Files
ragflow/rag/llm/sequence2txt_model.py

621 lines
24 KiB
Python
Raw Permalink Normal View History

#
# Copyright 2024 The InfiniFlow Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
#
import base64
import io
import json
import os
import re
2026-07-16 10:49:02 +08:00
import struct
from abc import ABC
import tempfile
feat: add FuturMix as model provider (#14419) ## Summary Add [FuturMix](https://futurmix.ai) as a new model provider. FuturMix is an OpenAI-compatible unified AI gateway that provides access to 22+ models (GPT, Claude, Gemini, DeepSeek, and more) through a single API endpoint and key. - **API Base**: `https://futurmix.ai/v1` (OpenAI-compatible) - **Supported capabilities**: Chat, Embedding, Image2Text, TTS, Speech2Text, Rerank ### Changes | File | Change | |------|--------| | `rag/llm/__init__.py` | Add `FuturMix` to `SupportedLiteLLMProvider` enum, `FACTORY_DEFAULT_BASE_URL`, and `LITELLM_PROVIDER_PREFIX` | | `rag/llm/chat_model.py` | Add `FuturMixChat(Base)` — follows Astraflow/Avian pattern | | `rag/llm/embedding_model.py` | Add `FuturMixEmbed(OpenAIEmbed)` — follows Astraflow pattern | | `rag/llm/cv_model.py` | Add `FuturMixCV(GptV4)` — follows SILICONFLOW/OpenRouter pattern | | `rag/llm/tts_model.py` | Add `FuturMixTTS(OpenAITTS)` — follows CometAPI/DeerAPI pattern | | `rag/llm/sequence2txt_model.py` | Add `FuturMixSeq2txt(GPTSeq2txt)` — follows StepFun pattern | | `rag/llm/rerank_model.py` | Add `FuturMixRerank(OpenAI_APIRerank)` | | `conf/llm_factories.json` | Add factory config with 8 chat, 2 embedding, 1 image2text, 2 TTS, 1 speech2text models | | `docs/guides/models/supported_models.mdx` | Add FuturMix to supported models table | ### Models included - **Chat**: claude-sonnet-4-20250514, claude-3.5-haiku, gpt-4o, gpt-4o-mini, gemini-2.5-flash, gemini-2.0-flash, deepseek-chat, deepseek-reasoner - **Embedding**: text-embedding-3-small, text-embedding-3-large - **Image2Text**: gpt-4o - **TTS**: tts-1, tts-1-hd - **Speech2Text**: whisper-1 ## Test plan - [ ] Verify FuturMix appears in the model provider list in RAGFlow UI - [ ] Configure FuturMix with API key and test chat completion - [ ] Test embedding model with document indexing - [ ] Test image2text with a sample image 🤖 Generated with [Claude Code](https://claude.com/claude-code) --------- Co-authored-by: Claude Opus 4.6 <noreply@anthropic.com>
2026-04-30 10:59:37 +08:00
import logging
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
from urllib.parse import urlparse
import requests
from openai import OpenAI
from openai.lib.azure import AzureOpenAI
from common.token_utils import num_tokens_from_string
2026-07-09 15:53:06 +08:00
from rag.utils.url_utils import ensure_v1
class Base(ABC):
def __init__(self, key, model_name, **kwargs):
"""
Abstract base class constructor.
Parameters are not stored; initialization is left to subclasses.
"""
pass
def transcription(self, audio_path, **kwargs):
with open(audio_path, "rb") as audio_file:
transcription = self.client.audio.transcriptions.create(model=self.model_name, file=audio_file)
return transcription.text.strip(), num_tokens_from_string(transcription.text.strip())
2026-07-16 10:49:02 +08:00
@staticmethod
def _generate_test_wav(duration_seconds=0.5, sample_rate=16000):
"""Generate a minimal silent WAV file as bytes (pure stdlib, no dependencies)."""
n_samples = int(sample_rate * duration_seconds)
data_size = n_samples * 2 # 16-bit mono = 2 bytes per sample
header = struct.pack(
"<4sI4s4sIHHIIHH4sI",
b"RIFF",
36 + data_size,
b"WAVE",
b"fmt ",
16,
1,
1,
sample_rate,
sample_rate * 2,
2,
16,
b"data",
data_size,
)
return header + b"\x00" * data_size
def check_available(self) -> tuple[bool, str]:
"""Check if the ASR model is available by transcribing a minimal test WAV."""
