mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-22 07:03:53 +08:00
add:支持Gizwits API渠道接口
This commit is contained in:
jzhuang
@@ -201,6 +201,7 @@ server:
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| TTS | CosyVoiceSiliconflow | 接口调用 | 消耗 token | 需申请硅基流动 API 密钥；输出格式为 wav |
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| TTS | TTS302AI | 接口调用 | 消耗 token | [点击创建密钥](https://dash.302.ai/apis/list) |
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| TTS | CozeCnTTS | 接口调用 | 消耗 token | 需提供 Coze API key；输出格式为 wav |
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| TTS | GizwitsTTS | 接口调用 | 消耗 token | [点击创建密钥](https://agentrouter.gizwitsapi.com) |
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| TTS | ACGNTTS | 接口调用 | 消耗 token | [联系网站管理员购买密钥](www.ttson.cn) |
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| TTS | OpenAITTS | 接口调用 | 消耗 token | 境外使用，境外购买 |
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| TTS | FishSpeech | 接口调用 | 免费/自定义 | 本地启动 TTS 服务；启动方法见配置文件内说明 |
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@@ -220,10 +221,11 @@ server:
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### ASR 语音识别
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| 类型 | 平台名称 | 使用方式 | 收费模式 | 备注 |
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|:---:|:---------:|:----:|:----:|:--:|
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| ASR | FunASR | 本地使用 | 免费 | |
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| ASR | DoubaoASR | 接口调用 | 收费 | |
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| 类型 | 平台名称 | 使用方式 | 收费模式 | 备注 |
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|:---:|:----------:|:----:|:----:|:------------------------------------------------------:|
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| ASR | FunASR | 本地使用 | 免费 | |
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| ASR | DoubaoASR | 接口调用 | 收费 | |
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| ASR | GizwitsASR | 接口调用 | 消耗 token | [点击创建密钥](https://agentrouter.gizwitsapi.com)，支持语音识别大模型 |
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---
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@@ -131,6 +131,13 @@ ASR:
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access_token: 你的火山引擎语音合成服务access_token
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cluster: volcengine_input_common
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output_dir: tmp/
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GizwitsASR:
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type: volcengine
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# 火山引擎作为基座，可以完全使用企业级火山引擎语音识别大模型服务，识别率更准确
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# 获取API Key地址：https://agentrouter.gizwitsapi.com/panel/token
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base_host: bytedance.gizwitsapi.com
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access_token: "你的机智云API key"
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output_dir: tmp/
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VAD:
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SileroVAD:
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threshold: 0.5
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@@ -399,6 +406,15 @@ TTS:
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voice: "zh_female_wanwanxiaohe_moon_bigtts"
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output_dir: tmp/
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access_token: "你的302API密钥"
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GizwitsTTS:
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type: doubao
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# 火山引擎作为基座，可以完全使用企业级火山引擎语音合成服务
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# 获取API Key地址：https://agentrouter.gizwitsapi.com/panel/token
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api_url: https://bytedance.gizwitsapi.com/api/v1/tts
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authorization: "Bearer "
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voice: "zh_female_wanwanxiaohe_moon_bigtts"
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output_file: tmp/
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access_token: "你的机智云API key"
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ACGNTTS:
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#在线网址：https://acgn.ttson.cn/
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#token购买：www.ttson.cn
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@@ -0,0 +1,300 @@
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import time
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import io
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import wave
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import os
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from typing import Optional, Tuple, List
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import uuid
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import websockets
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import json
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import gzip
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import opuslib_next
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from core.utils.util import check_model_key
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from core.providers.asr.base import ASRProviderBase
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from config.logger import setup_logging
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TAG = __name__
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logger = setup_logging()
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PROTOCOL_VERSION = 0b0001
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DEFAULT_HEADER_SIZE = 0b0001
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# Message Type:
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FULL_CLIENT_REQUEST = 0b0001
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AUDIO_ONLY_REQUEST = 0b0010
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FULL_SERVER_RESPONSE = 0b1001
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SERVER_ERROR_RESPONSE = 0b1111
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# Message Type Specific Flags
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FLAG_NO_SEQUENCE = 0b0000
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FLAG_WITH_SEQUENCE = 0b0001
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FLAG_LAST_PACKET = 0b0010
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# Message Serialization
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NO_SERIALIZATION = 0b0000
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JSON = 0b0001
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# Message Compression
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NO_COMPRESSION = 0b0000
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GZIP = 0b0001
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class ASRProvider(ASRProviderBase):
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def __init__(self, config: dict, delete_audio_file: bool):
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self.host = config.get("base_host", "openspeech.bytedance.com")
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self.ws_url = f"wss://{self.host}/api/v3/sauc/bigmodel"
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self.appid = config.get("appid")
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self.access_token = config.get("access_token")
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self.resource_id = config.get("resource_id", "volc.bigasr.sauc.duration")
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check_model_key("ASR", self.access_token)
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self.seg_duration = 15000
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# 确保输出目录存在
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self.output_dir = config.get("output_dir")
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os.makedirs(self.output_dir, exist_ok=True)
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def save_audio_to_file(self, opus_data: List[bytes], session_id: str) -> str:
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"""将Opus音频数据解码并保存为WAV文件"""
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file_name = f"asr_{session_id}_{uuid.uuid4()}.wav"
