1 | from __future__ import division 2 | 3 | import math 4 | import sys 5 | import time 6 | from queue import Queue 7 | from typing import Iterable, Optional, Union 8 | 9 | import numpy as np 10 | import pyaudio 11 | 12 | # from google.cloud import speech 13 | from google.cloud.speech_v2 import SpeechClient 14 | from google.cloud.speech_v2.types import cloud_speech as cloud_speech_types 15 | from six.moves import queue # type: ignore 16 | 17 | from .conf import GOOGLE_SPEECH_PROJECT_ID 18 | from .err_handler import ignoreStderr 19 | from .google_speech import MicrophoneStream 20 | 21 | # Audio recording parameters 22 | RATE = 16000 23 | CHUNK = int(RATE / 10) # 100ms 24 | 25 | 26 | class MicrophoneStreamV2(MicrophoneStream): 27 | """ 28 | マイクから音声をストリーミングするためのクラス。google STT v2用。 29 | 30 | """ 31 | 32 | def __init__( 33 | self, 34 | rate: float, 35 | chunk: float, 36 | _timeout_thresh: float = 0.5, 37 | _start_timeout_thresh: float = 4.0, 38 | _db_thresh: float = 55.0, 39 | ) -> None: 40 | """クラスの初期化メソッド。 41 | 42 | Args: 43 | rate (float): サンプリングレート。 44 | chunk (float): チャンクサイズ。 45 | _timeout_thresh (float): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 46 | _start_timeout_thresh (float): マイクの入力が開始しないまま終了するまでのタイムアウト閾値（秒）。デフォルトは4.0秒。 47 | _db_thresh (float): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 48 | 49 | """ 50 | self._rate = rate 51 | self._chunk = chunk 52 | self._buff: Queue[Union[None, bytes]] = queue.Queue() 53 | self.closed = True 54 | self.is_start = False 55 | self.is_start_callback = False 56 | self.is_finish = False 57 | self.timeout_thresh = _timeout_thresh 58 | # マイクの入力が開始しないまま終了するまでのthreshold時間[s] 59 | self.start_timeout_thresh = _start_timeout_thresh 60 | self.db_thresh = _db_thresh 61 | language_codes = ["ja-JP"] # a BCP-47 language tag 62 | self.client = SpeechClient() 63 | recognition_config = cloud_speech_types.RecognitionConfig( 64 | explicit_decoding_config=cloud_speech_types.ExplicitDecodingConfig( 65 | sample_rate_hertz=RATE, 66 | encoding=cloud_speech_types.ExplicitDecodingConfig.AudioEncoding.LINEAR16, 67 | audio_channel_count=1, 68 | ), 69 | language_codes=language_codes, 70 | model="long", 71 | ) 72 | streaming_config = cloud_speech_types.StreamingRecognitionConfig( 73 | config=recognition_config, 74 | streaming_features=cloud_speech_types.StreamingRecognitionFeatures( 75 | interim_results=True 76 | ), 77 | ) 78 | if GOOGLE_SPEECH_PROJECT_ID == "": 79 | raise ValueError("GOOGLE_SPEECH_PROJECT_ID is not set.") 80 | self.config_request = cloud_speech_types.StreamingRecognizeRequest( 81 | recognizer=f"projects/{GOOGLE_SPEECH_PROJECT_ID}/locations/global/recognizers/_", 82 | streaming_config=streaming_config, 83 | ) 84 | 85 | def requests( 86 | self, config: cloud_speech_types.RecognitionConfig, audio: list 87 | ) -> list: 88 | yield config 89 | for chunk in audio: 90 | yield cloud_speech_types.StreamingRecognizeRequest(audio=chunk) 91 | 92 | def transcribe( 93 | self, 94 | ) -> Optional[Iterable[cloud_speech_types.StreamingRecognizeResponse]]: 95 | """ストリームからの音声をGoogle Cloud Speech-to-Text APIでテキストに変換する。 96 | 97 | Returns: 98 | Optional[Iterable[speech.StreamingRecognizeResponse]]: ストリーミング認識の応答 99 | """ 100 | audio_generator = self.generator() 101 | self.start_time = time.time() 102 | self.start_callback() 103 | responses = None 104 | try: 105 | responses = self.client.streaming_recognize( 106 | requests=self.requests(self.config_request, audio_generator) 107 | ) 108 | except BaseException: 109 | pass 110 | return responses 111 | 112 | 113 | def get_db_thresh() -> float: 114 | """マイクからの周囲音量を測定。 115 | 116 | Returns: 117 | float: 測定された音量[db] 118 | """ 119 | with ignoreStderr(): 120 | p = pyaudio.PyAudio() 121 | stream = p.open( 122 | format=pyaudio.paInt16, 123 | channels=1, 124 | rate=RATE, 125 | input=True, 126 | frames_per_buffer=CHUNK, 127 | ) 128 | frames = [] 129 | print("Measuring Ambient Sound Levels…") 130 | for _ in range(int(RATE / CHUNK * 2)): 131 | data = stream.read(CHUNK) 132 | frames.append(data) 133 | audio_data = np.frombuffer(b"".join(frames), dtype=np.int16) 134 | rms2 = np.square(audio_data).mean() 135 | if rms2 > 0.0: 136 | rms = math.sqrt(np.square(audio_data).mean()) 137 | power = 20 * math.log10(rms) if rms > 0.0 else -math.inf # RMS to db 138 | else: 139 | power = 20 140 | print(f"Sound Levels: {power:.3f}db") 141 | stream.stop_stream() 142 | stream.close() 143 | p.terminate() 144 | return power 145 | 146 | 147 | def listen_print_loop(responses: object) -> str: 148 | """Google Cloud Speech-to-Text APIの応答からテキストを取得し、リアルタイムで出力。 149 | 150 | Args: 151 | responses (Any): ストリーミング認識の応答 152 | 153 | Returns: 154 | str: 認識されたテキスト 155 | 156 | """ 157 | num_chars_printed = 0 158 | transcript = "" 159 | overwrite_chars = "" 160 | for response in responses: 161 | # if response.error.code: 162 | # break 163 | if not response.results: 164 | continue 165 | result = response.results[0] 166 | if not result.alternatives: 167 | continue 168 | transcript = result.alternatives[0].transcript 169 | overwrite_chars = " " * (num_chars_printed - len(transcript)) 170 | if not result.is_final: 171 | sys.stdout.write(transcript + overwrite_chars + "\r") 172 | sys.stdout.flush() 173 | num_chars_printed = len(transcript) 174 | else: 175 | print(transcript + overwrite_chars) 176 | break 177 | return transcript + overwrite_chars 178 | -------------------------------------------------------------------------------- /lib/google_speech_v2_grpc.py: -------------------------------------------------------------------------------- 1 | from __future__ import division 2 | 3 | import math 4 | import os 5 | import struct 6 | import sys 7 | import time 8 | from typing import Any, Optional, Union 9 | 10 | import grpc 11 | import numpy as np 12 | import pyaudio 13 | 14 | from .google_speech_v2 import MicrophoneStreamV2 15 | 16 | sys.path.append(os.path.join(os.path.dirname(__file__), "grpc")) 17 | import gpt_server_pb2 18 | import gpt_server_pb2_grpc 19 | import motion_server_pb2 20 | import motion_server_pb2_grpc 21 | import voice_server_pb2 22 | import voice_server_pb2_grpc 23 | 24 | 25 | class MicrophoneStreamV2Grpc(MicrophoneStreamV2): 26 | """ 27 | マイクから音声をストリーミングするためのクラス。 28 | 29 | """ 30 | 31 | def __init__( 32 | self, 33 | rate: float, 34 | chunk: float, 35 | _timeout_thresh: float = 0.5, 36 | _start_timeout_thresh: float = 4.0, 37 | _db_thresh: float = 55.0, 38 | gpt_host: str = "127.0.0.1", 39 | gpt_port: str = "10001", 40 | voice_host: str = "127.0.0.1", 41 | voice_port: str = "10002", 42 | motion_server_host: Optional[str] = "127.0.0.1", 43 | motion_server_port: Optional[str] = "50055", 44 | ) -> None: 45 | """クラスの初期化メソッド。 46 | 47 | Args: 48 | rate (float): サンプリングレート。 49 | chunk (float): チャンクサイズ。 50 | _timeout_thresh (float): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 51 | _db_thresh (float): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 52 | gpt_host (str, optional): GPTサーバーのホスト名。デフォルトは"127.0.0.1"。 53 | gpt_port (str, optional): GPTサーバーのポート番号。デフォルトは"10001"。 54 | voice_host (str, optional): VoiceVoxサーバーのホスト名。デフォルトは"127.0.0.1"。 55 | voice_port (str, optional): VoiceVoxサーバーのポート番号。デフォルトは"10002"。 56 | motion_server_host (str, optional): モーションサーバーのIPアドレス。デフォルトは"127.0.0.1"。 57 | motion_server_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 58 | """ 59 | super().__init__( 60 | rate=rate, 61 | chunk=chunk, 62 | _timeout_thresh=_timeout_thresh, 63 | _start_timeout_thresh=_start_timeout_thresh, 64 | _db_thresh=_db_thresh, 65 | ) 66 | gpt_channel = grpc.insecure_channel(gpt_host + ":" + gpt_port) 67 | self.gpt_stub = gpt_server_pb2_grpc.GptServerServiceStub(gpt_channel) 68 | voice_channel = grpc.insecure_channel(voice_host + ":" + voice_port) 69 | self.voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 70 | self.motion_stub = None 71 | if motion_server_host is not None and motion_server_port is not None: 72 | motion_channel = grpc.insecure_channel( 73 | motion_server_host + ":" + motion_server_port 74 | ) 75 | self.motion_stub = motion_server_pb2_grpc.MotionServerServiceStub( 76 | motion_channel 77 | ) 78 | 79 | def __exit__( 80 | self, 81 | rate: float, 82 | chunk: float, 83 | _timeout_thresh: float = 0.5, 84 | _start_timeout_thresh: float = 4.0, 85 | _db_thresh: float = 55.0, 86 | ) -> None: 87 | """PyAudioストリームを閉じます。 88 | 89 | Args: 90 | rate (float): サンプリングレート。 91 | chunk (float): チャンクサイズ。 92 | _timeout_thresh (float, optional): 音声が停止したと判断するタイムアウト閾値（秒）。デフォルトは0.5秒。 93 | _start_timeout_thresh (float): マイクの入力が開始しないまま終了するまでのタイムアウト閾値（秒）。デフォルトは4.0秒。 94 | _db_thresh (float, optional): 音声が開始されたと判断する音量閾値（デシベル）。デフォルトは55.0デシベル。 95 | 96 | """ 97 | super().__exit__( 98 | rate, chunk, _timeout_thresh, _start_timeout_thresh, _db_thresh 99 | ) 100 | try: 101 | self.gpt_stub.SendMotion(gpt_server_pb2.SendMotionRequest()) 102 | except BaseException: 103 | print("Send motion error") 104 | pass 105 | 106 | def _fill_buffer( 107 | self, in_data: bytes, frame_count: int, time_info: Any, status_flags: Any 108 | ) -> Union[None, Any]: 109 | """マイクからの入力データをバッファーに書き込む。 110 | 111 | Args: 112 | in_data (bytes): 入力データ 113 | frame_count (int): フレーム数 114 | time_info (Any): 時間 115 | status_flags (Any): ステータスフラグ 116 | 117 | Returns: 118 | Union[None, Any]: Noneまたは続行のためのフラグ 119 | 120 | """ 121 | if self.is_start_callback: 122 | in_data2 = struct.unpack(f"{len(in_data) / 2:.0f}h", in_data) 123 | rms = math.sqrt(np.square(in_data2).mean()) 124 | power = 20 * math.log10(rms) if rms > 0.0 else -math.inf # RMS to db 125 | if power > self.db_thresh: 126 | if not self.is_start: 127 | self.is_start = True 128 | if self.motion_stub is not None: 129 | try: 130 | self.motion_stub.SetMotion( 131 | motion_server_pb2.SetMotionRequest( 132 | name="nod", priority=3, repeat=True 133 | ) 134 | ) 135 | except BaseException: 136 | pass 137 | self.start_time = time.time() 138 | if self.is_start: 139 | self._buff.put(in_data) 140 | if time.time() - self.start_time >= self.timeout_thresh: 141 | self.is_start = False 142 | self.closed = True 143 | try: 144 | self.voice_stub.EnableVoicePlay( 145 | voice_server_pb2.EnableVoicePlayRequest() 146 | ) 147 | except BaseException: 148 | print("EnableVoicePlay error") 149 | pass 150 | return None, pyaudio.paComplete 151 | return None, pyaudio.paContinue 152 | 153 | 154 | class GoogleSpeechV2Grpc(object): 155 | """ 156 | Google Speech-to-Text APIのレスポンスを処理するクラス。 157 | 158 | """ 159 | 160 | def __init__( 161 | self, 162 | gpt_host: str = "127.0.0.1", 163 | gpt_port: str = "10001", 164 | voice_host: str = "127.0.0.1", 165 | voice_port: str = "10002", 166 | ) -> None: 167 | """GoogleSpeechGrpcオブジェクトを初期化する。 168 | 169 | Args: 170 | gpt_host (str, optional): GPTサーバーのホスト名。デフォルトは"127.0.0.1"。 171 | gpt_port (str, optional): GPTサーバーのポート番号。デフォルトは"10001"。 172 | voice_host (str, optional): VoiceVoxサーバーのホスト名。デフォルトは"127.0.0.1"。 173 | voice_port (str, optional): VoiceVoxサーバーのポート番号。デフォルトは"10002"。 174 | """ 175 | 176 | gpt_channel = grpc.insecure_channel(gpt_host + ":" + gpt_port) 177 | self.gpt_stub = gpt_server_pb2_grpc.GptServerServiceStub(gpt_channel) 178 | voice_channel = grpc.insecure_channel(voice_host + ":" + voice_port) 179 | self.voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 180 | 181 | def listen_publisher_grpc( 182 | self, responses: Any, progress_report_len: int = 0 183 | ) -> str: 184 | """ 185 | Google Cloud Speech-to-Text APIの応答からテキストを取得し、リアルタイムで出力。 186 | 187 | Args: 188 | responses (Any): ストリーミング認識の応答 189 | progress_report_len (int, optional): ここで指定した文字数以上になると、その時点で一度GPTに結果を送信する。0の場合は途中での送信は無効となる。デフォルトは0。 190 | 191 | Returns: 192 | str: 認識されたテキスト 193 | """ 194 | is_progress_report = False 195 | num_chars_printed = 0 196 | transcript = "" 197 | overwrite_chars = "" 198 | try: 199 | self.voice_stub.DisableVoicePlay(voice_server_pb2.DisableVoicePlayRequest()) 200 | except BaseException: 201 | print("DisableVoicePlay error") 202 | pass 203 | try: 204 | self.voice_stub.InterruptVoice(voice_server_pb2.InterruptVoiceRequest()) 205 | except BaseException: 206 | print("InterruptVoice error") 207 | pass 208 | for response in responses: 209 | # if response.error.code: 210 | # break 211 | if not response.results: 212 | continue 213 | result = response.results[0] 214 | if not result.alternatives: 215 | continue 216 | transcript = result.alternatives[0].transcript 217 | overwrite_chars = " " * (num_chars_printed - len(transcript)) 218 | if not result.is_final: 219 | sys.stdout.write(transcript + overwrite_chars + "\r") 220 | sys.stdout.flush() 221 | num_chars_printed = len(transcript) 222 | if not is_progress_report and num_chars_printed > progress_report_len: 223 | if progress_report_len > 0: 224 | try: 225 | self.gpt_stub.SetGpt( 226 | gpt_server_pb2.SetGptRequest( 227 | text=transcript + overwrite_chars, is_finish=False 228 | ) 229 | ) 230 | except BaseException as e: 231 | print("SetGpt error:", e) 232 | pass 233 | is_progress_report = True 234 | else: 235 | if progress_report_len > 0: 236 | if ( 237 | not is_progress_report 238 | and num_chars_printed > progress_report_len 239 | ): 240 | try: 241 | self.gpt_stub.SetGpt( 242 | gpt_server_pb2.SetGptRequest( 243 | text=transcript + overwrite_chars, is_finish=False 244 | ) 245 | ) 246 | except BaseException as e: 247 | print("SetGpt error:", e) 248 | pass 249 | break 250 | try: 251 | self.gpt_stub.SetGpt( 252 | gpt_server_pb2.SetGptRequest( 253 | text=transcript + overwrite_chars, is_finish=True 254 | ) 255 | ) 256 | except BaseException as e: 257 | print("SetGpt error:", e) 258 | pass 259 | return transcript + overwrite_chars 260 | -------------------------------------------------------------------------------- /lib/grpc/gpt_server_pb2.py: -------------------------------------------------------------------------------- 1 | # -*- coding: utf-8 -*- 2 | # Generated by the protocol buffer compiler. DO NOT EDIT! 3 | # source: gpt_server.proto 4 | # Protobuf Python Version: 4.25.0 5 | """Generated protocol buffer code.""" 6 | from google.protobuf import descriptor as _descriptor 7 | from google.protobuf import descriptor_pool as _descriptor_pool 8 | from google.protobuf import symbol_database as _symbol_database 9 | from google.protobuf.internal import builder as _builder 10 | # @@protoc_insertion_point(imports) 11 | 12 | _sym_db = _symbol_database.Default() 13 | 14 | 15 | 16 | 17 | DESCRIPTOR = _descriptor_pool.Default().AddSerializedFile(b'\n\x10gpt_server.proto\x12\ngpt_server\"C\n\rSetGptRequest\x12\x0c\n\x04text\x18\x01 \x01(\t\x12\x16\n\tis_finish\x18\x02 \x01(\x08H\x00\x88\x01\x01\x42\x0c\n\n_is_finish\"\x1e\n\x0bSetGptReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x15\n\x13InterruptGptRequest\"$\n\x11InterruptGptReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x13\n\x11SendMotionRequest\"\"\n\x0fSendMotionReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\x32\xea\x01\n\x10GptServerService\x12<\n\x06SetGpt\x12\x19.gpt_server.SetGptRequest\x1a\x17.gpt_server.SetGptReply\x12N\n\x0cInterruptGpt\x12\x1f.gpt_server.InterruptGptRequest\x1a\x1d.gpt_server.InterruptGptReply\x12H\n\nSendMotion\x12\x1d.gpt_server.SendMotionRequest\x1a\x1b.gpt_server.SendMotionReplyb\x06proto3') 18 | 19 | _globals = globals() 20 | _builder.BuildMessageAndEnumDescriptors(DESCRIPTOR, _globals) 21 | _builder.BuildTopDescriptorsAndMessages(DESCRIPTOR, 'gpt_server_pb2', _globals) 22 | if _descriptor._USE_C_DESCRIPTORS == False: 23 | DESCRIPTOR._options = None 24 | _globals['_SETGPTREQUEST']._serialized_start=32 25 | _globals['_SETGPTREQUEST']._serialized_end=99 26 | _globals['_SETGPTREPLY']._serialized_start=101 27 | _globals['_SETGPTREPLY']._serialized_end=131 28 | _globals['_INTERRUPTGPTREQUEST']._serialized_start=133 29 | _globals['_INTERRUPTGPTREQUEST']._serialized_end=154 30 | _globals['_INTERRUPTGPTREPLY']._serialized_start=156 31 | _globals['_INTERRUPTGPTREPLY']._serialized_end=192 32 | _globals['_SENDMOTIONREQUEST']._serialized_start=194 33 | _globals['_SENDMOTIONREQUEST']._serialized_end=213 34 | _globals['_SENDMOTIONREPLY']._serialized_start=215 35 | _globals['_SENDMOTIONREPLY']._serialized_end=249 36 | _globals['_GPTSERVERSERVICE']._serialized_start=252 37 | _globals['_GPTSERVERSERVICE']._serialized_end=486 38 | # @@protoc_insertion_point(module_scope) 39 | -------------------------------------------------------------------------------- /lib/grpc/gpt_server_pb2_grpc.py: -------------------------------------------------------------------------------- 1 | # Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT! 2 | """Client and server classes corresponding to protobuf-defined services.""" 