try:
wav_data = self._generate_test_wav()
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as f:
f.write(wav_data)
temp_path = f.name
try:
text, _ = self.transcription(temp_path)
if text.find("**ERROR**") >= 0:
return False, text.replace("**ERROR**: ", "").strip()
return True, ""
finally:
if os.path.exists(temp_path):
os.unlink(temp_path)
except Exception as e:
return False, str(e)
def audio2base64(self, audio):
if isinstance(audio, bytes):
return base64.b64encode(audio).decode("utf-8")
if isinstance(audio, io.BytesIO):
return base64.b64encode(audio.getvalue()).decode("utf-8")
raise TypeError("The input audio file should be in binary format.")
class GPTSeq2txt(Base):
_FACTORY_NAME = "OpenAI"
def __init__(self, key, model_name="whisper-1", base_url="https://api.openai.com/v1", **kwargs):
if not base_url:
base_url = "https://api.openai.com/v1"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.client = OpenAI(api_key=key, base_url=self.base_url)
self.model_name = model_name
class StepFunSeq2txt(GPTSeq2txt):
_FACTORY_NAME = "StepFun"
def __init__(self, key, model_name="step-asr", lang="Chinese", base_url="https://api.stepfun.com/v1", **kwargs):
if not base_url:
base_url = "https://api.stepfun.com/v1"
super().__init__(key, model_name=model_name, base_url=base_url, **kwargs)
feat: add FuturMix as model provider (#14419) ## Summary Add [FuturMix](https://futurmix.ai) as a new model provider. FuturMix is an OpenAI-compatible unified AI gateway that provides access to 22+ models (GPT, Claude, Gemini, DeepSeek, and more) through a single API endpoint and key. - **API Base**: `https://futurmix.ai/v1` (OpenAI-compatible) - **Supported capabilities**: Chat, Embedding, Image2Text, TTS, Speech2Text, Rerank ### Changes | File | Change | |------|--------| | `rag/llm/__init__.py` | Add `FuturMix` to `SupportedLiteLLMProvider` enum, `FACTORY_DEFAULT_BASE_URL`, and `LITELLM_PROVIDER_PREFIX` | | `rag/llm/chat_model.py` | Add `FuturMixChat(Base)` — follows Astraflow/Avian pattern | | `rag/llm/embedding_model.py` | Add `FuturMixEmbed(OpenAIEmbed)` — follows Astraflow pattern | | `rag/llm/cv_model.py` | Add `FuturMixCV(GptV4)` — follows SILICONFLOW/OpenRouter pattern | | `rag/llm/tts_model.py` | Add `FuturMixTTS(OpenAITTS)` — follows CometAPI/DeerAPI pattern | | `rag/llm/sequence2txt_model.py` | Add `FuturMixSeq2txt(GPTSeq2txt)` — follows StepFun pattern | | `rag/llm/rerank_model.py` | Add `FuturMixRerank(OpenAI_APIRerank)` | | `conf/llm_factories.json` | Add factory config with 8 chat, 2 embedding, 1 image2text, 2 TTS, 1 speech2text models | | `docs/guides/models/supported_models.mdx` | Add FuturMix to supported models table | ### Models included - **Chat**: claude-sonnet-4-20250514, claude-3.5-haiku, gpt-4o, gpt-4o-mini, gemini-2.5-flash, gemini-2.0-flash, deepseek-chat, deepseek-reasoner - **Embedding**: text-embedding-3-small, text-embedding-3-large - **Image2Text**: gpt-4o - **TTS**: tts-1, tts-1-hd - **Speech2Text**: whisper-1 ## Test plan - [ ] Verify FuturMix appears in the model provider list in RAGFlow UI - [ ] Configure FuturMix with API key and test chat completion - [ ] Test embedding model with document indexing - [ ] Test image2text with a sample image 🤖 Generated with [Claude Code](https://claude.com/claude-code) --------- Co-authored-by: Claude Opus 4.6 <noreply@anthropic.com>
2026-04-30 10:59:37 +08:00
class FuturMixSeq2txt(GPTSeq2txt):
_FACTORY_NAME = "FuturMix"
def __init__(self, key, model_name="whisper-1", base_url="https://futurmix.ai/v1", **kwargs):
if not base_url:
base_url = "https://futurmix.ai/v1"
super().__init__(key, model_name=model_name, base_url=base_url, **kwargs)
logging.info("[FuturMix] Speech2Text initialized with model %s", model_name)
class QWenSeq2txt(Base):
_FACTORY_NAME = "Tongyi-Qianwen"
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
_FUN_ASR_FLASH_PREFIX = "fun-asr-flash"