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file_path = os.path.join(self.output_dir, file_name)
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decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
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pcm_data = []
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for opus_packet in opus_data:
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try:
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pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
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pcm_data.append(pcm_frame)
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except opuslib_next.OpusError as e:
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logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
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with wave.open(file_path, "wb") as wf:
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wf.setnchannels(1)
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wf.setsampwidth(2) # 2 bytes = 16-bit
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wf.setframerate(16000)
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wf.writeframes(b"".join(pcm_data))
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return file_path
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@staticmethod
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def _generate_header(message_type=FULL_CLIENT_REQUEST, message_type_specific_flags=FLAG_NO_SEQUENCE,
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serial_method=JSON, compression_type=GZIP, reserved_data=0x00) -> bytearray:
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"""
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protocol_version(4 bits), header_size(4 bits),
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message_type(4 bits), message_type_specific_flags(4 bits)
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serialization_method(4 bits) message_compression(4 bits)
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reserved(8bits) 保留字段
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"""
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header = bytearray()
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header.append((PROTOCOL_VERSION << 4) | DEFAULT_HEADER_SIZE)
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header.append((message_type << 4) | message_type_specific_flags)
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header.append((serial_method << 4) | compression_type)
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header.append(reserved_data)
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return header
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@staticmethod
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def _construct_request() -> dict:
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"""Construct the request payload."""
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return {
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"user": {
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"uid": str(uuid.uuid4()),
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},
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"audio": {
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"format": "wav",
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"codec": "raw",
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"rate": 16000,
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"bits": 16,
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"channel": 1,
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"language": "zh-CN",
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},
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"request": {
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"model_name": "bigmodel",
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"enable_itn": False,
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"enable_ddc": False,
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"enable_punc": False,
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"show_utterances": False,
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},
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}
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@staticmethod
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def _parse_response(res):
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"""
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protocol_version(4 bits), header_size(4 bits),
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message_type(4 bits), message_type_specific_flags(4 bits)
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serialization_method(4 bits) message_compression(4 bits)
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reserved(8 bits)保留字段
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header_extensions 扩展头(大小等于 8 * 4 * (header_size - 1))
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payload 类似与http 请求体
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"""
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protocol_version = res[0] >> 4
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if protocol_version != PROTOCOL_VERSION:
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return None
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header_size = res[0] & 0x0f
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message_type = res[1] >> 4
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message_type_specific_flags = res[1] & 0x0f
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serialization_method = res[2] >> 4
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message_compression = res[2] & 0x0f
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payload = res[header_size * 4:]
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result = {}
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payload_msg = None
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if message_type == FULL_SERVER_RESPONSE:
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if message_type_specific_flags & FLAG_WITH_SEQUENCE:
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# receive frame with sequence
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result['payload_sequence'] = int.from_bytes(payload[:4], "big", signed=True)
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payload = payload[4:]
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if message_type_specific_flags & FLAG_LAST_PACKET:
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# receive last package
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result['is_last_package'] = True
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payload_size = int.from_bytes(payload[:4], "big", signed=True)
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payload_msg = payload[4:4 + payload_size]
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elif message_type == SERVER_ERROR_RESPONSE:
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result['code'] = int.from_bytes(payload[:4], "big", signed=False)
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payload_size = int.from_bytes(payload[4:8], "big", signed=False)
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payload_msg = payload[8:8 + payload_size]
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if payload_msg is not None:
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if message_compression == GZIP:
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payload_msg = gzip.decompress(payload_msg)
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if serialization_method == JSON:
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payload_msg = json.loads(str(payload_msg, "utf-8"))
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elif serialization_method != NO_SERIALIZATION:
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payload_msg = str(payload_msg, "utf-8")
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result['payload_msg'] = payload_msg
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return result
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async def _send_request(self, audio_data: List[bytes], segment_size: int) -> Optional[str]:
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"""Send request to VolcEngine ASR service."""