3 | import grpc 4 | 5 | import gpt_server_pb2 as gpt__server__pb2 6 | 7 | 8 | class GptServerServiceStub(object): 9 | """Missing associated documentation comment in .proto file.""" 10 | 11 | def __init__(self, channel): 12 | """Constructor. 13 | 14 | Args: 15 | channel: A grpc.Channel. 16 | """ 17 | self.SetGpt = channel.unary_unary( 18 | '/gpt_server.GptServerService/SetGpt', 19 | request_serializer=gpt__server__pb2.SetGptRequest.SerializeToString, 20 | response_deserializer=gpt__server__pb2.SetGptReply.FromString, 21 | ) 22 | self.InterruptGpt = channel.unary_unary( 23 | '/gpt_server.GptServerService/InterruptGpt', 24 | request_serializer=gpt__server__pb2.InterruptGptRequest.SerializeToString, 25 | response_deserializer=gpt__server__pb2.InterruptGptReply.FromString, 26 | ) 27 | self.SendMotion = channel.unary_unary( 28 | '/gpt_server.GptServerService/SendMotion', 29 | request_serializer=gpt__server__pb2.SendMotionRequest.SerializeToString, 30 | response_deserializer=gpt__server__pb2.SendMotionReply.FromString, 31 | ) 32 | 33 | 34 | class GptServerServiceServicer(object): 35 | """Missing associated documentation comment in .proto file.""" 36 | 37 | def SetGpt(self, request, context): 38 | """Missing associated documentation comment in .proto file.""" 39 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 40 | context.set_details('Method not implemented!') 41 | raise NotImplementedError('Method not implemented!') 42 | 43 | def InterruptGpt(self, request, context): 44 | """Missing associated documentation comment in .proto file.""" 45 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 46 | context.set_details('Method not implemented!') 47 | raise NotImplementedError('Method not implemented!') 48 | 49 | def SendMotion(self, request, context): 50 | """Missing associated documentation comment in .proto file.""" 51 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 52 | context.set_details('Method not implemented!') 53 | raise NotImplementedError('Method not implemented!') 54 | 55 | 56 | def add_GptServerServiceServicer_to_server(servicer, server): 57 | rpc_method_handlers = { 58 | 'SetGpt': grpc.unary_unary_rpc_method_handler( 59 | servicer.SetGpt, 60 | request_deserializer=gpt__server__pb2.SetGptRequest.FromString, 61 | response_serializer=gpt__server__pb2.SetGptReply.SerializeToString, 62 | ), 63 | 'InterruptGpt': grpc.unary_unary_rpc_method_handler( 64 | servicer.InterruptGpt, 65 | request_deserializer=gpt__server__pb2.InterruptGptRequest.FromString, 66 | response_serializer=gpt__server__pb2.InterruptGptReply.SerializeToString, 67 | ), 68 | 'SendMotion': grpc.unary_unary_rpc_method_handler( 69 | servicer.SendMotion, 70 | request_deserializer=gpt__server__pb2.SendMotionRequest.FromString, 71 | response_serializer=gpt__server__pb2.SendMotionReply.SerializeToString, 72 | ), 73 | } 74 | generic_handler = grpc.method_handlers_generic_handler( 75 | 'gpt_server.GptServerService', rpc_method_handlers) 76 | server.add_generic_rpc_handlers((generic_handler,)) 77 | 78 | 79 | # This class is part of an EXPERIMENTAL API. 80 | class GptServerService(object): 81 | """Missing associated documentation comment in .proto file.""" 82 | 83 | @staticmethod 84 | def SetGpt(request, 85 | target, 86 | options=(), 87 | channel_credentials=None, 88 | call_credentials=None, 89 | insecure=False, 90 | compression=None, 91 | wait_for_ready=None, 92 | timeout=None, 93 | metadata=None): 94 | return grpc.experimental.unary_unary(request, target, '/gpt_server.GptServerService/SetGpt', 95 | gpt__server__pb2.SetGptRequest.SerializeToString, 96 | gpt__server__pb2.SetGptReply.FromString, 97 | options, channel_credentials, 98 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 99 | 100 | @staticmethod 101 | def InterruptGpt(request, 102 | target, 103 | options=(), 104 | channel_credentials=None, 105 | call_credentials=None, 106 | insecure=False, 107 | compression=None, 108 | wait_for_ready=None, 109 | timeout=None, 110 | metadata=None): 111 | return grpc.experimental.unary_unary(request, target, '/gpt_server.GptServerService/InterruptGpt', 112 | gpt__server__pb2.InterruptGptRequest.SerializeToString, 113 | gpt__server__pb2.InterruptGptReply.FromString, 114 | options, channel_credentials, 115 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 116 | 117 | @staticmethod 118 | def SendMotion(request, 119 | target, 120 | options=(), 121 | channel_credentials=None, 122 | call_credentials=None, 123 | insecure=False, 124 | compression=None, 125 | wait_for_ready=None, 126 | timeout=None, 127 | metadata=None): 128 | return grpc.experimental.unary_unary(request, target, '/gpt_server.GptServerService/SendMotion', 129 | gpt__server__pb2.SendMotionRequest.SerializeToString, 130 | gpt__server__pb2.SendMotionReply.FromString, 131 | options, channel_credentials, 132 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 133 | -------------------------------------------------------------------------------- /lib/grpc/motion_server_pb2.py: -------------------------------------------------------------------------------- 1 | # -*- coding: utf-8 -*- 2 | # Generated by the protocol buffer compiler. DO NOT EDIT! 3 | # source: motion_server.proto 4 | # Protobuf Python Version: 4.25.0 5 | """Generated protocol buffer code.""" 6 | from google.protobuf import descriptor as _descriptor 7 | from google.protobuf import descriptor_pool as _descriptor_pool 8 | from google.protobuf import symbol_database as _symbol_database 9 | from google.protobuf.internal import builder as _builder 10 | # @@protoc_insertion_point(imports) 11 | 12 | _sym_db = _symbol_database.Default() 13 | 14 | 15 | 16 | 17 | DESCRIPTOR = _descriptor_pool.Default().AddSerializedFile(b'\n\x13motion_server.proto\x12\rmotion_server\"\x82\x01\n\x10SetMotionRequest\x12\x0c\n\x04name\x18\x01 \x01(\t\x12\x15\n\x08priority\x18\x02 \x01(\x05H\x00\x88\x01\x01\x12\x13\n\x06repeat\x18\x03 \x01(\x08H\x01\x88\x01\x01\x12\x12\n\x05\x63lear\x18\x04 \x01(\x08H\x02\x88\x01\x01\x42\x0b\n\t_priorityB\t\n\x07_repeatB\x08\n\x06_clear\"!\n\x0eSetMotionReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"8\n\x12\x43learMotionRequest\x12\x15\n\x08priority\x18\x01 \x01(\x05H\x00\x88\x01\x01\x42\x0b\n\t_priority\"#\n\x10\x43learMotionReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"7\n\x11StopRepeatRequest\x12\x15\n\x08priority\x18\x01 \x01(\x05H\x00\x88\x01\x01\x42\x0b\n\t_priority\"\"\n\x0fStopRepeatReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"`\n\x0eSetWaitRequest\x12\x0c\n\x04time\x18\x01 \x01(\x02\x12\x15\n\x08priority\x18\x02 \x01(\x05H\x00\x88\x01\x01\x12\x12\n\x05\x63lear\x18\x03 \x01(\x08H\x01\x88\x01\x01\x42\x0b\n\t_priorityB\x08\n\x06_clear\"\x1f\n\x0cSetWaitReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"i\n\rSetPosRequest\x12\x10\n\x03pan\x18\x01 \x01(\x02H\x00\x88\x01\x01\x12\x11\n\x04tilt\x18\x02 \x01(\x02H\x01\x88\x01\x01\x12\x15\n\x08priority\x18\x03 \x01(\x05H\x02\x88\x01\x01\x42\x06\n\x04_panB\x07\n\x05_tiltB\x0b\n\t_priority\"\x1e\n\x0bSetPosReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"i\n\rSetVelRequest\x12\x10\n\x03pan\x18\x01 \x01(\x02H\x00\x88\x01\x01\x12\x11\n\x04tilt\x18\x02 \x01(\x02H\x01\x88\x01\x01\x12\x15\n\x08priority\x18\x03 \x01(\x05H\x02\x88\x01\x01\x42\x06\n\x04_panB\x07\n\x05_tiltB\x0b\n\t_priority\"\x1e\n\x0bSetVelReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"i\n\rSetAccRequest\x12\x10\n\x03pan\x18\x01 \x01(\x02H\x00\x88\x01\x01\x12\x11\n\x04tilt\x18\x02 \x01(\x02H\x01\x88\x01\x01\x12\x15\n\x08priority\x18\x03 \x01(\x05H\x02\x88\x01\x01\x42\x06\n\x04_panB\x07\n\x05_tiltB\x0b\n\t_priority\"\x1e\n\x0bSetAccReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\x32\xa6\x04\n\x13MotionServerService\x12M\n\tSetMotion\x12\x1f.motion_server.SetMotionRequest\x1a\x1d.motion_server.SetMotionReply\"\x00\x12S\n\x0b\x43learMotion\x12!.motion_server.ClearMotionRequest\x1a\x1f.motion_server.ClearMotionReply\"\x00\x12P\n\nStopRepeat\x12 .motion_server.StopRepeatRequest\x1a\x1e.motion_server.StopRepeatReply\"\x00\x12G\n\x07SetWait\x12\x1d.motion_server.SetWaitRequest\x1a\x1b.motion_server.SetWaitReply\"\x00\x12\x44\n\x06SetPos\x12\x1c.motion_server.SetPosRequest\x1a\x1a.motion_server.SetPosReply\"\x00\x12\x44\n\x06SetVel\x12\x1c.motion_server.SetVelRequest\x1a\x1a.motion_server.SetVelReply\"\x00\x12\x44\n\x06SetAcc\x12\x1c.motion_server.SetAccRequest\x1a\x1a.motion_server.SetAccReply\"\x00\x62\x06proto3') 18 | 19 | _globals = globals() 20 | _builder.BuildMessageAndEnumDescriptors(DESCRIPTOR, _globals) 21 | _builder.BuildTopDescriptorsAndMessages(DESCRIPTOR, 'motion_server_pb2', _globals) 22 | if _descriptor._USE_C_DESCRIPTORS == False: 23 | DESCRIPTOR._options = None 24 | _globals['_SETMOTIONREQUEST']._serialized_start=39 25 | _globals['_SETMOTIONREQUEST']._serialized_end=169 26 | _globals['_SETMOTIONREPLY']._serialized_start=171 27 | _globals['_SETMOTIONREPLY']._serialized_end=204 28 | _globals['_CLEARMOTIONREQUEST']._serialized_start=206 29 | _globals['_CLEARMOTIONREQUEST']._serialized_end=262 30 | _globals['_CLEARMOTIONREPLY']._serialized_start=264 31 | _globals['_CLEARMOTIONREPLY']._serialized_end=299 32 | _globals['_STOPREPEATREQUEST']._serialized_start=301 33 | _globals['_STOPREPEATREQUEST']._serialized_end=356 34 | _globals['_STOPREPEATREPLY']._serialized_start=358 35 | _globals['_STOPREPEATREPLY']._serialized_end=392 36 | _globals['_SETWAITREQUEST']._serialized_start=394 37 | _globals['_SETWAITREQUEST']._serialized_end=490 38 | _globals['_SETWAITREPLY']._serialized_start=492 39 | _globals['_SETWAITREPLY']._serialized_end=523 40 | _globals['_SETPOSREQUEST']._serialized_start=525 41 | _globals['_SETPOSREQUEST']._serialized_end=630 42 | _globals['_SETPOSREPLY']._serialized_start=632 43 | _globals['_SETPOSREPLY']._serialized_end=662 44 | _globals['_SETVELREQUEST']._serialized_start=664 45 | _globals['_SETVELREQUEST']._serialized_end=769 46 | _globals['_SETVELREPLY']._serialized_start=771 47 | _globals['_SETVELREPLY']._serialized_end=801 48 | _globals['_SETACCREQUEST']._serialized_start=803 49 | _globals['_SETACCREQUEST']._serialized_end=908 50 | _globals['_SETACCREPLY']._serialized_start=910 51 | _globals['_SETACCREPLY']._serialized_end=940 52 | _globals['_MOTIONSERVERSERVICE']._serialized_start=943 53 | _globals['_MOTIONSERVERSERVICE']._serialized_end=1493 54 | # @@protoc_insertion_point(module_scope) 55 | -------------------------------------------------------------------------------- /lib/grpc/motion_server_pb2_grpc.py: -------------------------------------------------------------------------------- 1 | # Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT! 2 | """Client and server classes corresponding to protobuf-defined services.""" 3 | import grpc 4 | 5 | import motion_server_pb2 as motion__server__pb2 6 | 7 | 8 | class MotionServerServiceStub(object): 9 | """Missing associated documentation comment in .proto file.""" 10 | 11 | def __init__(self, channel): 12 | """Constructor. 13 | 14 | Args: 15 | channel: A grpc.Channel. 16 | """ 17 | self.SetMotion = channel.unary_unary( 18 | '/motion_server.MotionServerService/SetMotion', 19 | request_serializer=motion__server__pb2.SetMotionRequest.SerializeToString, 20 | response_deserializer=motion__server__pb2.SetMotionReply.FromString, 21 | ) 22 | self.ClearMotion = channel.unary_unary( 23 | '/motion_server.MotionServerService/ClearMotion', 24 | request_serializer=motion__server__pb2.ClearMotionRequest.SerializeToString, 25 | response_deserializer=motion__server__pb2.ClearMotionReply.FromString, 26 | ) 27 | self.StopRepeat = channel.unary_unary( 28 | '/motion_server.MotionServerService/StopRepeat', 29 | request_serializer=motion__server__pb2.StopRepeatRequest.SerializeToString, 30 | response_deserializer=motion__server__pb2.StopRepeatReply.FromString, 31 | ) 32 | self.SetWait = channel.unary_unary( 33 | '/motion_server.MotionServerService/SetWait', 34 | request_serializer=motion__server__pb2.SetWaitRequest.SerializeToString, 35 | response_deserializer=motion__server__pb2.SetWaitReply.FromString, 36 | ) 37 | self.SetPos = channel.unary_unary( 38 | '/motion_server.MotionServerService/SetPos', 39 | request_serializer=motion__server__pb2.SetPosRequest.SerializeToString, 40 | response_deserializer=motion__server__pb2.SetPosReply.FromString, 41 | ) 42 | self.SetVel = channel.unary_unary( 43 | '/motion_server.MotionServerService/SetVel', 44 | request_serializer=motion__server__pb2.SetVelRequest.SerializeToString, 45 | response_deserializer=motion__server__pb2.SetVelReply.FromString, 46 | ) 47 | self.SetAcc = channel.unary_unary( 48 | '/motion_server.MotionServerService/SetAcc', 49 | request_serializer=motion__server__pb2.SetAccRequest.SerializeToString, 50 | response_deserializer=motion__server__pb2.SetAccReply.FromString, 51 | ) 52 | 53 | 54 | class MotionServerServiceServicer(object): 55 | """Missing associated documentation comment in .proto file.""" 56 | 57 | def SetMotion(self, request, context): 58 | """Missing associated documentation comment in .proto file.""" 59 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 60 | context.set_details('Method not implemented!') 