_FUN_ASR_BASE64_MAX_SIZE = 10 * 1024 * 1024
_DASHSCOPE_API_BASE = "https://dashscope.aliyuncs.com/api/v1"
_AUDIO_MIME_FORMATS = {
"audio/mpeg": "mp3",
"audio/mp3": "mp3",
"audio/wav": "wav",
"audio/wave": "wav",
"audio/x-wav": "wav",
}
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
def __init__(self, key, model_name="qwen-audio-asr", base_url=None, **kwargs):
import dashscope
dashscope.api_key = key
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
self.api_key = key
self.model_name = model_name
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
self.base_url = (base_url or self._DASHSCOPE_API_BASE).rstrip("/")
def transcription(self, audio_path):
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
# Fun-ASR-Flash uses DashScope's workspace-scoped native multimodal
# endpoint and payload instead of MultiModalConversation.
if self.model_name.startswith(self._FUN_ASR_FLASH_PREFIX):
return self._transcribe_fun_asr_flash(audio_path)
return self._transcribe_qwen_audio(audio_path)
def _transcribe_qwen_audio(self, audio_path):
import dashscope
if audio_path.startswith("http"):
audio_input = audio_path
else:
audio_input = f"file://{audio_path}"
messages = [{"role": "system", "content": [{"text": ""}]}, {"role": "user", "content": [{"audio": audio_input}]}]
resp = dashscope.MultiModalConversation.call(model=self.model_name, messages=messages, result_format="message", asr_options={"enable_lid": True, "enable_itn": False})
try:
text = resp["output"]["choices"][0]["message"].content[0]["text"]
except Exception as e:
text = "**ERROR**: " + str(e)
return text, num_tokens_from_string(text)
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
@classmethod
def _fun_asr_audio_format(cls, audio_path):
"""Derive the Fun-ASR audio format from a data URI, URL, or path."""
if audio_path.startswith("data:"):
mime_type = audio_path[5:].split(";", 1)[0].lower()
if not mime_type.startswith("audio/"):
raise ValueError(f"Unsupported audio data URI MIME type: {mime_type or 'missing'}")
audio_format = cls._AUDIO_MIME_FORMATS.get(mime_type, mime_type.split("/", 1)[1].removeprefix("x-"))
else:
path = urlparse(audio_path).path if audio_path.startswith(("http://", "https://")) else audio_path
audio_format = os.path.splitext(path)[1].lower().lstrip(".")
if audio_format == "wave":
audio_format = "wav"
if not audio_format:
raise ValueError("Cannot determine audio format; use a URL/path extension or an audio data URI MIME type")
return audio_format
@classmethod
def _validate_fun_asr_base64_size(cls, encoded_size):
if encoded_size > cls._FUN_ASR_BASE64_MAX_SIZE:
raise ValueError("Fun-ASR-Flash Base64 audio exceeds the 10 MB encoded-input limit; provide a publicly accessible URL (for example, OSS) instead")
def _fun_asr_flash_request(self, audio_path, *, stream=False):
audio_format = self._fun_asr_audio_format(audio_path)
if audio_path.startswith(("http://", "https://")):
audio_input = audio_path
elif audio_path.startswith("data:"):
_, separator, encoded_audio = audio_path.partition(",")
if not separator:
raise ValueError("Invalid audio data URI: missing Base64 payload")
self._validate_fun_asr_base64_size(len(encoded_audio.encode("utf-8")))
audio_input = audio_path
else:
file_size = os.path.getsize(audio_path)
encoded_size = 4 * ((file_size + 2) // 3)
self._validate_fun_asr_base64_size(encoded_size)
mime_type = "audio/mpeg" if audio_format == "mp3" else f"audio/{audio_format}"
with open(audio_path, "rb") as audio_file:
audio_input = f"data:{mime_type};base64,{base64.b64encode(audio_file.read()).decode('utf-8')}"
api_base = self.base_url
if api_base.endswith("/compatible-mode/v1"):
api_base = api_base[: -len("/compatible-mode/v1")] + "/api/v1"
url = f"{api_base}/services/aigc/multimodal-generation/generation"
payload = {
"model": self.model_name,
"input": {
"messages": [
{
"role": "user",
"content": [{"type": "input_audio", "input_audio": {"data": audio_input}}],
}
]