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try:
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auth_header = {
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"X-Api-App-Key": self.appid,
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"X-Api-Access-Key": self.access_token,
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"X-Api-Resource-Id": self.resource_id,
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"X-Api-Request-Id": str(uuid.uuid4())
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}
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async with websockets.connect(self.ws_url, additional_headers=auth_header,
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max_size=128 * 1024 * 1024) as websocket:
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sequence = 1
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# Send header and metadata
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request_params = self._construct_request()
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payload_bytes = str.encode(json.dumps(request_params))
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payload_bytes = gzip.compress(payload_bytes)
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full_client_request = self._generate_header(message_type_specific_flags=FLAG_WITH_SEQUENCE)
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full_client_request.extend(sequence.to_bytes(4, 'big', signed=True))
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full_client_request.extend((len(payload_bytes)).to_bytes(4, 'big')) # payload size(4 bytes)
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full_client_request.extend(payload_bytes) # payload
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await websocket.send(full_client_request)
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res = await websocket.recv()
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result = self._parse_response(res)
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if 'code' in result:
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logger.bind(tag=TAG).error(f"ASR error: {result['payload_msg']}")
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return None
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for _, (chunk, last) in enumerate(self.slice_data(audio_data, segment_size), 1):
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sequence += 1
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if last:
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sequence = -sequence # last package
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audio_only_request = self._generate_header(
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message_type=AUDIO_ONLY_REQUEST,
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message_type_specific_flags=FLAG_WITH_SEQUENCE | FLAG_LAST_PACKET
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)
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else:
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audio_only_request = self._generate_header(
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message_type=AUDIO_ONLY_REQUEST,
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message_type_specific_flags=FLAG_WITH_SEQUENCE
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)
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audio_only_request.extend(sequence.to_bytes(4, 'big', signed=True)) # sequence
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payload_bytes = gzip.compress(chunk)
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audio_only_request.extend((len(payload_bytes)).to_bytes(4, 'big')) # payload size(4 bytes)
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audio_only_request.extend(payload_bytes) # payload
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# Send audio data
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await websocket.send(audio_only_request)
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# Receive response
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for _ in range(1, -sequence):
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response = await websocket.recv()
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result = self._parse_response(response)
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if 'code' in result:
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logger.bind(tag=TAG).error(f"ASR error: {result['payload_msg']}")
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return None
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if 'is_last_package' in result and result['is_last_package'] is True:
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return result.get('payload_msg', {}).get('result', {}).get('text')
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raise Exception("not received last package")
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except Exception as e:
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logger.bind(tag=TAG).error(f"ASR request failed: {e}", exc_info=True)
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return None
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@staticmethod
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def decode_opus(opus_data: List[bytes], session_id: str) -> List[bytes]:
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decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
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pcm_data = []
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for opus_packet in opus_data:
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try:
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pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
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pcm_data.append(pcm_frame)
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except opuslib_next.OpusError as e:
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logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
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return pcm_data
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@staticmethod
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def read_wav_info(data: io.BytesIO = None) -> (int, int, int, int, int):
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with io.BytesIO(data) as _f:
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wave_fp = wave.open(_f, 'rb')
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nchannels, sampwidth, framerate, nframes = wave_fp.getparams()[:4]
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wave_bytes = wave_fp.readframes(nframes)
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return nchannels, sampwidth, framerate, nframes, len(wave_bytes)
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@staticmethod
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def slice_data(data: bytes, chunk_size: int) -> (list, bool):
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"""
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slice data
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:param data: wav data
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:param chunk_size: the segment size in one request
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:return: segment data, last flag
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"""
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data_len = len(data)
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offset = 0
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while offset + chunk_size < data_len:
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yield data[offset: offset + chunk_size], False
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offset += chunk_size
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else:
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yield data[offset: data_len], True
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async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
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"""将语音数据转换为文本"""
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try:
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# 合并所有opus数据包
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pcm_data = self.decode_opus(opus_data, session_id)
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combined_pcm_data = b''.join(pcm_data)
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wav_buffer = io.BytesIO()
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with wave.open(wav_buffer, "wb") as wav_file:
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wav_file.setnchannels(1) # 设置声道数
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wav_file.setsampwidth(2) # 设置采样宽度
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wav_file.setframerate(16000) # 设置采样率
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wav_file.writeframes(combined_pcm_data) # 写入 PCM 数据
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# 获取封装后的 WAV 数据
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wav_data = wav_buffer.getvalue()
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nchannels, sampwidth, framerate, nframes, wav_len = self.read_wav_info(wav_data)
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size_per_sec = nchannels * sampwidth * framerate
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segment_size = int(size_per_sec * self.seg_duration / 1000)
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# 语音识别
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start_time = time.time()
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text = await self._send_request(wav_data, segment_size)
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if text:
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logger.bind(tag=TAG).debug(f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}")
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return text, None
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return "", None
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except Exception as e:
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logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
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return "", None
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