61 | raise NotImplementedError('Method not implemented!') 62 | 63 | def ClearMotion(self, request, context): 64 | """Missing associated documentation comment in .proto file.""" 65 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 66 | context.set_details('Method not implemented!') 67 | raise NotImplementedError('Method not implemented!') 68 | 69 | def StopRepeat(self, request, context): 70 | """Missing associated documentation comment in .proto file.""" 71 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 72 | context.set_details('Method not implemented!') 73 | raise NotImplementedError('Method not implemented!') 74 | 75 | def SetWait(self, request, context): 76 | """Missing associated documentation comment in .proto file.""" 77 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 78 | context.set_details('Method not implemented!') 79 | raise NotImplementedError('Method not implemented!') 80 | 81 | def SetPos(self, request, context): 82 | """Missing associated documentation comment in .proto file.""" 83 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 84 | context.set_details('Method not implemented!') 85 | raise NotImplementedError('Method not implemented!') 86 | 87 | def SetVel(self, request, context): 88 | """Missing associated documentation comment in .proto file.""" 89 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 90 | context.set_details('Method not implemented!') 91 | raise NotImplementedError('Method not implemented!') 92 | 93 | def SetAcc(self, request, context): 94 | """Missing associated documentation comment in .proto file.""" 95 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 96 | context.set_details('Method not implemented!') 97 | raise NotImplementedError('Method not implemented!') 98 | 99 | 100 | def add_MotionServerServiceServicer_to_server(servicer, server): 101 | rpc_method_handlers = { 102 | 'SetMotion': grpc.unary_unary_rpc_method_handler( 103 | servicer.SetMotion, 104 | request_deserializer=motion__server__pb2.SetMotionRequest.FromString, 105 | response_serializer=motion__server__pb2.SetMotionReply.SerializeToString, 106 | ), 107 | 'ClearMotion': grpc.unary_unary_rpc_method_handler( 108 | servicer.ClearMotion, 109 | request_deserializer=motion__server__pb2.ClearMotionRequest.FromString, 110 | response_serializer=motion__server__pb2.ClearMotionReply.SerializeToString, 111 | ), 112 | 'StopRepeat': grpc.unary_unary_rpc_method_handler( 113 | servicer.StopRepeat, 114 | request_deserializer=motion__server__pb2.StopRepeatRequest.FromString, 115 | response_serializer=motion__server__pb2.StopRepeatReply.SerializeToString, 116 | ), 117 | 'SetWait': grpc.unary_unary_rpc_method_handler( 118 | servicer.SetWait, 119 | request_deserializer=motion__server__pb2.SetWaitRequest.FromString, 120 | response_serializer=motion__server__pb2.SetWaitReply.SerializeToString, 121 | ), 122 | 'SetPos': grpc.unary_unary_rpc_method_handler( 123 | servicer.SetPos, 124 | request_deserializer=motion__server__pb2.SetPosRequest.FromString, 125 | response_serializer=motion__server__pb2.SetPosReply.SerializeToString, 126 | ), 127 | 'SetVel': grpc.unary_unary_rpc_method_handler( 128 | servicer.SetVel, 129 | request_deserializer=motion__server__pb2.SetVelRequest.FromString, 130 | response_serializer=motion__server__pb2.SetVelReply.SerializeToString, 131 | ), 132 | 'SetAcc': grpc.unary_unary_rpc_method_handler( 133 | servicer.SetAcc, 134 | request_deserializer=motion__server__pb2.SetAccRequest.FromString, 135 | response_serializer=motion__server__pb2.SetAccReply.SerializeToString, 136 | ), 137 | } 138 | generic_handler = grpc.method_handlers_generic_handler( 139 | 'motion_server.MotionServerService', rpc_method_handlers) 140 | server.add_generic_rpc_handlers((generic_handler,)) 141 | 142 | 143 | # This class is part of an EXPERIMENTAL API. 144 | class MotionServerService(object): 145 | """Missing associated documentation comment in .proto file.""" 146 | 147 | @staticmethod 148 | def SetMotion(request, 149 | target, 150 | options=(), 151 | channel_credentials=None, 152 | call_credentials=None, 153 | insecure=False, 154 | compression=None, 155 | wait_for_ready=None, 156 | timeout=None, 157 | metadata=None): 158 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/SetMotion', 159 | motion__server__pb2.SetMotionRequest.SerializeToString, 160 | motion__server__pb2.SetMotionReply.FromString, 161 | options, channel_credentials, 162 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 163 | 164 | @staticmethod 165 | def ClearMotion(request, 166 | target, 167 | options=(), 168 | channel_credentials=None, 169 | call_credentials=None, 170 | insecure=False, 171 | compression=None, 172 | wait_for_ready=None, 173 | timeout=None, 174 | metadata=None): 175 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/ClearMotion', 176 | motion__server__pb2.ClearMotionRequest.SerializeToString, 177 | motion__server__pb2.ClearMotionReply.FromString, 178 | options, channel_credentials, 179 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 180 | 181 | @staticmethod 182 | def StopRepeat(request, 183 | target, 184 | options=(), 185 | channel_credentials=None, 186 | call_credentials=None, 187 | insecure=False, 188 | compression=None, 189 | wait_for_ready=None, 190 | timeout=None, 191 | metadata=None): 192 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/StopRepeat', 193 | motion__server__pb2.StopRepeatRequest.SerializeToString, 194 | motion__server__pb2.StopRepeatReply.FromString, 195 | options, channel_credentials, 196 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 197 | 198 | @staticmethod 199 | def SetWait(request, 200 | target, 201 | options=(), 202 | channel_credentials=None, 203 | call_credentials=None, 204 | insecure=False, 205 | compression=None, 206 | wait_for_ready=None, 207 | timeout=None, 208 | metadata=None): 209 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/SetWait', 210 | motion__server__pb2.SetWaitRequest.SerializeToString, 211 | motion__server__pb2.SetWaitReply.FromString, 212 | options, channel_credentials, 213 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 214 | 215 | @staticmethod 216 | def SetPos(request, 217 | target, 218 | options=(), 219 | channel_credentials=None, 220 | call_credentials=None, 221 | insecure=False, 222 | compression=None, 223 | wait_for_ready=None, 224 | timeout=None, 225 | metadata=None): 226 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/SetPos', 227 | motion__server__pb2.SetPosRequest.SerializeToString, 228 | motion__server__pb2.SetPosReply.FromString, 229 | options, channel_credentials, 230 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 231 | 232 | @staticmethod 233 | def SetVel(request, 234 | target, 235 | options=(), 236 | channel_credentials=None, 237 | call_credentials=None, 238 | insecure=False, 239 | compression=None, 240 | wait_for_ready=None, 241 | timeout=None, 242 | metadata=None): 243 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/SetVel', 244 | motion__server__pb2.SetVelRequest.SerializeToString, 245 | motion__server__pb2.SetVelReply.FromString, 246 | options, channel_credentials, 247 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 248 | 249 | @staticmethod 250 | def SetAcc(request, 251 | target, 252 | options=(), 253 | channel_credentials=None, 254 | call_credentials=None, 255 | insecure=False, 256 | compression=None, 257 | wait_for_ready=None, 258 | timeout=None, 259 | metadata=None): 260 | return grpc.experimental.unary_unary(request, target, '/motion_server.MotionServerService/SetAcc', 261 | motion__server__pb2.SetAccRequest.SerializeToString, 262 | motion__server__pb2.SetAccReply.FromString, 263 | options, channel_credentials, 264 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 265 | -------------------------------------------------------------------------------- /lib/grpc/speech_server_pb2.py: -------------------------------------------------------------------------------- 1 | # -*- coding: utf-8 -*- 2 | # Generated by the protocol buffer compiler. DO NOT EDIT! 3 | # source: speech_server.proto 4 | # Protobuf Python Version: 4.25.0 5 | """Generated protocol buffer code.""" 6 | from google.protobuf import descriptor as _descriptor 7 | from google.protobuf import descriptor_pool as _descriptor_pool 8 | from google.protobuf import symbol_database as _symbol_database 9 | from google.protobuf.internal import builder as _builder 10 | # @@protoc_insertion_point(imports) 11 | 12 | _sym_db = _symbol_database.Default() 13 | 14 | 15 | 16 | 17 | DESCRIPTOR = _descriptor_pool.Default().AddSerializedFile(b'\n\x13speech_server.proto\x12\rspeech_server\"%\n\x13ToggleSpeechRequest\x12\x0e\n\x06\x65nable\x18\x01 \x01(\x08\"$\n\x11ToggleSpeechReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\x32k\n\x13SpeechServerService\x12T\n\x0cToggleSpeech\x12\".speech_server.ToggleSpeechRequest\x1a .speech_server.ToggleSpeechReplyb\x06proto3') 18 | 19 | _globals = globals() 20 | _builder.BuildMessageAndEnumDescriptors(DESCRIPTOR, _globals) 21 | _builder.BuildTopDescriptorsAndMessages(DESCRIPTOR, 'speech_server_pb2', _globals) 22 | if _descriptor._USE_C_DESCRIPTORS == False: 23 | DESCRIPTOR._options = None 24 | _globals['_TOGGLESPEECHREQUEST']._serialized_start=38 25 | _globals['_TOGGLESPEECHREQUEST']._serialized_end=75 26 | _globals['_TOGGLESPEECHREPLY']._serialized_start=77 27 | _globals['_TOGGLESPEECHREPLY']._serialized_end=113 28 | _globals['_SPEECHSERVERSERVICE']._serialized_start=115 29 | _globals['_SPEECHSERVERSERVICE']._serialized_end=222 30 | # @@protoc_insertion_point(module_scope) 31 | -------------------------------------------------------------------------------- /lib/grpc/speech_server_pb2_grpc.py: -------------------------------------------------------------------------------- 1 | # Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT! 2 | """Client and server classes corresponding to protobuf-defined services.""" 3 | import grpc 4 | 5 | import speech_server_pb2 as speech__server__pb2 6 | 7 | 8 | class SpeechServerServiceStub(object): 9 | """Missing associated documentation comment in .proto file.""" 10 | 11 | def __init__(self, channel): 12 | """Constructor. 13 | 14 | Args: 15 | channel: A grpc.Channel. 16 | """ 17 | self.ToggleSpeech = channel.unary_unary( 18 | '/speech_server.SpeechServerService/ToggleSpeech', 19 | request_serializer=speech__server__pb2.ToggleSpeechRequest.SerializeToString, 20 | response_deserializer=speech__server__pb2.ToggleSpeechReply.FromString, 21 | ) 22 | 23 | 24 | class SpeechServerServiceServicer(object): 25 | """Missing associated documentation comment in .proto file.""" 26 | 27 | def ToggleSpeech(self, request, context): 28 | """Missing associated documentation comment in .proto file.""" 29 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 30 | context.set_details('Method not implemented!') 31 | raise NotImplementedError('Method not implemented!') 32 | 33 | 34 | def add_SpeechServerServiceServicer_to_server(servicer, server): 35 | rpc_method_handlers = { 36 | 'ToggleSpeech': grpc.unary_unary_rpc_method_handler( 37 | servicer.ToggleSpeech, 38 | request_deserializer=speech__server__pb2.ToggleSpeechRequest.FromString, 39 | response_serializer=speech__server__pb2.ToggleSpeechReply.SerializeToString, 40 | ), 41 | } 42 | generic_handler = grpc.method_handlers_generic_handler( 43 | 'speech_server.SpeechServerService', rpc_method_handlers) 44 | server.add_generic_rpc_handlers((generic_handler,)) 45 | 46 | 47 | # This class is part of an EXPERIMENTAL API. 48 | class SpeechServerService(object): 49 | """Missing associated documentation comment in .proto file.""" 50 | 51 | @staticmethod 52 | def ToggleSpeech(request, 53 | target, 54 | options=(), 55 | channel_credentials=None, 56 | call_credentials=None, 57 | insecure=False, 58 | compression=None, 59 | wait_for_ready=None, 60 | timeout=None, 61 | metadata=None): 62 | return grpc.experimental.unary_unary(request, target, '/speech_server.SpeechServerService/ToggleSpeech', 63 | speech__server__pb2.ToggleSpeechRequest.SerializeToString, 64 | speech__server__pb2.ToggleSpeechReply.FromString, 65 | options, channel_credentials, 66 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 67 | -------------------------------------------------------------------------------- /lib/grpc/voice_server_pb2.py: -------------------------------------------------------------------------------- 1 | # -*- coding: utf-8 -*- 2 | # Generated by the protocol buffer compiler. DO NOT EDIT! 3 | # source: voice_server.proto 4 | # Protobuf Python Version: 4.25.0 5 | """Generated protocol buffer code.""" 6 | from google.protobuf import descriptor as _descriptor 7 | from google.protobuf import descriptor_pool as _descriptor_pool 8 | from google.protobuf import symbol_database as _symbol_database 9 | from google.protobuf.internal import builder as _builder 10 | # @@protoc_insertion_point(imports) 11 | 12 | _sym_db = _symbol_database.Default() 13 | 14 | 15 | 16 | 17 | DESCRIPTOR = _descriptor_pool.Default().AddSerializedFile(b'\n\x12voice_server.proto\x12\x0cvoice_server\"\x1e\n\x0eSetTextRequest\x12\x0c\n\x04text\x18\x01 \x01(\t\"\x1f\n\x0cSetTextReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\xd4\x01\n\x1cSetStyleBertVitsParamRequest\x12\x17\n\nmodel_name\x18\x01 \x01(\tH\x00\x88\x01\x01\x12\x15\n\x08model_id\x18\x02 \x01(\x05H\x01\x88\x01\x01\x12\x13\n\x06length\x18\x03 \x01(\x02H\x02\x88\x01\x01\x12\x12\n\x05style\x18\x04 \x01(\tH\x03\x88\x01\x01\x12\x19\n\x0cstyle_weight\x18\x05 \x01(\x02H\x04\x88\x01\x01\x42\r\n\x0b_model_nameB\x0b\n\t_model_idB\t\n\x07_lengthB\x08\n\x06_styleB\x0f\n\r_style_weight\"-\n\x1aSetStyleBertVitsParamReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"e\n\x17SetVoicevoxParamRequest\x12\x14\n\x07speaker\x18\x01 \x01(\x05H\x00\x88\x01\x01\x12\x18\n\x0bspeed_scale\x18\x02 \x01(\x02H\x01\x88\x01\x01\x42\n\n\x08_speakerB\x0e\n\x0c_speed_scale\"(\n\x15SetVoicevoxParamReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x80\x01\n\x14SetAivisParamRequest\x12\x14\n\x07speaker\x18\x01 \x01(\tH\x00\x88\x01\x01\x12\x12\n\x05style\x18\x02 \x01(\tH\x01\x88\x01\x01\x12\x18\n\x0bspeed_scale\x18\x03 \x01(\x02H\x02\x88\x01\x01\x42\n\n\x08_speakerB\x08\n\x06_styleB\x0e\n\x0c_speed_scale\"%\n\x12SetAivisParamReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x17\n\x15InterruptVoiceRequest\"&\n\x13InterruptVoiceReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x18\n\x16\x45nableVoicePlayRequest\"\'\n\x14\x45nableVoicePlayReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x19\n\x17\x44isableVoicePlayRequest\"(\n\x15\x44isableVoicePlayReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x17\n\x15IsVoicePlayingRequest\")\n\x13IsVoicePlayingReply\x12\x12\n\nis_playing\x18\x01 \x01(\x08\"\x14\n\x12SentenceEndRequest\"#\n\x10SentenceEndReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\"\x19\n\x17StartHeadControlRequest\"(\n\x15StartHeadControlReply\x12\x0f\n\x07success\x18\x01 \x01(\x08\x32\xa1\x07\n\x12VoiceServerService\x12\x43\n\x07SetText\x12\x1c.voice_server.SetTextRequest\x1a\x1a.voice_server.SetTextReply\x12m\n\x15SetStyleBertVitsParam\x12*.voice_server.SetStyleBertVitsParamRequest\x1a(.voice_server.SetStyleBertVitsParamReply\x12^\n\x10SetVoicevoxParam\x12%.voice_server.SetVoicevoxParamRequest\x1a#.voice_server.SetVoicevoxParamReply\x12U\n\rSetAivisParam\x12\".voice_server.SetAivisParamRequest\x1a .voice_server.SetAivisParamReply\x12X\n\x0eInterruptVoice\x12#.voice_server.InterruptVoiceRequest\x1a!.voice_server.InterruptVoiceReply\x12[\n\x0f\x45nableVoicePlay\x12$.voice_server.EnableVoicePlayRequest\x1a\".voice_server.EnableVoicePlayReply\x12^\n\x10\x44isableVoicePlay\x12%.voice_server.DisableVoicePlayRequest\x1a#.voice_server.DisableVoicePlayReply\x12X\n\x0eIsVoicePlaying\x12#.voice_server.IsVoicePlayingRequest\x1a!.voice_server.IsVoicePlayingReply\x12O\n\x0bSentenceEnd\x12 .voice_server.SentenceEndRequest\x1a\x1e.voice_server.SentenceEndReply\x12^\n\x10StartHeadControl\x12%.voice_server.StartHeadControlRequest\x1a#.voice_server.StartHeadControlReplyb\x06proto3') 18 | 19 | _globals = globals() 20 | _builder.BuildMessageAndEnumDescriptors(DESCRIPTOR, _globals) 21 | _builder.BuildTopDescriptorsAndMessages(DESCRIPTOR, 'voice_server_pb2', _globals) 22 | if _descriptor._USE_C_DESCRIPTORS == False: 23 | DESCRIPTOR._options = None 24 | _globals['_SETTEXTREQUEST']._serialized_start=36 25 | _globals['_SETTEXTREQUEST']._serialized_end=66 26 | _globals['_SETTEXTREPLY']._serialized_start=68 27 | _globals['_SETTEXTREPLY']._serialized_end=99 28 | _globals['_SETSTYLEBERTVITSPARAMREQUEST']._serialized_start=102 29 | _globals['_SETSTYLEBERTVITSPARAMREQUEST']._serialized_end=314 30 | _globals['_SETSTYLEBERTVITSPARAMREPLY']._serialized_start=316 31 | _globals['_SETSTYLEBERTVITSPARAMREPLY']._serialized_end=361 32 | _globals['_SETVOICEVOXPARAMREQUEST']._serialized_start=363 33 | _globals['_SETVOICEVOXPARAMREQUEST']._serialized_end=464 34 | _globals['_SETVOICEVOXPARAMREPLY']._serialized_start=466 35 | _globals['_SETVOICEVOXPARAMREPLY']._serialized_end=506 36 | _globals['_SETAIVISPARAMREQUEST']._serialized_start=509 37 | _globals['_SETAIVISPARAMREQUEST']._serialized_end=637 38 | _globals['_SETAIVISPARAMREPLY']._serialized_start=639 39 | _globals['_SETAIVISPARAMREPLY']._serialized_end=676 40 | _globals['_INTERRUPTVOICEREQUEST']._serialized_start=678 41 | _globals['_INTERRUPTVOICEREQUEST']._serialized_end=701 42 | _globals['_INTERRUPTVOICEREPLY']._serialized_start=703 43 | _globals['_INTERRUPTVOICEREPLY']._serialized_end=741 44 | _globals['_ENABLEVOICEPLAYREQUEST']._serialized_start=743 45 | _globals['_ENABLEVOICEPLAYREQUEST']._serialized_end=767 46 | _globals['_ENABLEVOICEPLAYREPLY']._serialized_start=769 47 | _globals['_ENABLEVOICEPLAYREPLY']._serialized_end=808 48 | _globals['_DISABLEVOICEPLAYREQUEST']._serialized_start=810 49 | _globals['_DISABLEVOICEPLAYREQUEST']._serialized_end=835 50 | _globals['_DISABLEVOICEPLAYREPLY']._serialized_start=837 51 | _globals['_DISABLEVOICEPLAYREPLY']._serialized_end=877 52 | _globals['_ISVOICEPLAYINGREQUEST']._serialized_start=879 53 | _globals['_ISVOICEPLAYINGREQUEST']._serialized_end=902 54 | _globals['_ISVOICEPLAYINGREPLY']._serialized_start=904 55 | _globals['_ISVOICEPLAYINGREPLY']._serialized_end=945 56 | _globals['_SENTENCEENDREQUEST']._serialized_start=947 57 | _globals['_SENTENCEENDREQUEST']._serialized_end=967 58 | _globals['_SENTENCEENDREPLY']._serialized_start=969 59 | _globals['_SENTENCEENDREPLY']._serialized_end=1004 60 | _globals['_STARTHEADCONTROLREQUEST']._serialized_start=1006 61 | _globals['_STARTHEADCONTROLREQUEST']._serialized_end=1031 62 | _globals['_STARTHEADCONTROLREPLY']._serialized_start=1033 63 | _globals['_STARTHEADCONTROLREPLY']._serialized_end=1073 64 | _globals['_VOICESERVERSERVICE']._serialized_start=1076 65 | _globals['_VOICESERVERSERVICE']._serialized_end=2005 66 | # @@protoc_insertion_point(module_scope) 67 | -------------------------------------------------------------------------------- /lib/grpc/voice_server_pb2_grpc.py: -------------------------------------------------------------------------------- 1 | # Generated by the gRPC Python protocol compiler plugin. DO NOT EDIT! 2 | """Client and server classes corresponding to protobuf-defined services.""" 3 | import grpc 4 | 5 | import voice_server_pb2 as voice__server__pb2 6 | 7 | 8 | class VoiceServerServiceStub(object): 9 | """Missing associated documentation comment in .proto file.""" 10 | 11 | def __init__(self, channel): 12 | """Constructor. 13 | 14 | Args: 15 | channel: A grpc.Channel. 16 | """ 17 | self.SetText = channel.unary_unary( 18 | '/voice_server.VoiceServerService/SetText', 19 | request_serializer=voice__server__pb2.SetTextRequest.SerializeToString, 20 | response_deserializer=voice__server__pb2.SetTextReply.FromString, 21 | ) 22 | self.SetStyleBertVitsParam = channel.unary_unary( 23 | '/voice_server.VoiceServerService/SetStyleBertVitsParam', 24 | request_serializer=voice__server__pb2.SetStyleBertVitsParamRequest.SerializeToString, 25 | response_deserializer=voice__server__pb2.SetStyleBertVitsParamReply.FromString, 26 | ) 27 | self.SetVoicevoxParam = channel.unary_unary( 28 | '/voice_server.VoiceServerService/SetVoicevoxParam', 29 | request_serializer=voice__server__pb2.SetVoicevoxParamRequest.SerializeToString, 30 | response_deserializer=voice__server__pb2.SetVoicevoxParamReply.FromString, 31 | ) 32 | self.SetAivisParam = channel.unary_unary( 33 | '/voice_server.VoiceServerService/SetAivisParam', 34 | request_serializer=voice__server__pb2.SetAivisParamRequest.SerializeToString, 35 | response_deserializer=voice__server__pb2.SetAivisParamReply.FromString, 36 | ) 37 | self.InterruptVoice = channel.unary_unary( 38 | '/voice_server.VoiceServerService/InterruptVoice', 39 | request_serializer=voice__server__pb2.InterruptVoiceRequest.SerializeToString, 40 | response_deserializer=voice__server__pb2.InterruptVoiceReply.FromString, 41 | ) 42 | self.EnableVoicePlay = channel.unary_unary( 43 | '/voice_server.VoiceServerService/EnableVoicePlay', 44 | request_serializer=voice__server__pb2.EnableVoicePlayRequest.SerializeToString, 45 | response_deserializer=voice__server__pb2.EnableVoicePlayReply.FromString, 46 | ) 47 | self.DisableVoicePlay = channel.unary_unary( 48 | '/voice_server.VoiceServerService/DisableVoicePlay', 49 | request_serializer=voice__server__pb2.DisableVoicePlayRequest.SerializeToString, 50 | response_deserializer=voice__server__pb2.DisableVoicePlayReply.FromString, 51 | ) 52 | self.IsVoicePlaying = channel.unary_unary( 53 | '/voice_server.VoiceServerService/IsVoicePlaying', 54 | request_serializer=voice__server__pb2.IsVoicePlayingRequest.SerializeToString, 55 | response_deserializer=voice__server__pb2.IsVoicePlayingReply.FromString, 56 | ) 57 | self.SentenceEnd = channel.unary_unary( 58 | '/voice_server.VoiceServerService/SentenceEnd', 59 | request_serializer=voice__server__pb2.SentenceEndRequest.SerializeToString, 60 | response_deserializer=voice__server__pb2.SentenceEndReply.FromString, 61 | ) 62 | self.StartHeadControl = channel.unary_unary( 63 | '/voice_server.VoiceServerService/StartHeadControl', 64 | request_serializer=voice__server__pb2.StartHeadControlRequest.SerializeToString, 65 | response_deserializer=voice__server__pb2.StartHeadControlReply.FromString, 66 | ) 67 | 68 | 69 | class VoiceServerServiceServicer(object): 70 | """Missing associated documentation comment in .proto file.""" 71 | 72 | def SetText(self, request, context): 73 | """Missing associated documentation comment in .proto file.""" 74 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 75 | context.set_details('Method not implemented!') 76 | raise NotImplementedError('Method not implemented!') 77 | 78 | def SetStyleBertVitsParam(self, request, context): 79 | """Missing associated documentation comment in .proto file.""" 80 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 81 | context.set_details('Method not implemented!') 82 | raise NotImplementedError('Method not implemented!') 83 | 84 | def SetVoicevoxParam(self, request, context): 85 | """Missing associated documentation comment in .proto file.""" 86 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 87 | context.set_details('Method not implemented!') 88 | raise NotImplementedError('Method not implemented!') 89 | 90 | def SetAivisParam(self, request, context): 91 | """Missing associated documentation comment in .proto file.""" 92 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 93 | context.set_details('Method not implemented!') 94 | raise NotImplementedError('Method not implemented!') 95 | 96 | def InterruptVoice(self, request, context): 97 | """Missing associated documentation comment in .proto file.""" 98 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 99 | context.set_details('Method not implemented!') 100 | raise NotImplementedError('Method not implemented!') 101 | 102 | def EnableVoicePlay(self, request, context): 103 | """Missing associated documentation comment in .proto file.""" 104 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 105 | context.set_details('Method not implemented!') 106 | raise NotImplementedError('Method not implemented!') 107 | 108 | def DisableVoicePlay(self, request, context): 109 | """Missing associated documentation comment in .proto file.""" 110 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 111 | context.set_details('Method not implemented!') 112 | raise NotImplementedError('Method not implemented!') 113 | 114 | def IsVoicePlaying(self, request, context): 115 | """Missing associated documentation comment in .proto file.""" 116 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 117 | context.set_details('Method not implemented!') 118 | raise NotImplementedError('Method not implemented!') 119 | 120 | def SentenceEnd(self, request, context): 121 | """Missing associated documentation comment in .proto file.""" 122 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 123 | context.set_details('Method not implemented!') 124 | raise NotImplementedError('Method not implemented!') 125 | 126 | def StartHeadControl(self, request, context): 127 | """Missing associated documentation comment in .proto file.""" 128 | context.set_code(grpc.StatusCode.UNIMPLEMENTED) 129 | context.set_details('Method not implemented!') 130 | raise NotImplementedError('Method not implemented!') 131 | 132 | 133 | def add_VoiceServerServiceServicer_to_server(servicer, server): 134 | rpc_method_handlers = { 135 | 'SetText': grpc.unary_unary_rpc_method_handler( 136 | servicer.SetText, 137 | request_deserializer=voice__server__pb2.SetTextRequest.FromString, 138 | response_serializer=voice__server__pb2.SetTextReply.SerializeToString, 139 | ), 140 | 'SetStyleBertVitsParam': grpc.unary_unary_rpc_method_handler( 141 | servicer.SetStyleBertVitsParam, 142 | request_deserializer=voice__server__pb2.SetStyleBertVitsParamRequest.FromString, 143 | response_serializer=voice__server__pb2.SetStyleBertVitsParamReply.SerializeToString, 144 | ), 145 | 'SetVoicevoxParam': grpc.unary_unary_rpc_method_handler( 146 | servicer.SetVoicevoxParam, 147 | request_deserializer=voice__server__pb2.SetVoicevoxParamRequest.FromString, 148 | response_serializer=voice__server__pb2.SetVoicevoxParamReply.SerializeToString, 149 | ), 150 | 'SetAivisParam': grpc.unary_unary_rpc_method_handler( 151 | servicer.SetAivisParam, 152 | request_deserializer=voice__server__pb2.SetAivisParamRequest.FromString, 153 | response_serializer=voice__server__pb2.SetAivisParamReply.SerializeToString, 154 | ), 155 | 'InterruptVoice': grpc.unary_unary_rpc_method_handler( 156 | servicer.InterruptVoice, 157 | request_deserializer=voice__server__pb2.InterruptVoiceRequest.FromString, 158 | response_serializer=voice__server__pb2.InterruptVoiceReply.SerializeToString, 159 | ), 160 | 'EnableVoicePlay': grpc.unary_unary_rpc_method_handler( 161 | servicer.EnableVoicePlay, 162 | request_deserializer=voice__server__pb2.EnableVoicePlayRequest.FromString, 163 | response_serializer=voice__server__pb2.EnableVoicePlayReply.SerializeToString, 164 | ), 165 | 'DisableVoicePlay': grpc.unary_unary_rpc_method_handler( 166 | servicer.DisableVoicePlay, 167 | request_deserializer=voice__server__pb2.DisableVoicePlayRequest.FromString, 168 | response_serializer=voice__server__pb2.DisableVoicePlayReply.SerializeToString, 169 | ), 170 | 'IsVoicePlaying': grpc.unary_unary_rpc_method_handler( 171 | servicer.IsVoicePlaying, 172 | request_deserializer=voice__server__pb2.IsVoicePlayingRequest.FromString, 173 | response_serializer=voice__server__pb2.IsVoicePlayingReply.SerializeToString, 174 | ), 175 | 'SentenceEnd': grpc.unary_unary_rpc_method_handler( 176 | servicer.SentenceEnd, 177 | request_deserializer=voice__server__pb2.SentenceEndRequest.FromString, 178 | response_serializer=voice__server__pb2.SentenceEndReply.SerializeToString, 179 | ), 180 | 'StartHeadControl': grpc.unary_unary_rpc_method_handler( 181 | servicer.StartHeadControl, 182 | request_deserializer=voice__server__pb2.StartHeadControlRequest.FromString, 183 | response_serializer=voice__server__pb2.StartHeadControlReply.SerializeToString, 184 | ), 185 | } 186 | generic_handler = grpc.method_handlers_generic_handler( 187 | 'voice_server.VoiceServerService', rpc_method_handlers) 188 | server.add_generic_rpc_handlers((generic_handler,)) 189 | 190 | 191 | # This class is part of an EXPERIMENTAL API. 192 | class VoiceServerService(object): 193 | """Missing associated documentation comment in .proto file.""" 194 | 195 | @staticmethod 196 | def SetText(request, 197 | target, 198 | options=(), 199 | channel_credentials=None, 200 | call_credentials=None, 201 | insecure=False, 202 | compression=None, 203 | wait_for_ready=None, 204 | timeout=None, 205 | metadata=None): 206 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/SetText', 207 | voice__server__pb2.SetTextRequest.SerializeToString, 208 | voice__server__pb2.SetTextReply.FromString, 209 | options, channel_credentials, 210 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 211 | 212 | @staticmethod 213 | def SetStyleBertVitsParam(request, 214 | target, 215 | options=(), 216 | channel_credentials=None, 217 | call_credentials=None, 218 | insecure=False, 219 | compression=None, 220 | wait_for_ready=None, 221 | timeout=None, 222 | metadata=None): 223 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/SetStyleBertVitsParam', 224 | voice__server__pb2.SetStyleBertVitsParamRequest.SerializeToString, 