},
# sample_rate is optional in the Fun-ASR-Flash API. Omitting it
# avoids declaring incorrect metadata for remote or compressed audio.
"parameters": {"format": audio_format},
}
headers = {
"Authorization": f"Bearer {self.api_key}",
"Content-Type": "application/json",
"X-DashScope-SSE": "enable" if stream else "disable",
}
return url, headers, payload
def _transcribe_fun_asr_flash(self, audio_path):
try:
url, headers, payload = self._fun_asr_flash_request(audio_path)
response = requests.post(url, headers=headers, json=payload, timeout=60)
response.raise_for_status()
result = response.json()
text = result.get("text") or result.get("output", {}).get("text")
if not text:
raise ValueError("Missing transcription text in Fun-ASR-Flash response")
text = text.strip()
return text, num_tokens_from_string(text)
except Exception as e:
logging.exception("Fun-ASR-Flash transcription failed for model %s", self.model_name)
return "**ERROR**: " + str(e), 0
def _stream_fun_asr_flash(self, audio_path):
response = None
try:
url, headers, payload = self._fun_asr_flash_request(audio_path, stream=True)
response = requests.post(url, headers=headers, json=payload, timeout=60, stream=True)
response.raise_for_status()
full = ""
for line in response.iter_lines(decode_unicode=True):
if not line or not line.startswith("data:"):
continue
event_data = line[5:].strip()
if not event_data or event_data == "[DONE]":
continue
result = json.loads(event_data)
text = result.get("text") or result.get("output", {}).get("text")
if not text:
continue
full = text.strip()
yield {"event": "delta", "text": full}
if not full:
raise ValueError("Missing transcription text in Fun-ASR-Flash stream")
yield {"event": "final", "text": full}
except Exception as e:
logging.exception("Fun-ASR-Flash streaming transcription failed for model %s", self.model_name)
yield {"event": "error", "text": "**ERROR**: " + str(e)}
finally:
if response is not None:
response.close()
def stream_transcription(self, audio_path):
feat(stt): support Fun-ASR-Flash in Tongyi-Qianwen provider (#16844) ## What this PR does Adds support for Alibaba Cloud's hosted Fun-ASR-Flash snapshots to the existing Tongyi-Qianwen speech-to-text provider. - registers `fun-asr-flash-2026-06-15` as a speech-to-text model; - routes only `fun-asr-flash*` models to the documented workspace-native multimodal-generation endpoint; - supports local audio through size-checked data URIs as well as URL/data-URI inputs; - uses the documented SSE response mode for incremental streaming transcription; - closes the streamed HTTP response on completion, failure, or early consumer cancellation; - preserves the existing `dashscope.MultiModalConversation` path for all other Qwen audio models; - keeps RAGFlow's existing synchronous and streaming adapter interfaces. ## Why Fun-ASR-Flash does not use the legacy Qwen audio request shape currently used by `QWenSeq2txt`. Its synchronous API expects `input_audio` at: `/api/v1/services/aigc/multimodal-generation/generation` Without a narrowly scoped adapter path, the hosted model cannot be selected successfully through RAGFlow's Tongyi-Qianwen speech-to-text provider. Closes #16843. ## Compatibility The new behavior is gated by the `fun-asr-flash` model-name prefix. Existing Qwen audio models continue through the original code path unchanged. ## Validation - `pytest test/unit_test/rag/llm/test_sequence2txt_model.py`: 10 passed - Ruff check: passed - Ruff format check: passed - `llm_factories.json` validation: passed - Real hosted-API validation with WAV audio - Real RAGFlow upload/indexing validation with MP3 audio The unit tests cover the native Fun-ASR-Flash request, regression behavior for the legacy Qwen path, SSE streaming, and early response cleanup. ## Documentation - https://help.aliyun.com/document_detail/2979031.html - https://help.aliyun.com/document_detail/2869541.html ### Why a dedicated adapter path is necessary (official evidence) Alibaba Cloud's [Fun-ASR RESTful API reference](https://help.aliyun.com/en/model-studio/fun-asr-recorded-speech-recognition-http-api) makes the incompatibilities with RAGFlow's existing Qwen audio path explicit: | Adapter change | Official API requirement | Why the existing path is insufficient | | --- | --- | --- | | Call the workspace-native HTTP endpoint | The Fun-ASR-Flash synchronous section states that SDK calls are not supported and specifies `POST /api/v1/services/aigc/multimodal-generation/generation`. | The existing adapter calls `dashscope.MultiModalConversation`, so a direct HTTP path is required. | | Use the `input_audio` message shape | `input.messages`, `content`, `type: input_audio`, `input_audio`, and `input_audio.data` are documented as required for an audio request. | The existing Qwen path sends the legacy `audio` content shape, which does not match this API contract. | | Send `parameters.format` | The request schema marks `parameters` and `format` as **Required**, and says the value must match the actual audio format. | The legacy request has no Fun-ASR-Flash `parameters.format` field, so the adapter must derive and send it. | | Encode local files as Data URIs | `input_audio.data` accepts either a public URL or a Base64 Data URI; the reference gives the exact `data:{MIME_TYPE};base64,...` form. | RAGFlow supplies local file paths, which the remote API cannot read directly. | | Parse `output.text` | The documented non-streaming response returns the accumulated transcription in `output.text`. | The legacy Qwen response parser reads `output.choices[].message.content`, so a separate response parser is required. | | Enforce the Base64 input limit | The reference requires the Base64-encoded audio to remain within the 10 MB input limit. | The adapter checks encoded size before reading/sending local audio and directs oversized inputs to the existing public-URL path. | | Use SSE for streaming | The reference specifies `X-DashScope-SSE: enable` and documents intermediate and final SSE events. | The adapter parses those events instead of wrapping one blocking response as a synthetic stream. | | Release streamed responses | Streaming responses must be closed when iteration completes or stops early. | A `finally` cleanup releases the HTTP response on completion, errors, and consumer cancellation. | `sample_rate` is documented as **Optional**. The implementation omits it instead of declaring a fixed value that may not match remote or compressed audio. The [official speech-to-text model list](https://help.aliyun.com/en/model-studio/asr-model/) separately confirms that `fun-asr-flash-2026-06-15` is an offline HTTP model with a five-minute audio limit. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: openhands <openhands@all-hands.dev> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-16 09:37:37 +08:00
if self.model_name.startswith(self._FUN_ASR_FLASH_PREFIX):
yield from self._stream_fun_asr_flash(audio_path)
return
import dashscope
if audio_path.startswith("http"):
audio_input = audio_path
else:
audio_input = f"file://{audio_path}"
messages = [{"role": "system", "content": [{"text": ""}]}, {"role": "user", "content": [{"audio": audio_input}]}]
stream = dashscope.MultiModalConversation.call(model=self.model_name, messages=messages, result_format="message", stream=True, asr_options={"enable_lid": True, "enable_itn": False})
full = ""
for chunk in stream:
try:
piece = chunk["output"]["choices"][0]["message"].content[0]["text"]
full = piece
yield {"event": "delta", "text": piece}
except Exception as e:
yield {"event": "error", "text": str(e)}
yield {"event": "final", "text": full}
class AzureSeq2txt(Base):
_FACTORY_NAME = "Azure-OpenAI"
def __init__(self, key, model_name, lang="Chinese", **kwargs):
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(kwargs["base_url"])