225 | voice__server__pb2.SetStyleBertVitsParamReply.FromString, 226 | options, channel_credentials, 227 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 228 | 229 | @staticmethod 230 | def SetVoicevoxParam(request, 231 | target, 232 | options=(), 233 | channel_credentials=None, 234 | call_credentials=None, 235 | insecure=False, 236 | compression=None, 237 | wait_for_ready=None, 238 | timeout=None, 239 | metadata=None): 240 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/SetVoicevoxParam', 241 | voice__server__pb2.SetVoicevoxParamRequest.SerializeToString, 242 | voice__server__pb2.SetVoicevoxParamReply.FromString, 243 | options, channel_credentials, 244 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 245 | 246 | @staticmethod 247 | def SetAivisParam(request, 248 | target, 249 | options=(), 250 | channel_credentials=None, 251 | call_credentials=None, 252 | insecure=False, 253 | compression=None, 254 | wait_for_ready=None, 255 | timeout=None, 256 | metadata=None): 257 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/SetAivisParam', 258 | voice__server__pb2.SetAivisParamRequest.SerializeToString, 259 | voice__server__pb2.SetAivisParamReply.FromString, 260 | options, channel_credentials, 261 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 262 | 263 | @staticmethod 264 | def InterruptVoice(request, 265 | target, 266 | options=(), 267 | channel_credentials=None, 268 | call_credentials=None, 269 | insecure=False, 270 | compression=None, 271 | wait_for_ready=None, 272 | timeout=None, 273 | metadata=None): 274 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/InterruptVoice', 275 | voice__server__pb2.InterruptVoiceRequest.SerializeToString, 276 | voice__server__pb2.InterruptVoiceReply.FromString, 277 | options, channel_credentials, 278 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 279 | 280 | @staticmethod 281 | def EnableVoicePlay(request, 282 | target, 283 | options=(), 284 | channel_credentials=None, 285 | call_credentials=None, 286 | insecure=False, 287 | compression=None, 288 | wait_for_ready=None, 289 | timeout=None, 290 | metadata=None): 291 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/EnableVoicePlay', 292 | voice__server__pb2.EnableVoicePlayRequest.SerializeToString, 293 | voice__server__pb2.EnableVoicePlayReply.FromString, 294 | options, channel_credentials, 295 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 296 | 297 | @staticmethod 298 | def DisableVoicePlay(request, 299 | target, 300 | options=(), 301 | channel_credentials=None, 302 | call_credentials=None, 303 | insecure=False, 304 | compression=None, 305 | wait_for_ready=None, 306 | timeout=None, 307 | metadata=None): 308 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/DisableVoicePlay', 309 | voice__server__pb2.DisableVoicePlayRequest.SerializeToString, 310 | voice__server__pb2.DisableVoicePlayReply.FromString, 311 | options, channel_credentials, 312 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 313 | 314 | @staticmethod 315 | def IsVoicePlaying(request, 316 | target, 317 | options=(), 318 | channel_credentials=None, 319 | call_credentials=None, 320 | insecure=False, 321 | compression=None, 322 | wait_for_ready=None, 323 | timeout=None, 324 | metadata=None): 325 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/IsVoicePlaying', 326 | voice__server__pb2.IsVoicePlayingRequest.SerializeToString, 327 | voice__server__pb2.IsVoicePlayingReply.FromString, 328 | options, channel_credentials, 329 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 330 | 331 | @staticmethod 332 | def SentenceEnd(request, 333 | target, 334 | options=(), 335 | channel_credentials=None, 336 | call_credentials=None, 337 | insecure=False, 338 | compression=None, 339 | wait_for_ready=None, 340 | timeout=None, 341 | metadata=None): 342 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/SentenceEnd', 343 | voice__server__pb2.SentenceEndRequest.SerializeToString, 344 | voice__server__pb2.SentenceEndReply.FromString, 345 | options, channel_credentials, 346 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 347 | 348 | @staticmethod 349 | def StartHeadControl(request, 350 | target, 351 | options=(), 352 | channel_credentials=None, 353 | call_credentials=None, 354 | insecure=False, 355 | compression=None, 356 | wait_for_ready=None, 357 | timeout=None, 358 | metadata=None): 359 | return grpc.experimental.unary_unary(request, target, '/voice_server.VoiceServerService/StartHeadControl', 360 | voice__server__pb2.StartHeadControlRequest.SerializeToString, 361 | voice__server__pb2.StartHeadControlReply.FromString, 362 | options, channel_credentials, 363 | insecure, call_credentials, compression, wait_for_ready, timeout, metadata) 364 | -------------------------------------------------------------------------------- /lib/style_bert_vits.py: -------------------------------------------------------------------------------- 1 | import json 2 | from typing import Optional 3 | from urllib.parse import urlencode 4 | from urllib.request import Request, urlopen 5 | 6 | from lib.text_to_voice import TextToVoice 7 | 8 | 9 | class TextToStyleBertVits(TextToVoice): 10 | """ 11 | Style-Bert-VITS2を使用してテキストから音声を生成するクラス。 12 | """ 13 | 14 | def __init__( 15 | self, 16 | host: str = "127.0.0.1", 17 | port: str = "5000", 18 | motion_host: Optional[str] = "127.0.0.1", 19 | motion_port: Optional[str] = "50055", 20 | ) -> None: 21 | """クラスの初期化メソッド。 22 | Args: 23 | host (str, optional): Style-Bert-VITS2サーバーのホスト名。デフォルトは "127.0.0.1"。 24 | port (str, optional): Style-Bert-VITS2サーバーのポート番号。デフォルトは"5000"。 25 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 26 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 27 | 28 | """ 29 | super().__init__(host, port, motion_host, motion_port) 30 | self.model_id = 0 31 | self.length = 1.0 32 | self.style = "Neutral" 33 | self.style_weight = 1.0 34 | # 話者モデル名を指定 35 | self.set_param(model_name="jvnv-F1-jp") 36 | 37 | def get_model_id_from_name(self, model_name: str) -> int: 38 | """ 39 | モデル名からモデル番号を取得する。 40 | 41 | Args: 42 | model_name (str): モデル名。 43 | 44 | Returns: 45 | int: モデル番号。 46 | 47 | """ 48 | headers = {"accept": "application/json"} 49 | address = "http://" + self.host + ":" + self.port + "/models/info" 50 | # GETリクエストを作成 51 | req = Request(address, headers=headers, method="GET") 52 | with urlopen(req) as res: 53 | model_info = res.read() 54 | model_info_json = json.loads(model_info) 55 | for key, details in model_info_json.items(): 56 | if model_name == details["id2spk"]["0"]: 57 | return key 58 | raise ValueError("Model name not found") 59 | 60 | def set_param( 61 | self, 62 | model_name: Optional[str] = None, 63 | model_id: Optional[int] = None, 64 | length: Optional[float] = None, 65 | style: Optional[str] = None, 66 | style_weight: Optional[float] = None, 67 | ) -> None: 68 | """ 69 | 音声合成のパラメータを設定する。 70 | 71 | Args: 72 | model_name (str, optional): Style-Bert-VITS2のモデル名。デフォルトはNone。 73 | model_id (int, optional): Style-Bert-VITS2のモデル番号。デフォルトはNone。 74 | length (float, optional): 音声の再生速度。大きくする程読み上げ速度が遅くなる。デフォルトはNone。 75 | style (str, optional): 音声の感情スタイル。デフォルトはNone。 76 | style_weight (float, optional): 音声の感情スタイルの重み。値が大きいほど感情の影響が大きくなる。デフォルトはNone。 77 | 78 | """ 79 | if model_name is not None: 80 | self.model_id = self.get_model_id_from_name(model_name) 81 | elif model_id is not None: 82 | self.model_id = model_id 83 | if length is not None: 84 | self.length = length 85 | if style is not None: 86 | self.style = style 87 | if style_weight is not None: 88 | self.style_weight = style_weight 89 | 90 | def post_synthesis( 91 | self, 92 | text: str, 93 | ) -> Optional[bytes]: 94 | """ 95 | Style-Bert-VITS2サーバーに音声合成要求を送信し、合成された音声データを取得する。 96 | 97 | Args: 98 | text (str): 音声合成対象のテキスト。 99 | 100 | Returns: 101 | Any: 音声合成クエリの応答。 102 | 103 | """ 104 | if len(text.strip()) <= 0: 105 | return None 106 | headers = {"accept": "audio/wav"} 107 | params = { 108 | "text": text, 109 | "model_id": self.model_id, 110 | "length": self.length, 111 | "style": self.style, 112 | "style_weight": self.style_weight, 113 | } 114 | address = ( 115 | "http://" + self.host + ":" + self.port + "/voice" + "?" + urlencode(params) 116 | ) 117 | # GETリクエストを作成 118 | req = Request(address, headers=headers, method="GET") 119 | with urlopen(req) as res: 120 | return res.read() 121 | 122 | def text_to_voice(self, text: str) -> None: 123 | """ 124 | テキストから音声を合成して再生する。 125 | Args: 126 | text (str): 音声合成対象のテキスト。 127 | """ 128 | wav = self.post_synthesis(text) 129 | if wav is not None: 130 | print(f"[Play] {text}") 131 | self.play_wav(wav) 132 | -------------------------------------------------------------------------------- /lib/text_to_voice.py: -------------------------------------------------------------------------------- 1 | import io 2 | import os 3 | import sys 4 | import time 5 | import wave 6 | from abc import ABCMeta, abstractmethod 7 | from queue import Queue 8 | from threading import Event, Thread 9 | from typing import Any, Optional 10 | 11 | import grpc 12 | import numpy as np 13 | import pyaudio 14 | from lib.en_to_jp import EnToJp 15 | 16 | from .err_handler import ignoreStderr 17 | 18 | sys.path.append(os.path.join(os.path.dirname(__file__), "grpc")) 19 | import motion_server_pb2 20 | import motion_server_pb2_grpc 21 | 22 | 23 | class TextToVoice(metaclass=ABCMeta): 24 | """ 25 | 音声合成を使用してテキストから音声を生成するクラス。 26 | """ 27 | 28 | def __init__( 29 | self, 30 | host: str = "127.0.0.1", 31 | port: str = "52001", 32 | motion_host: Optional[str] = "127.0.0.1", 33 | motion_port: Optional[str] = "50055", 34 | ) -> None: 35 | """クラスの初期化メソッド。 36 | Args: 37 | host (str, optional): サーバーのホスト名。デフォルトは "127.0.0.1"。 38 | port (str, optional): サーバーのポート番号。デフォルトは "52001"。 39 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 40 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 41 | 42 | """ 43 | self.queue: Queue[str] = Queue() 44 | self.host = host 45 | self.port = port 46 | self.motion_stub = None 47 | if motion_host is not None or motion_port is not None: 48 | motion_channel = grpc.insecure_channel(motion_host + ":" + motion_port) 49 | self.motion_stub = motion_server_pb2_grpc.MotionServerServiceStub( 50 | motion_channel 51 | ) 52 | self.finished = True # 音声再生が完了したかどうかを示すフラグ 53 | self.sentence_end_flg = False # 一文の終わりを示すフラグ 54 | self.sentence_end_timeout = 5.0 # 一文の終わりを判定するタイムアウト時間 55 | self.tilt_rate = 0.0 # 送信するtiltのrate(0.0~1.0) 56 | self.HEAD_RESET_INTERVAL = 0.3 # この時間更新がなければ、tiltの指令値を0にリセットする[sec] 57 | self.TILT_GAIN = -0.8 # 音声出力の音量からtiltのrateに変換するゲイン 58 | self.TILT_RATE_DB_MAX = 40.0 # tilt_rate上限の音声出力値[dB] 59 | self.TILT_RATE_DB_MIN = 5.0 # tilt_rate下限の音声出力値[dB] 60 | self.TILT_ANGLE_MAX = 0.35 # Tiltの最大角度[rad] 61 | self.TILT_ANGLE_MIN = -0.1 # Tiltの最小角度[rad] 62 | self.HEAD_MOTION_INTERVAL = 0.15 # ヘッドモーションの更新周期[sec] 63 | self.event = Event() 64 | self.head_motion_thread = Thread(target=self.head_motion_control, daemon=True) 65 | if self.motion_stub is not None: 66 | self.head_motion_thread.start() 67 | self.text_to_voice_event = Event() 68 | self.voice_thread = Thread(target=self.text_to_voice_thread) 69 | self.voice_thread.start() 70 | self.en_to_jp = EnToJp() 71 | 72 | def __exit__(self) -> None: 73 | """音声合成スレッドを終了する。""" 74 | self.voice_thread.join() 75 | 76 | def sentence_end(self) -> None: 77 | """音声合成の一文の終わりを示すフラグを立てる。""" 78 | self.sentence_end_flg = True 79 | 80 | def enable_voice_play(self) -> None: 81 | """音声再生を開始する。""" 82 | self.text_to_voice_event.set() 83 | 84 | def disable_voice_play(self) -> None: 85 | """音声再生を停止する。""" 86 | self.text_to_voice_event.clear() 87 | 88 | def text_to_voice_thread(self) -> None: 89 | """ 90 | 音声合成スレッドの実行関数。 91 | キューからテキストを取り出し、text_to_voice関数を呼び出す。 92 | 93 | """ 94 | last_queue_time = time.time() 95 | queue_start = False 96 | while True: 97 | self.text_to_voice_event.wait() 98 | if self.queue.qsize() > 0: 99 | queue_start = True 100 | last_queue_time = time.time() 101 | text = self.queue.get() 102 | # textに含まれる英語を極力かな変換する 103 | text = self.en_to_jp.text_to_kana(text, True, True, True) 104 | self.text_to_voice(text) 105 | else: 106 | # queueが空の状態でsentence_endが送られる、もしくはsentence_end_timeout秒経過した場合finishedにする。 107 | if self.sentence_end_flg or ( 108 | queue_start 109 | and time.time() - last_queue_time > self.sentence_end_timeout 110 | ): 111 | self.finished = True 112 | queue_start = False 113 | if self.motion_stub is not None: 114 | self.event.clear() 115 | if self.motion_stub is not None: 116 | # 初期位置にヘッドを戻す 117 | try: 118 | self.motion_stub.SetPos( 119 | motion_server_pb2.SetPosRequest( 120 | tilt=self.TILT_ANGLE_MAX, priority=3 121 | ) 122 | ) 123 | except BaseException as e: 124 | print(f"Failed to send SetPos command: {e}") 125 | pass 126 | self.sentence_end_flg = False 127 | self.text_to_voice_event.clear() 128 | 129 | def put_text( 130 | self, text: str, play_now: bool = True, blocking: bool = False 131 | ) -> None: 132 | """ 133 | 音声合成のためのテキストをキューに追加する。 134 | 135 | Args: 136 | text (str): 音声合成対象のテキスト。 137 | play_now (bool, optional): すぐに音声再生を開始するかどうか。デフォルトはTrue。 138 | blocking (bool, optional): 音声合成が完了するまでブロックするかどうか。デフォルトはFalse。 139 | 140 | """ 141 | if play_now: 142 | self.text_to_voice_event.set() 143 | self.queue.put(text) 144 | self.finished = False 145 | if blocking: 146 | self.wait_finish() 147 | 148 | def wait_finish(self) -> None: 149 | """ 150 | 音声合成が完了するまで待機するループ関数。 151 | 152 | """ 153 | while not self.finished: 154 | time.sleep(0.01) 155 | 156 | @abstractmethod 157 | def set_param( 158 | self, 159 | speaker: Optional[int] = None, 160 | speed_scale: Optional[float] = None, 161 | ) -> None: 162 | """ 163 | 音声合成のパラメータを設定する。 164 | 165 | Args: 166 | speaker (Optional[int], optional): VoiceVoxの話者番号。デフォルトはNone。 167 | speed_scale (Optional[float], optional): 音声の再生速度スケール。デフォルトはNone。 168 | 169 | """ 170 | ... 171 | 172 | def play_wav(self, wav_file: bytes) -> None: 173 | """合成された音声データを再生する。 174 | 175 | Args: 176 | wav_file (bytes): 合成された音声データ。 177 | 178 | """ 179 | wr: wave.Wave_read = wave.open(io.BytesIO(wav_file)) 180 | with ignoreStderr(): 181 | p = pyaudio.PyAudio() 182 | stream = p.open( 183 | format=p.get_format_from_width(wr.getsampwidth()), 184 | channels=wr.getnchannels(), 185 | rate=wr.getframerate(), 186 | output=True, 187 | ) 188 | chunk = 1024 189 | data = wr.readframes(chunk) 190 | while data: 191 | audio_data = np.frombuffer(data, dtype=np.int16) 192 | rms = np.sqrt(np.mean(audio_data**2)) 193 | db = 20 * np.log10(rms) if rms > 0.0 else 0.0 194 | self.tilt_rate = self.db_to_head_rate(db) 195 | stream.write(data) 196 | data = wr.readframes(chunk) 197 | time.sleep(0.2) 198 | stream.close() 199 | p.terminate() 200 | 201 | @abstractmethod 202 | def text_to_voice(self, text: str) -> None: 203 | """ 204 | テキストから音声を合成して再生する。 