self.client = AzureOpenAI(api_key=key, azure_endpoint=self.base_url, api_version="2024-02-01")
self.model_name = model_name
self.lang = lang
class XinferenceSeq2txt(Base):
_FACTORY_NAME = "Xinference"
def __init__(self, key, model_name="whisper-small", **kwargs):
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(kwargs["base_url"]) if kwargs.get("base_url") else None
self.model_name = model_name
self.key = key
def transcription(self, audio, language="zh", prompt=None, response_format="json", temperature=0.7):
if isinstance(audio, str):
with open(audio, "rb") as audio_file:
audio_data = audio_file.read()
audio_file_name = audio.split("/")[-1]
else:
audio_data = audio
audio_file_name = "audio.wav"
payload = {"model": self.model_name, "language": language, "prompt": prompt, "response_format": response_format, "temperature": temperature}
files = {"file": (audio_file_name, audio_data, "audio/wav")}
try:
response = requests.post(f"{self.base_url}/v1/audio/transcriptions", files=files, data=payload, timeout=60)
response.raise_for_status()
result = response.json()
if "text" in result:
transcription_text = result["text"].strip()
return transcription_text, num_tokens_from_string(transcription_text)
else:
return "**ERROR**: Failed to retrieve transcription.", 0
except requests.exceptions.RequestException as e:
return f"**ERROR**: {str(e)}", 0
class TencentCloudSeq2txt(Base):
_FACTORY_NAME = "Tencent Cloud"
def __init__(self, key, model_name="16k_zh", base_url="https://asr.tencentcloudapi.com"):
from tencentcloud.asr.v20190614 import asr_client
from tencentcloud.common import credential
key = json.loads(key)
sid = key.get("tencent_cloud_sid", "")
sk = key.get("tencent_cloud_sk", "")
cred = credential.Credential(sid, sk)
self.client = asr_client.AsrClient(cred, "")
self.model_name = model_name
2026-07-16 10:49:02 +08:00
def check_available(self) -> tuple[bool, str]:
"""Tencent Cloud ASR transcription expects raw bytes, not a file path."""
try:
wav_data = self._generate_test_wav()
text, _ = self.transcription(wav_data)
if text.find("**ERROR**") >= 0:
return False, text.replace("**ERROR**: ", "").strip()
return True, ""
except Exception as e:
return False, str(e)
def transcription(self, audio, max_retries=60, retry_interval=5):
import time
from tencentcloud.asr.v20190614 import models
from tencentcloud.common.exception.tencent_cloud_sdk_exception import (
TencentCloudSDKException,
)
b64 = self.audio2base64(audio)
try:
# dispatch disk
req = models.CreateRecTaskRequest()
params = {
"EngineModelType": self.model_name,
"ChannelNum": 1,
"ResTextFormat": 0,
"SourceType": 1,
"Data": b64,
}
req.from_json_string(json.dumps(params))
resp = self.client.CreateRecTask(req)
# loop query
req = models.DescribeTaskStatusRequest()
params = {"TaskId": resp.Data.TaskId}
req.from_json_string(json.dumps(params))
retries = 0
while retries < max_retries:
resp = self.client.DescribeTaskStatus(req)
if resp.Data.StatusStr == "success":
text = re.sub(r"\[\d+:\d+\.\d+,\d+:\d+\.\d+\]\s*", "", resp.Data.Result).strip()
return text, num_tokens_from_string(text)
elif resp.Data.StatusStr == "failed":
return (
"**ERROR**: Failed to retrieve speech recognition results.",
0,
)
else:
time.sleep(retry_interval)
retries += 1
return "**ERROR**: Max retries exceeded. Task may still be processing.", 0
except TencentCloudSDKException as e:
return "**ERROR**: " + str(e), 0
except Exception as e:
return "**ERROR**: " + str(e), 0
class GPUStackSeq2txt(Base):
_FACTORY_NAME = "GPUStack"
def __init__(self, key, model_name, base_url):
if not base_url:
raise ValueError("url cannot be None")
if base_url.split("/")[-1] != "v1":
base_url = os.path.join(base_url, "v1")
self.base_url = base_url
self.model_name = model_name
self.key = key
2026-07-16 10:49:02 +08:00
def check_available(self) -> tuple[bool, str]:
"""GPUStack ASR transcription endpoint is not yet implemented."""