205 | 206 | Args: 207 | text (str): 音声合成対象のテキスト。 208 | 209 | """ 210 | ... 211 | 212 | def is_playing(self) -> bool: 213 | """ 214 | 音声再生が実行中かどうかを返す。 215 | queueの中身が0かつ再生中の音声がなければFalseを返す。 216 | 217 | Returns: 218 | bool: 音声再生中の場合はTrue。 219 | 220 | """ 221 | return self.finished 222 | 223 | def db_to_head_rate(self, db: float) -> float: 224 | """ 225 | 音声の音量[dB]からヘッドの動き具合を算出する。 226 | Args: 227 | db (float): 音声の音量[dB]。 228 | Returns: 229 | float: ヘッドの動き具合。 230 | """ 231 | if db > self.TILT_RATE_DB_MAX: 232 | return 1.0 233 | elif db < self.TILT_RATE_DB_MIN: 234 | return 0.0 235 | return (db - self.TILT_RATE_DB_MIN) / ( 236 | self.TILT_RATE_DB_MAX - self.TILT_RATE_DB_MIN 237 | ) 238 | 239 | def head_motion_control(self) -> None: 240 | """ 241 | 音声出力に合わせてヘッドを動かす。 242 | """ 243 | last_update_time = time.time() 244 | prev_tilt_rate = 0.0 245 | while True: 246 | self.event.wait() 247 | loop_start_time = time.time() 248 | if self.tilt_rate != prev_tilt_rate: 249 | val = ( 250 | -1 * self.tilt_rate * (self.TILT_ANGLE_MAX - self.TILT_ANGLE_MIN) 251 | + self.TILT_ANGLE_MAX 252 | ) 253 | if self.motion_stub is not None: 254 | try: 255 | self.motion_stub.ClearMotion( 256 | motion_server_pb2.ClearMotionRequest(priority=3) 257 | ) 258 | except BaseException as e: 259 | print(f"Failed to ClearMotion command: {e}") 260 | pass 261 | try: 262 | self.motion_stub.SetPos( 263 | motion_server_pb2.SetPosRequest(tilt=val, priority=3) 264 | ) 265 | except BaseException as e: 266 | print(f"Failed to send SetPos command: {e}") 267 | pass 268 | last_update_time = time.time() 269 | prev_tilt_rate = self.tilt_rate 270 | if time.time() - last_update_time > self.HEAD_RESET_INTERVAL: 271 | self.tilt_rate = 0.0 272 | wait_time = self.HEAD_MOTION_INTERVAL - (time.time() - loop_start_time) 273 | if wait_time > 0: 274 | time.sleep(wait_time) 275 | 276 | def start_head_control(self) -> None: 277 | """ 278 | ヘッドモーションを開始する。 279 | """ 280 | if self.motion_stub is not None: 281 | self.event.set() 282 | -------------------------------------------------------------------------------- /lib/voicevox.py: -------------------------------------------------------------------------------- 1 | import json 2 | from queue import Queue 3 | from typing import Any, Optional 4 | 5 | import requests 6 | from lib.text_to_voice import TextToVoice 7 | 8 | 9 | class TextToVoiceVox(TextToVoice): 10 | """ 11 | VoiceVoxを使用してテキストから音声を生成するクラス。 12 | """ 13 | 14 | def __init__( 15 | self, 16 | host: str = "127.0.0.1", 17 | port: str = "52001", 18 | motion_host: Optional[str] = "127.0.0.1", 19 | motion_port: Optional[str] = "50055", 20 | ) -> None: 21 | """クラスの初期化メソッド。 22 | Args: 23 | host (str, optional): VoiceVoxサーバーのホスト名。デフォルトは "127.0.0.1"。 24 | port (str, optional): VoiceVoxサーバーのポート番号。デフォルトは "52001"。 25 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 26 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 27 | 28 | """ 29 | super().__init__( 30 | host=host, port=port, motion_host=motion_host, motion_port=motion_port 31 | ) 32 | # デフォルトのspeakerは8(春日部つむぎ) 33 | self.speaker = 8 34 | self.speed_scale = 1.0 35 | 36 | def set_param( 37 | self, 38 | speaker: Optional[int] = None, 39 | speed_scale: Optional[float] = None, 40 | ) -> None: 41 | """ 42 | 音声合成のパラメータを設定する。 43 | 44 | Args: 45 | speaker (Optional[int], optional): VoiceVoxの話者番号。デフォルトはNone。 46 | speed_scale (Optional[float], optional): 音声の再生速度スケール。デフォルトはNone。 47 | 48 | """ 49 | if speaker is not None: 50 | self.speaker = speaker 51 | elif speed_scale is not None: 52 | self.speed_scale = speed_scale 53 | 54 | def post_audio_query( 55 | self, 56 | text: str, 57 | ) -> Any: 58 | """VoiceVoxサーバーに音声合成クエリを送信する。 59 | 60 | Args: 61 | text (str): 音声合成対象のテキスト。 62 | speaker (int, optional): VoiceVoxの話者番号。デフォルトは8(春日部つむぎ)。 63 | speed_scale (float, optional): 音声の再生速度スケール。デフォルトは1.0。 64 | 65 | Returns: 66 | Any: 音声合成クエリの応答。 67 | 68 | """ 69 | if len(text.strip()) <= 0: 70 | return None 71 | params = { 72 | "text": text, 73 | "speaker": self.speaker, 74 | "speedScale": self.speed_scale, 75 | "prePhonemeLength": 0, 76 | "postPhonemeLength": 0, 77 | } 78 | address = "http://" + self.host + ":" + self.port + "/audio_query" 79 | res = requests.post(address, params=params) 80 | return res.json() 81 | 82 | def post_synthesis( 83 | self, 84 | audio_query_response: dict, 85 | ) -> bytes: 86 | """ 87 | VoiceVoxサーバーに音声合成要求を送信し、合成された音声データを取得する。 88 | 89 | Args: 90 | audio_query_response (dict): 音声合成クエリの応答。 91 | 92 | Returns: 93 | bytes: 合成された音声データ。 94 | """ 95 | params = {"speaker": self.speaker} 96 | headers = {"content-type": "application/json"} 97 | audio_query_response["speedScale"] = self.speed_scale 98 | audio_query_response_json = json.dumps(audio_query_response) 99 | address = "http://" + self.host + ":" + self.port + "/synthesis" 100 | res = requests.post( 101 | address, data=audio_query_response_json, params=params, headers=headers 102 | ) 103 | return res.content 104 | 105 | def text_to_voice(self, text: str) -> None: 106 | """ 107 | テキストから音声を合成して再生する。 108 | 109 | Args: 110 | text (str): 音声合成対象のテキスト。 111 | 112 | """ 113 | res = self.post_audio_query(text) 114 | if res is None: 115 | return 116 | wav = self.post_synthesis(res) 117 | if wav is not None: 118 | print(f"[Play] {text}") 119 | self.play_wav(wav) 120 | 121 | 122 | class TextToVoiceVoxWeb(TextToVoiceVox): 123 | """ 124 | VoiceVox(web版)を使用してテキストから音声を生成するクラス。 125 | """ 126 | 127 | def __init__( 128 | self, 129 | apikey: str, 130 | motion_host: Optional[str] = "127.0.0.1", 131 | motion_port: Optional[str] = "50055", 132 | ) -> None: 133 | """クラスの初期化メソッド。 134 | Args: 135 | apikey (str): VoiceVox wweb版のAPIキー。 136 | motion_host (str, optional): モーションサーバーのホスト名。デフォルトは"127.0.0.1"。 137 | motion_port (str, optional): モーションサーバーのポート番号。デフォルトは"50055"。 138 | 139 | """ 140 | super().__init__( 141 | host="127.0.0.1", 142 | port="0000", 143 | motion_host=motion_host, 144 | motion_port=motion_port, 145 | ) 146 | self.queue: Queue[str] = Queue() 147 | self.apikey = apikey 148 | 149 | def post_web( 150 | self, 151 | text: str, 152 | speaker: int = 8, 153 | pitch: int = 0, 154 | intonation_scale: int = 1, 155 | speed: int = 1, 156 | ) -> Optional[bytes]: 157 | """ 158 | VoiceVoxウェブAPIに音声合成要求を送信し、合成された音声データを取得。 159 | 160 | Args: 161 | text (str): 音声合成対象のテキスト。 162 | speaker (int, optional): VoiceVoxの話者番号。デフォルトは8(春日部つむぎ)。 163 | pitch (int, optional): ピッチ。デフォルトは0。 164 | intonation_scale (int, optional): イントネーションスケール。デフォルトは1。 165 | speed (int, optional): 音声の速度。デフォルトは1。 166 | 167 | Returns: 168 | bytes: 合成された音声データ。 169 | 170 | """ 171 | if len(text.strip()) <= 0: 172 | return None 173 | address = ( 174 | "https://deprecatedapis.tts.quest/v2/voicevox/audio/?key=" 175 | + self.apikey 176 | + "&speaker=" 177 | + str(speaker) 178 | + "&pitch=" 179 | + str(pitch) 180 | + "&intonationScale=" 181 | + str(intonation_scale) 182 | + "&speed=" 183 | + str(speed) 184 | + "&text=" 185 | + text 186 | ) 187 | res = requests.post(address) 188 | return res.content 189 | 190 | def text_to_voice(self, text: str) -> None: 191 | """ 192 | テキストから音声を合成して再生する。 193 | 194 | Args: 195 | text (str): 音声合成対象のテキスト。 196 | 197 | """ 198 | wav = self.post_web(text=text) 199 | if wav is not None: 200 | print(f"[Play] {text}") 201 | self.play_wav(wav) 202 | -------------------------------------------------------------------------------- /manual_grpc_publisher_for_gpt.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | 6 | import grpc 7 | 8 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 9 | import gpt_server_pb2 10 | import gpt_server_pb2_grpc 11 | import voice_server_pb2 12 | import voice_server_pb2_grpc 13 | 14 | 15 | def main() -> None: 16 | global enable_input 17 | parser = argparse.ArgumentParser() 18 | parser.add_argument( 19 | "--gpt_ip", help="Gpt server ip address", default="127.0.0.1", type=str 20 | ) 21 | parser.add_argument( 22 | "--gpt_port", help="Gpt server port number", default="10001", type=str 23 | ) 24 | parser.add_argument( 25 | "--voice_ip", help="Voice server ip address", default="127.0.0.1", type=str 26 | ) 27 | parser.add_argument( 28 | "--voice_port", help="Voice server port number", default="10002", type=str 29 | ) 30 | parser.add_argument( 31 | "--no_motion", 32 | help="Not play nod motion", 33 | action="store_true", 34 | ) 35 | args = parser.parse_args() 36 | # grpc stubの設定 37 | gpt_channel = grpc.insecure_channel(args.gpt_ip + ":" + args.gpt_port) 38 | gpt_stub = gpt_server_pb2_grpc.GptServerServiceStub(gpt_channel) 39 | voice_channel = grpc.insecure_channel(args.voice_ip + ":" + args.voice_port) 40 | voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 41 | 42 | while True: 43 | print("文章をキーボード入力後、Enterを押してください。") 44 | text = input("Input: ") 45 | # userメッセージの追加 46 | print(f"User : {text}") 47 | try: 48 | voice_stub.EnableVoicePlay(voice_server_pb2.EnableVoicePlayRequest()) 49 | except BaseException: 50 | pass 51 | try: 52 | gpt_stub.SetGpt(gpt_server_pb2.SetGptRequest(text=text, is_finish=True)) 53 | except BaseException: 54 | print("SetGpt error") 55 | pass 56 | 57 | 58 | if __name__ == "__main__": 59 | main() 60 | -------------------------------------------------------------------------------- /manual_grpc_publisher_for_voice.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | 6 | import grpc 7 | 8 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 9 | import voice_server_pb2 10 | import voice_server_pb2_grpc 11 | 12 | 13 | def main() -> None: 14 | global enable_input 15 | parser = argparse.ArgumentParser() 16 | parser.add_argument( 17 | "--voice_ip", help="Voice server ip address", default="127.0.0.1", type=str 18 | ) 19 | parser.add_argument( 20 | "--voice_port", help="Voice server port number", default="10002", type=str 21 | ) 22 | parser.add_argument( 23 | "--no_motion", 24 | help="Not play nod motion", 25 | action="store_true", 26 | ) 27 | args = parser.parse_args() 28 | # grpc stubの設定 29 | voice_channel = grpc.insecure_channel(args.voice_ip + ":" + args.voice_port) 30 | voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 31 | 32 | while True: 33 | print("文章をキーボード入力後、Enterを押してください。") 34 | text = input("Input: ") 35 | # userメッセージの追加 36 | print(f"User : {text}") 37 | try: 38 | voice_stub.EnableVoicePlay(voice_server_pb2.EnableVoicePlayRequest()) 39 | voice_stub.SetText(voice_server_pb2.SetTextRequest(text=text)) 40 | voice_stub.SentenceEnd(voice_server_pb2.SentenceEndRequest()) 41 | except BaseException: 42 | pass 43 | 44 | 45 | if __name__ == "__main__": 46 | main() 47 | -------------------------------------------------------------------------------- /proto/codegen.py: -------------------------------------------------------------------------------- 1 | from grpc.tools import protoc 2 | 3 | protoc.main( 4 | ( 5 | "", 6 | "-I.", 7 | "--python_out=../lib/grpc", 8 | "--grpc_python_out=../lib/grpc", 9 | "speech_server.proto", 10 | ) 11 | ) 12 | protoc.main( 13 | ( 14 | "", 15 | "-I.", 16 | "--python_out=../lib/grpc", 17 | "--grpc_python_out=../lib/grpc", 18 | "gpt_server.proto", 19 | ) 20 | ) 21 | protoc.main( 22 | ( 23 | "", 24 | "-I.", 25 | "--python_out=../lib/grpc", 26 | "--grpc_python_out=../lib/grpc", 27 | "voice_server.proto", 28 | ) 29 | ) 30 | -------------------------------------------------------------------------------- /proto/gpt_server.proto: -------------------------------------------------------------------------------- 1 | syntax = "proto3"; 2 | package gpt_server; 3 | 4 | message SetGptRequest { 5 | string text = 1; 6 | optional bool is_finish =2; 7 | } 8 | 9 | message SetGptReply { 10 | bool success =1; 11 | } 12 | 13 | message InterruptGptRequest {} 14 | 15 | message InterruptGptReply { 16 | bool success =1; 17 | } 18 | 19 | message SendMotionRequest {} 20 | 21 | message SendMotionReply { 22 | bool success =1; 23 | } 24 | 25 | service GptServerService { 26 | rpc SetGpt(SetGptRequest) 27 | returns (SetGptReply); 28 | rpc InterruptGpt(InterruptGptRequest) 29 | returns (InterruptGptReply); 30 | rpc SendMotion(SendMotionRequest) 31 | returns (SendMotionReply); 32 | } 33 | -------------------------------------------------------------------------------- /proto/speech_server.proto: -------------------------------------------------------------------------------- 1 | syntax = "proto3"; 2 | package speech_server; 3 | 4 | message ToggleSpeechRequest { 5 | bool enable =1; 6 | } 7 | 8 | message ToggleSpeechReply { 9 | bool success =1; 10 | } 11 | 12 | service SpeechServerService { 13 | rpc ToggleSpeech(ToggleSpeechRequest) 14 | returns (ToggleSpeechReply); 15 | } 16 | -------------------------------------------------------------------------------- /proto/voice_server.proto: -------------------------------------------------------------------------------- 1 | syntax = "proto3"; 2 | package voice_server; 3 | 4 | message SetTextRequest { 5 | string text = 1; 6 | } 7 | 8 | message SetTextReply { 9 | bool success =1; 10 | } 11 | 12 | message SetStyleBertVitsParamRequest { 13 | optional string model_name = 1; 14 | optional int32 model_id = 2; 15 | optional float length = 3; 16 | optional string style = 4; 17 | optional float style_weight = 5; 18 | } 19 | 20 | message SetStyleBertVitsParamReply { 21 | bool success =1; 22 | } 23 | 24 | message SetVoicevoxParamRequest { 25 | optional int32 speaker = 1; 26 | optional float speed_scale = 2; 27 | } 28 | 29 | message SetVoicevoxParamReply { 30 | bool success =1; 31 | } 32 | 33 | message SetAivisParamRequest { 34 | optional string speaker = 1; 35 | optional string style = 2; 36 | optional float speed_scale = 3; 37 | } 38 | 39 | message SetAivisParamReply { 40 | bool success =1; 41 | } 42 | 43 | message InterruptVoiceRequest {} 44 | 45 | message InterruptVoiceReply { 46 | bool success =1; 47 | } 48 | 49 | message EnableVoicePlayRequest { 50 | } 51 | 52 | message EnableVoicePlayReply { 53 | bool success =1; 54 | } 55 | 56 | message DisableVoicePlayRequest { 57 | } 58 | 59 | message DisableVoicePlayReply { 60 | bool success =1; 61 | } 62 | 63 | message IsVoicePlayingRequest {} 64 | 65 | message IsVoicePlayingReply { 66 | bool is_playing =1; 67 | } 68 | 69 | message SentenceEndRequest {} 70 | 71 | message SentenceEndReply { 72 | bool success =1; 73 | } 74 | 75 | message StartHeadControlRequest {} 76 | 77 | message StartHeadControlReply { 78 | bool success =1; 79 | } 80 | 81 | 82 | service VoiceServerService { 83 | rpc SetText(SetTextRequest) 84 | returns (SetTextReply); 85 | rpc SetStyleBertVitsParam(SetStyleBertVitsParamRequest) 86 | returns (SetStyleBertVitsParamReply); 87 | rpc SetVoicevoxParam(SetVoicevoxParamRequest) 88 | returns (SetVoicevoxParamReply); 89 | rpc SetAivisParam(SetAivisParamRequest) 90 | returns (SetAivisParamReply); 91 | rpc InterruptVoice(InterruptVoiceRequest) 92 | returns (InterruptVoiceReply); 93 | rpc EnableVoicePlay(EnableVoicePlayRequest) 94 | returns (EnableVoicePlayReply); 95 | rpc DisableVoicePlay(DisableVoicePlayRequest) 96 | returns (DisableVoicePlayReply); 97 | rpc IsVoicePlaying(IsVoicePlayingRequest) 98 | returns (IsVoicePlayingReply); 99 | rpc SentenceEnd(SentenceEndRequest) 100 | returns (SentenceEndReply); 101 | rpc StartHeadControl(StartHeadControlRequest) 102 | returns (StartHeadControlReply); 103 | } 104 | -------------------------------------------------------------------------------- /pysen.toml: -------------------------------------------------------------------------------- 1 | [tool.pysen] 2 | version = "0.10" 3 | 4 | [tool.pysen.lint] 5 | enable_black = true 6 | enable_flake8 = true 7 | enable_isort = true 8 | enable_mypy = true 9 | mypy_preset = "strict" 10 | line_length = 88 11 | py_version = "py38" 12 | mypy_ignore_packages = ["akari_proto.*"] 13 | 14 | [[tool.pysen.lint.mypy_targets]] 15 | paths = [".", "lib/"] 16 | 17 | [tool.pysen.lint.source] 18 | includes = [".", "lib/"] 19 | excludes = [] 20 | exclude_globs = [ 21 | "lib/grpc/", 22 | ] 23 | -------------------------------------------------------------------------------- /requirements.txt: -------------------------------------------------------------------------------- 1 | akari-client[depthai] 2 | akari-proto 3 | alkana 4 | anthropic 5 | google-cloud-speech 6 | google-genai 7 | google-generativeai 8 | -e gpt-stream-json-parser/ 9 | grpcio 10 | grpcio-tools 11 | openai 12 | numpy 13 | pydantic>=2.0.0 14 | PyAudio 15 | PyJapanglish 16 | python-dotenv 17 | six 18 | SpeechRecognition 19 | -------------------------------------------------------------------------------- /script/faster_chatbot.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="voicevox_server" -- bash -ic "python3 voicevox_server.py --voicevox_local --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8" 28 | ) 29 | -------------------------------------------------------------------------------- /script/faster_chatbot_aivis.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="aivis_server" -- bash -ic "python3 aivis_server.py --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8" 28 | ) 29 | -------------------------------------------------------------------------------- /script/faster_chatbot_aivis_auto.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="aivis_server" -- bash -ic "python3 aivis_server.py --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8 --auto" 28 | gnome-terminal --title="talk_controller" -- bash -ic "python3 talk_controller_client.py" 29 | ) 30 | -------------------------------------------------------------------------------- /script/faster_chatbot_auto.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="voicevox_server" -- bash -ic "python3 voicevox_server.py --voicevox_local --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8 --auto" 28 | gnome-terminal --title="talk_controller" -- bash -ic "python3 talk_controller_client.py" 29 | ) 30 | -------------------------------------------------------------------------------- /script/faster_chatbot_bert_vits.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="style_bert_vits_server" -- bash -ic "python3 style_bert_vits_server.py --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8" 28 | ) 29 | -------------------------------------------------------------------------------- /script/faster_chatbot_bert_vits_auto.sh: -------------------------------------------------------------------------------- 1 | #!/bin/bash 2 | # -*- coding: utf-8 -*- 3 | ## シェルオプション 4 | set -e # コマンド実行に失敗したらエラー 5 | set -u # 未定義の変数にアクセスしたらエラー 6 | set -o pipefail # パイプのコマンドが失敗したらエラー（bashのみ） 7 | 8 | ip=$1 9 | 10 | echo ${ip} 11 | 12 | #第２引数でakari_motion_serverのパスが記載されていた場合は、そちらも起動する。 13 | if [ "$#" -ge 2 ]; then 14 | ( 15 | cd $2 16 | . venv/bin/activate 17 | gnome-terminal --title="motion_server" -- bash -ic "python3 server.py" 18 | ) 19 | fi 20 | 21 | 22 | ( 23 | cd ../ 24 | . venv/bin/activate 25 | gnome-terminal --title="style_bert_vits_server" -- bash -ic "python3 style_bert_vits_server.py --voice_host ${ip}" 26 | gnome-terminal --title="gpt_publisher" -- bash -ic "python3 gpt_publisher.py" 27 | gnome-terminal --title="speech_publisher" -- bash -ic "python3 speech_publisher.py --timeout 0.8 --auto" 28 | gnome-terminal --title="talk_controller" -- bash -ic "python3 talk_controller_client.py" 29 | ) 30 | -------------------------------------------------------------------------------- /speech_publisher.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | from concurrent import futures 6 | 7 | import grpc 8 | from lib.google_speech import get_db_thresh 9 | 10 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 11 | import motion_server_pb2 12 | import motion_server_pb2_grpc 13 | import speech_server_pb2 14 | import speech_server_pb2_grpc 15 | import voice_server_pb2 16 | import voice_server_pb2_grpc 17 | 18 | RATE = 16000 19 | CHUNK = int(RATE / 10) # 100ms 20 | POWER_THRESH_DIFF = 20 # 周辺音量にこの値を足したものをpower_threshouldとする 21 | enable_input = True 22 | 23 | 24 | class SpeechServer(speech_server_pb2_grpc.SpeechServerServiceServicer): 25 | """ 26 | 音声入力の制御用のgRPCサーバ 27 | """ 28 | 29 | def ToggleSpeech( 30 | self, 31 | request: speech_server_pb2.ToggleSpeechRequest, 32 | context: grpc.ServicerContext, 33 | ) -> speech_server_pb2.ToggleSpeechReply: 34 | global enable_input 35 | enable_input = request.enable 36 | return speech_server_pb2.ToggleSpeechReply(success=True) 37 | 38 | 39 | def main() -> None: 40 | global enable_input 41 | parser = argparse.ArgumentParser() 42 | parser.add_argument( 43 | "--robot_ip", help="Robot ip address", default="127.0.0.1", type=str 44 | ) 45 | parser.add_argument( 46 | "--robot_port", help="Robot port number", default="50055", type=str 47 | ) 48 | parser.add_argument( 49 | "--gpt_ip", help="Gpt server ip address", default="127.0.0.1", type=str 50 | ) 51 | parser.add_argument( 52 | "--gpt_port", help="Gpt server port number", default="10001", type=str 53 | ) 54 | parser.add_argument( 55 | "--voice_ip", help="Voice server ip address", default="127.0.0.1", type=str 56 | ) 57 | parser.add_argument( 58 | "--voice_port", help="Voice server port number", default="10002", type=str 59 | ) 60 | parser.add_argument( 61 | "-t", 62 | "--timeout", 63 | type=float, 64 | default=0.5, 65 | help="Microphone input power timeout", 66 | ) 67 | parser.add_argument( 68 | "-p", 69 | "--power_threshold", 70 | type=float, 71 | default=0, 72 | help="Microphone input power threshold", 73 | ) 74 | parser.add_argument( 75 | "--progress_report_len", 76 | type=int, 77 | default=8, 78 | help="Send the progress of speech recognition if recognition word count over this number ", 79 | ) 80 | parser.add_argument( 81 | "--no_motion", 82 | help="Not play nod motion", 83 | action="store_true", 84 | ) 85 | parser.add_argument( 86 | "--auto", 87 | help="Skip keyboard input for speech recognition", 88 | action="store_true", 89 | ) 90 | parser.add_argument( 91 | "--v2", 92 | action="store_true", 93 | help="Use google speech v2 instead of v1", 94 | ) 95 | args = parser.parse_args() 96 | if args.v2: 97 | from lib.google_speech_v2_grpc import GoogleSpeechV2Grpc as GoogleSpeechGrpc 98 | from lib.google_speech_v2_grpc import ( 99 | MicrophoneStreamV2Grpc as MicrophoneStreamGrpc, 100 | ) 101 | else: 102 | from lib.google_speech_grpc import GoogleSpeechGrpc as GoogleSpeechGrpc 103 | from lib.google_speech_grpc import MicrophoneStreamGrpc as MicrophoneStreamGrpc 104 | motion_server_host = None 105 | motion_server_port = None 106 | if not args.no_motion: 107 | motion_server_host = args.robot_ip 108 | motion_server_port = args.robot_port 109 | timeout: float = args.timeout 110 | power_threshold: float = args.power_threshold 111 | 112 | server = grpc.server(futures.ThreadPoolExecutor(max_workers=10)) 113 | speech_server_pb2_grpc.add_SpeechServerServiceServicer_to_server( 114 | SpeechServer(), server 115 | ) 116 | port = "10003" 117 | server.add_insecure_port("[::]:" + port) 118 | server.start() 119 | print(f"speech_server start. port: {port}") 120 | 121 | # grpc stubの設定 122 | voice_channel = grpc.insecure_channel(args.voice_ip + ":" + args.voice_port) 123 | voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 124 | 125 | google_speech_grpc = GoogleSpeechGrpc( 126 | gpt_host=args.gpt_ip, 127 | gpt_port=args.gpt_port, 128 | voice_host=args.voice_ip, 129 | voice_port=args.voice_port, 130 | ) 131 | # power_threshouldが指定されていない場合、周辺音量を収録し、発話判定閾値を決定 132 | if power_threshold == 0: 133 | power_threshold = get_db_thresh() + POWER_THRESH_DIFF 134 | print(f"power_threshold set to {power_threshold:.3f}db") 135 | 136 | while True: 137 | responses = None 138 | while not enable_input: 139 | time.sleep(0.01) 140 | with MicrophoneStreamGrpc( 141 | rate=RATE, 142 | chunk=CHUNK, 143 | _timeout_thresh=timeout, 144 | _db_thresh=power_threshold, 145 | gpt_host=args.gpt_ip, 146 | gpt_port=args.gpt_port, 147 | voice_host=args.voice_ip, 148 | voice_port=args.voice_port, 149 | motion_server_host=motion_server_host, 150 | motion_server_port=motion_server_port, 151 | ) as stream: 152 | if not args.auto: 153 | print("Enterを入力してから、マイクに話しかけてください") 154 | input() 155 | try: 156 | voice_stub.DisableVoicePlay( 157 | voice_server_pb2.DisableVoicePlayRequest() 158 | ) 159 | except BaseException: 160 | pass 161 | try: 162 | responses = stream.transcribe() 163 | except BaseException: 164 | google_speech_grpc.interrupt() 165 | continue 166 | if responses is not None: 167 | try: 168 | google_speech_grpc.listen_publisher_grpc( 169 | responses, progress_report_len=args.progress_report_len 170 | ) 171 | except BaseException as e: 172 | print(e) 173 | google_speech_grpc.interrupt() 174 | continue 175 | print("") 176 | 177 | 178 | if __name__ == "__main__": 179 | main() 180 | -------------------------------------------------------------------------------- /speech_to_text_example.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | 3 | RATE = 16000 4 | CHUNK = int(RATE / 10) # 100ms 5 | POWER_THRESH_DIFF = 25 # 周辺音量にこの値を足したものをpower_threshouldとする 6 | 7 | 8 | def main() -> None: 9 | global host 10 | global port 11 | parser = argparse.ArgumentParser() 12 | parser.add_argument( 13 | "-t", 14 | "--timeout", 15 | type=float, 16 | default=0.5, 17 | help="Microphone input power timeout", 18 | ) 19 | parser.add_argument( 20 | "-p", 21 | "--power_threshold", 22 | type=float, 23 | default=0, 24 | help="Microphone input power threshold", 25 | ) 26 | parser.add_argument( 27 | "--v2", 28 | action="store_true", 29 | help="Use google speech v2 instead of v1", 30 | ) 31 | args = parser.parse_args() 32 | if args.v2: 33 | from lib.google_speech_v2 import MicrophoneStreamV2 as MicrophoneStream 34 | from lib.google_speech_v2 import get_db_thresh, listen_print_loop 35 | else: 36 | from lib.google_speech import MicrophoneStream, get_db_thresh, listen_print_loop 37 | timeout: float = args.timeout 38 | power_threshold: float = args.power_threshold 39 | if power_threshold == 0: 40 | power_threshold = get_db_thresh() + POWER_THRESH_DIFF 41 | print(f"power_threshold set to {power_threshold:.3f}db") 42 | 43 | print("マイクに話しかけてください") 44 | while True: 45 | responses = None 46 | with MicrophoneStream( 47 | rate=RATE, chunk=CHUNK, _timeout_thresh=timeout, _db_thresh=power_threshold 48 | ) as stream: 49 | responses = stream.transcribe() 50 | if responses is not None: 51 | listen_print_loop(responses) 52 | 53 | 54 | if __name__ == "__main__": 55 | main() 56 | -------------------------------------------------------------------------------- /style_bert_vits_example.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | 3 | from lib.style_bert_vits import TextToStyleBertVits 4 | 5 | 6 | def main() -> None: 7 | host = "" 8 | port = "" 9 | parser = argparse.ArgumentParser() 10 | parser.add_argument( 11 | "--voice_host", 12 | type=str, 13 | default="127.0.0.1", 14 | help="Voice server host", 15 | ) 16 | parser.add_argument( 17 | "--voice_port", 18 | type=str, 19 | default="5000", 20 | help="Voice server port", 21 | ) 22 | args = parser.parse_args() 23 | host = args.voice_host 24 | port = args.voice_port 25 | text_to_voice = TextToStyleBertVits(host, port) 26 | 27 | # set_paramメソッドでモデル名や音声再生速度、感情スタイルなどを指定することができます。 28 | # モデル名を指定 29 | # text_to_voice.set_param(model_name='jvnv-F1-jp') 30 | # 音声再生速度を指定 31 | # text_to_voice.set_param(length=2.0) 32 | # 感情スタイルを指定 33 | # text_to_voice.set_param(style='Happy') 34 | # 感情スタイルの重みを指定 35 | # text_to_voice.set_param(style_weight=3.0) 36 | 37 | print("発話させたい文章をキーボード入力後、Enterを押してください。") 38 | while True: 39 | text = input("Input: ") 40 | text_to_voice.put_text( 41 | text=text, 42 | ) 43 | print("") 44 | 45 | 46 | if __name__ == "__main__": 47 | main() 48 | -------------------------------------------------------------------------------- /style_bert_vits_server.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | from concurrent import futures 6 | from typing import Any 7 | 8 | import grpc 9 | from lib.style_bert_vits import TextToStyleBertVits 10 | 11 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 12 | import voice_server_pb2 13 | import voice_server_pb2_grpc 14 | 15 | 16 | class VoiceServer(voice_server_pb2_grpc.VoiceServerServiceServicer): 17 | """ 18 | StyleBertVitsにtextを送信し、音声を再生するgprcサーバ 19 | """ 20 | 21 | def __init__(self, text_to_voice: Any) -> None: 22 | self.text_to_voice = text_to_voice 23 | 24 | def SetText( 25 | self, 26 | request: voice_server_pb2.SetTextRequest(), 27 | context: grpc.ServicerContext, 28 | ) -> voice_server_pb2.SetTextReply: 29 | # 即時再生しないようにis_playはFalseで実行 30 | print(f"Send text: {request.text}") 31 | self.text_to_voice.put_text(request.text, play_now=False) 32 | return voice_server_pb2.SetTextReply(success=True) 33 | 34 | def SetStyleBertVitsParam( 35 | self, 36 | request: voice_server_pb2.SetStyleBertVitsParamRequest(), 37 | context: grpc.ServicerContext, 38 | ) -> voice_server_pb2.SetStyleBertVitsParamReply: 39 | if request.model_name: 40 | self.text_to_voice.set_param(model_name=request.model_name) 41 | if request.length: 42 | self.text_to_voice.set_param(length=request.length) 43 | if request.style: 44 | self.text_to_voice.set_param(style=request.style) 45 | if request.style_weight: 46 | self.text_to_voice.set_param(style_weight=request.style_weight) 47 | return voice_server_pb2.SetStyleBertVitsParamReply(success=True) 48 | 49 | def SetVoicevoxParam( 50 | self, 51 | request: voice_server_pb2.SetVoicevoxParamRequest(), 52 | context: grpc.ServicerContext, 53 | ) -> voice_server_pb2.SetVoicevoxParamReply: 54 | print("SetVoicevoxParam is not supported on style_bert_vits_server.") 55 | return voice_server_pb2.SetVoicevoxParamReply(success=False) 56 | 57 | def SetAivisParam( 58 | self, 59 | request: voice_server_pb2.SetAivisParamRequest(), 60 | context: grpc.ServicerContext, 61 | ) -> voice_server_pb2.SetAivisParamReply: 62 | print("SetAivisParam is not supported on style_bert_vits_server.") 63 | return voice_server_pb2.SetAivisParamReply(success=False) 64 | 65 | def InterruptVoice( 66 | self, 67 | request: voice_server_pb2.InterruptVoiceRequest(), 68 | context: grpc.ServicerContext, 69 | ) -> voice_server_pb2.InterruptVoiceReply: 70 | while not self.text_to_voice.queue.empty(): 71 | self.text_to_voice.queue.get() 72 | return voice_server_pb2.InterruptVoiceReply(success=True) 73 | 74 | def EnableVoicePlay( 75 | self, 76 | request: voice_server_pb2.EnableVoicePlayRequest(), 77 | context: grpc.ServicerContext, 78 | ) -> voice_server_pb2.EnableVoicePlayReply: 79 | self.text_to_voice.enable_voice_play() 80 | return voice_server_pb2.EnableVoicePlayReply(success=True) 81 | 82 | def DisableVoicePlay( 83 | self, 84 | request: voice_server_pb2.DisableVoicePlayRequest(), 85 | context: grpc.ServicerContext, 86 | ) -> voice_server_pb2.DisableVoicePlayReply: 87 | self.text_to_voice.disable_voice_play() 88 | return voice_server_pb2.DisableVoicePlayReply(success=True) 89 | 90 | def IsVoicePlaying( 91 | self, 92 | request: voice_server_pb2.IsVoicePlayingRequest(), 93 | context: grpc.ServicerContext, 94 | ) -> voice_server_pb2.IsVoicePlayingReply: 95 | return voice_server_pb2.IsVoicePlayingReply( 96 | is_playing=not self.text_to_voice.is_playing() 97 | ) 98 | 99 | def SentenceEnd( 100 | self, 101 | request: voice_server_pb2.SentenceEndRequest(), 102 | context: grpc.ServicerContext, 103 | ) -> voice_server_pb2.SentenceEndReply: 104 | self.text_to_voice.sentence_end() 105 | return voice_server_pb2.SentenceEndReply(success=True) 106 | 107 | def StartHeadControl( 108 | self, 109 | request: voice_server_pb2.StartHeadControlRequest(), 110 | context: grpc.ServicerContext, 111 | ) -> voice_server_pb2.StartHeadControlReply: 112 | self.text_to_voice.start_head_control() 113 | return voice_server_pb2.StartHeadControlReply(success=False) 114 | 115 | 116 | def main() -> None: 117 | parser = argparse.ArgumentParser() 118 | parser.add_argument( 119 | "--voice_host", 120 | type=str, 121 | default="127.0.0.1", 122 | help="Style-Bert-VITS2 server host", 123 | ) 124 | parser.add_argument( 125 | "--voice_port", 126 | type=str, 127 | default="5000", 128 | help="Style-Bert-VITS2 server port", 129 | ) 130 | parser.add_argument( 131 | "--robot_ip", help="Robot ip address", default="127.0.0.1", type=str 132 | ) 133 | parser.add_argument( 134 | "--robot_port", help="Robot port number", default="50055", type=str 135 | ) 136 | parser.add_argument( 137 | "--no_motion", 138 | help="Not play nod motion", 139 | action="store_true", 140 | ) 141 | args = parser.parse_args() 142 | 143 | host = args.voice_host 144 | port = args.voice_port 145 | motion_server_host = None 146 | motion_server_port = None 147 | if not args.no_motion: 148 | motion_server_host = args.robot_ip 149 | motion_server_port = args.robot_port 150 | text_to_voice = TextToStyleBertVits( 151 | host=host, 152 | port=port, 153 | motion_host=motion_server_host, 154 | motion_port=motion_server_port, 155 | ) 156 | 157 | server = grpc.server(futures.ThreadPoolExecutor(max_workers=10)) 158 | voice_server_pb2_grpc.add_VoiceServerServiceServicer_to_server( 159 | VoiceServer(text_to_voice), server 160 | ) 161 | port = "10002" 162 | server.add_insecure_port("[::]:" + port) 163 | server.start() 164 | print(f"voice_server start. port: {port}") 165 | server.wait_for_termination() 166 | 167 | 168 | if __name__ == "__main__": 169 | main() 170 | -------------------------------------------------------------------------------- /talk_controller_client.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | 6 | import grpc 7 | 8 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 9 | import speech_server_pb2 10 | import speech_server_pb2_grpc 11 | import voice_server_pb2 12 | import voice_server_pb2_grpc 13 | 14 | 15 | def main() -> None: 16 | global enable_input 17 | parser = argparse.ArgumentParser() 18 | parser.add_argument( 19 | "--speech_ip", help="speech publisher ip address", default="127.0.0.1", type=str 20 | ) 21 | parser.add_argument( 22 | "--speech_port", help="speech publisher port number", default="10003", type=str 23 | ) 24 | parser.add_argument( 25 | "--voice_ip", help="Voice server ip address", default="127.0.0.1", type=str 26 | ) 27 | parser.add_argument( 28 | "--voice_port", help="Voice server port number", default="10002", type=str 29 | ) 30 | args = parser.parse_args() 31 | 32 | # grpc stubの設定 33 | speech_channel = grpc.insecure_channel(args.speech_ip + ":" + str(args.speech_port)) 34 | voice_channel = grpc.insecure_channel(args.voice_ip + ":" + args.voice_port) 35 | voice_stub = None 36 | speech_stub = None 37 | # Voice serverの接続確認 38 | while True: 39 | try: 40 | grpc.channel_ready_future(voice_channel).result(timeout=0.5) 41 | voice_stub = voice_server_pb2_grpc.VoiceServerServiceStub(voice_channel) 42 | break 43 | except grpc.FutureTimeoutError: 44 | print("Connecting to voice server timeout. Retrying") 45 | continue 46 | except KeyboardInterrupt: 47 | return 48 | except BaseException as e: 49 | print(f"RPC error: {e}") 50 | continue 51 | print("Connected to voice server") 52 | # Speech serverの接続確認 53 | while True: 54 | try: 55 | grpc.channel_ready_future(speech_channel).result(timeout=0.5) 56 | speech_stub = speech_server_pb2_grpc.SpeechServerServiceStub(speech_channel) 57 | break 58 | except grpc.FutureTimeoutError: 59 | print("Connecting to speech server timeout. Retrying") 60 | continue 61 | except KeyboardInterrupt: 62 | return 63 | except BaseException as e: 64 | print(f"RPC error: {e}") 65 | continue 66 | print("Connected to speech server") 67 | is_voice_playing = False 68 | 69 | while True: 70 | if not is_voice_playing: 71 | try: 72 | ret = voice_stub.IsVoicePlaying( 73 | voice_server_pb2.IsVoicePlayingRequest() 74 | ) 75 | is_voice_playing = ret.is_playing 76 | except KeyboardInterrupt: 77 | return 78 | except BaseException: 79 | print("Voice server connection error!") 80 | if is_voice_playing: 81 | speech_stub.ToggleSpeech( 82 | speech_server_pb2.ToggleSpeechRequest(enable=False) 83 | ) 84 | else: 85 | try: 86 | ret = voice_stub.IsVoicePlaying( 87 | voice_server_pb2.IsVoicePlayingRequest() 88 | ) 89 | is_voice_playing = ret.is_playing 90 | except KeyboardInterrupt: 91 | return 92 | except BaseException: 93 | print("Voice server connection error!") 94 | if not is_voice_playing: 95 | speech_stub.ToggleSpeech( 96 | speech_server_pb2.ToggleSpeechRequest(enable=True) 97 | ) 98 | time.sleep(0.1) 99 | 100 | 101 | if __name__ == "__main__": 102 | main() 103 | -------------------------------------------------------------------------------- /text_to_kana_example.py: -------------------------------------------------------------------------------- 1 | from lib.en_to_jp import EnToJp 2 | 3 | if __name__ == "__main__": 4 | en_to_jp = EnToJp() 5 | while True: 6 | text = input("文章中の英単語をカタカナに変換します。文章を入力してください。\n> ") 7 | print("alkana Only") 8 | print(f" {en_to_jp.text_to_kana(text, True,False)}") 9 | print("japanglish Only") 10 | print(f" {en_to_jp.text_to_kana(text, False,True, False)}") 11 | print("japanglish inference Only") 12 | print(f" {en_to_jp.text_to_kana(text, False,True, True)}") 13 | print("alkana japanglish no inference") 14 | print(f" {en_to_jp.text_to_kana(text, True,True,False)}") 15 | print("alkana japanglish inference") 16 | print(f" {en_to_jp.text_to_kana(text, True,True,True)}") 17 | -------------------------------------------------------------------------------- /voicevox_example.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | 3 | 4 | def main() -> None: 5 | host = "" 6 | port = "" 7 | parser = argparse.ArgumentParser() 8 | parser.add_argument("--voicevox_local", action="store_true") 9 | parser.add_argument( 10 | "--voice_host", 11 | type=str, 12 | default="127.0.0.1", 13 | help="VoiceVox server host", 14 | ) 15 | parser.add_argument( 16 | "--voice_port", 17 | type=str, 18 | default="50021", 19 | help="VoiceVox server port", 20 | ) 21 | args = parser.parse_args() 22 | if args.voicevox_local: 23 | from lib.voicevox import TextToVoiceVox 24 | 25 | host = args.voice_host 26 | port = args.voice_port 27 | text_to_voice = TextToVoiceVox(host, port) 28 | print("voicevox local pc ver.") 29 | else: 30 | from lib.conf import VOICEVOX_APIKEY 31 | from lib.voicevox import TextToVoiceVoxWeb 32 | 33 | text_to_voice = TextToVoiceVoxWeb(apikey=VOICEVOX_APIKEY) 34 | print("voicevox web ver.") 35 | 36 | print("発話させたい文章をキーボード入力後、Enterを押してください。") 37 | while True: 38 | text = input("Input: ") 39 | text_to_voice.put_text(text) 40 | print("") 41 | 42 | 43 | if __name__ == "__main__": 44 | main() 45 | -------------------------------------------------------------------------------- /voicevox_server.py: -------------------------------------------------------------------------------- 1 | import argparse 2 | import os 3 | import sys 4 | import time 5 | from concurrent import futures 6 | from typing import Any 7 | 8 | import grpc 9 | 10 | sys.path.append(os.path.join(os.path.dirname(__file__), "lib/grpc")) 11 | import voice_server_pb2 12 | import voice_server_pb2_grpc 13 | 14 | 15 | class VoiceServer(voice_server_pb2_grpc.VoiceServerServiceServicer): 16 | """ 17 | Voicevoxにtextを送信し、音声を再生するgprcサーバ 18 | """ 19 | 20 | def __init__(self, text_to_voice: Any) -> None: 21 | self.text_to_voice = text_to_voice 22 | 23 | def SetText( 24 | self, 25 | request: voice_server_pb2.SetTextRequest(), 26 | context: grpc.ServicerContext, 27 | ) -> voice_server_pb2.SetTextReply: 28 | # 即時再生しないようにis_playはFalseで実行 29 | print(f"Send text: {request.text}") 30 | self.text_to_voice.put_text(request.text, play_now=False) 31 | return voice_server_pb2.SetTextReply(success=True) 32 | 33 | def SetStyleBertVitsParam( 34 | self, 35 | request: voice_server_pb2.SetStyleBertVitsParamRequest(), 36 | context: grpc.ServicerContext, 37 | ) -> voice_server_pb2.SetStyleBertVitsParamReply: 38 | print("SetStyleBertVitsParam is not supported on voicevox_server.") 39 | return voice_server_pb2.SetStyleBertVitsParamReply(success=False) 40 | 41 | def SetVoicevoxParam( 42 | self, 43 | request: voice_server_pb2.SetVoicevoxParamRequest(), 44 | context: grpc.ServicerContext, 45 | ) -> voice_server_pb2.SetVoicevoxParamReply: 46 | if request.speaker: 47 | self.text_to_voice.set_param(speaker=request.speaker) 48 | if request.speed_scale: 49 | self.text_to_voice.set_param(speed_scale=request.speed_scale) 50 | return voice_server_pb2.SetVoicevoxParamReply(success=True) 51 | 52 | def SetAivisParam( 53 | self, 54 | request: voice_server_pb2.SetAivisParamRequest(), 55 | context: grpc.ServicerContext, 56 | ) -> voice_server_pb2.SetAivisParamReply: 57 | print("SetAivisParam is not supported on voicevox_server.") 58 | return voice_server_pb2.SetAivisParamReply(success=False) 59 | 60 | def InterruptVoice( 61 | self, 62 | request: voice_server_pb2.InterruptVoiceRequest(), 63 | context: grpc.ServicerContext, 64 | ) -> voice_server_pb2.InterruptVoiceReply: 65 | while not self.text_to_voice.queue.empty(): 66 | self.text_to_voice.queue.get() 67 | return voice_server_pb2.InterruptVoiceReply(success=True) 68 | 69 | def EnableVoicePlay( 70 | self, 71 | request: voice_server_pb2.EnableVoicePlayRequest(), 72 | context: grpc.ServicerContext, 73 | ) -> voice_server_pb2.EnableVoicePlayReply: 74 | self.text_to_voice.enable_voice_play() 75 | return voice_server_pb2.EnableVoicePlayReply(success=True) 76 | 77 | def DisableVoicePlay( 78 | self, 79 | request: voice_server_pb2.DisableVoicePlayRequest(), 80 | context: grpc.ServicerContext, 81 | ) -> voice_server_pb2.DisableVoicePlayReply: 82 | self.text_to_voice.disable_voice_play() 83 | return voice_server_pb2.DisableVoicePlayReply(success=True) 84 | 85 | def IsVoicePlaying( 86 | self, 87 | request: voice_server_pb2.IsVoicePlayingRequest(), 88 | context: grpc.ServicerContext, 89 | ) -> voice_server_pb2.IsVoicePlayingReply: 90 | return voice_server_pb2.IsVoicePlayingReply( 91 | is_playing=not self.text_to_voice.is_playing() 92 | ) 93 | 94 | def SentenceEnd( 95 | self, 96 | request: voice_server_pb2.SentenceEndRequest(), 97 | context: grpc.ServicerContext, 98 | ) -> voice_server_pb2.SentenceEndReply: 99 | self.text_to_voice.sentence_end() 100 | return voice_server_pb2.SentenceEndReply(success=True) 101 | 102 | def StartHeadControl( 103 | self, 104 | request: voice_server_pb2.StartHeadControlRequest(), 105 | context: grpc.ServicerContext, 106 | ) -> voice_server_pb2.StartHeadControlReply: 107 | self.text_to_voice.start_head_control() 108 | return voice_server_pb2.StartHeadControlReply(success=False) 109 | 110 | 111 | def main() -> None: 112 | parser = argparse.ArgumentParser() 113 | parser.add_argument("--voicevox_local", action="store_true") 114 | parser.add_argument( 115 | "--voice_host", 116 | type=str, 117 | default="127.0.0.1", 118 | help="VoiceVox server host", 119 | ) 120 | parser.add_argument( 121 | "--voice_port", 122 | type=str, 123 | default="50021", 124 | help="VoiceVox server port", 125 | ) 126 | parser.add_argument( 127 | "--robot_ip", help="Robot ip address", default="127.0.0.1", type=str 128 | ) 129 | parser.add_argument( 130 | "--robot_port", help="Robot port number", default="50055", type=str 131 | ) 132 | parser.add_argument( 133 | "--no_motion", 134 | help="Not play nod motion", 135 | action="store_true", 136 | ) 137 | args = parser.parse_args() 138 | motion_server_host = None 139 | motion_server_port = None 140 | if not args.no_motion: 141 | motion_server_host = args.robot_ip 142 | motion_server_port = args.robot_port 143 | if args.voicevox_local: 144 | # local版の場合 145 | from lib.voicevox import TextToVoiceVox 146 | 147 | text_to_voice = TextToVoiceVox( 148 | host=args.voice_host, 149 | port=args.voice_port, 150 | motion_host=motion_server_host, 151 | motion_port=motion_server_port, 152 | ) 153 | print("voicevox local pc ver.") 154 | else: 155 | # web版の場合 156 | from lib.conf import VOICEVOX_APIKEY 157 | from lib.voicevox import TextToVoiceVoxWeb 158 | 159 | text_to_voice = TextToVoiceVoxWeb( 160 | apikey=VOICEVOX_APIKEY, 161 | motion_host=motion_server_host, 162 | motion_port=motion_server_port, 163 | ) 164 | print("voicevox web ver.") 165 | 166 | server = grpc.server(futures.ThreadPoolExecutor(max_workers=10)) 167 | voice_server_pb2_grpc.add_VoiceServerServiceServicer_to_server( 168 | VoiceServer(text_to_voice), server 169 | ) 170 | port = "10002" 171 | server.add_insecure_port("[::]:" + port) 172 | server.start() 173 | print(f"voice_server start. port: {port}") 174 | server.wait_for_termination() 175 | 176 | 177 | if __name__ == "__main__": 178 | main() 179 | --------------------------------------------------------------------------------