return False, "GPUStack ASR transcription is not yet implemented"
class GiteeSeq2txt(Base):
_FACTORY_NAME = "GiteeAI"
def __init__(self, key, model_name="whisper-1", base_url="https://ai.gitee.com/v1/", **kwargs):
if not base_url:
base_url = "https://ai.gitee.com/v1/"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.client = OpenAI(api_key=key, base_url=self.base_url)
self.model_name = model_name
class DeepInfraSeq2txt(Base):
_FACTORY_NAME = "DeepInfra"
def __init__(self, key, model_name, base_url="https://api.deepinfra.com/v1/openai", **kwargs):
if not base_url:
base_url = "https://api.deepinfra.com/v1/openai"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.client = OpenAI(api_key=key, base_url=self.base_url)
self.model_name = model_name
class CometAPISeq2txt(Base):
_FACTORY_NAME = "CometAPI"
def __init__(self, key, model_name="whisper-1", base_url="https://api.cometapi.com/v1", **kwargs):
if not base_url:
base_url = "https://api.cometapi.com/v1"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.client = OpenAI(api_key=key, base_url=self.base_url)
self.model_name = model_name
class DeerAPISeq2txt(Base):
_FACTORY_NAME = "DeerAPI"
def __init__(self, key, model_name="whisper-1", base_url="https://api.deerapi.com/v1", **kwargs):
if not base_url:
base_url = "https://api.deerapi.com/v1"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.client = OpenAI(api_key=key, base_url=self.base_url)
self.model_name = model_name
class ZhipuSeq2txt(Base):
_FACTORY_NAME = "ZHIPU-AI"
def __init__(self, key, model_name="glm-asr", base_url="https://open.bigmodel.cn/api/paas/v4", **kwargs):
if not base_url:
base_url = "https://open.bigmodel.cn/api/paas/v4"
self.base_url = base_url
self.api_key = key
self.model_name = model_name
self.gen_conf = kwargs.get("gen_conf", {})
self.stream = kwargs.get("stream", False)
def _convert_to_wav(self, input_path):
ext = os.path.splitext(input_path)[1].lower()
if ext in [".wav", ".mp3"]:
return input_path
fd, out_path = tempfile.mkstemp(suffix=".wav")
os.close(fd)
try:
import ffmpeg
import imageio_ffmpeg as ffmpeg_exe
ffmpeg_path = ffmpeg_exe.get_ffmpeg_exe()
(ffmpeg.input(input_path).output(out_path, ar=16000, ac=1).overwrite_output().run(cmd=ffmpeg_path, quiet=True))
return out_path
except Exception as e:
raise RuntimeError(f"audio convert failed: {e}")
def transcription(self, audio_path):
payload = {
"model": self.model_name,
"temperature": str(self.gen_conf.get("temperature", 0.75)) or "0.75",
"stream": self.stream,
}
headers = {"Authorization": f"Bearer {self.api_key}"}
converted = self._convert_to_wav(audio_path)
with open(converted, "rb") as audio_file:
files = {"file": audio_file}
try:
response = requests.post(
url=f"{self.base_url}/audio/transcriptions",
data=payload,
files=files,
headers=headers,
timeout=60,
)
body = response.json()
if response.status_code == 200:
full_content = body["text"]
return full_content, num_tokens_from_string(full_content)
else:
error = body["error"]
return f"**ERROR**: code: {error['code']}, message: {error['message']}", 0
except Exception as e:
return "**ERROR**: " + str(e), 0
class RAGconSeq2txt(Base):
"""
RAGcon Sequence2Text Provider - routes through LiteLLM proxy
Speech-to-text models routed through LiteLLM.
Default Base URL: https://connect.ragcon.com/v1
"""
_FACTORY_NAME = "RAGcon"
def __init__(self, key, model_name, base_url=None, lang="English", **kwargs):
# Use provided base_url or fallback to default
if not base_url:
base_url = "https://connect.ragcon.com/v1"
2026-07-09 15:53:06 +08:00
self.base_url = ensure_v1(base_url)
self.model_name = model_name
self.key = key
self.lang = lang
self.client = OpenAI(api_key=key, base_url=self.base_url)
def transcription(self, audio_path, **kwargs):
"""
Transcribe audio file using RAGcon's OpenAI-compatible API.
Uses Whisper's automatic language detection for German and English audio.
Args:
audio_path: Path to the audio file
**kwargs: Additional parameters (currently unused but maintained for compatibility)
Returns:
tuple: (transcribed_text, token_count)
"""
with open(audio_path, "rb") as audio_file:
# Call RAGcon API - Whisper will auto-detect language
transcription = self.client.audio.transcriptions.create(model=self.model_name, file=audio_file)
# Return text and token count
text = transcription.text.strip()
return text, num_tokens_from_string(text)
Feat: Add New API model provider for OpenAI-compatible gateways (#15991) ## Summary Add support for **"New API"** as a model provider, enabling connection to [New API](https://github.com/QuantumNous/new-api) / [one-api](https://github.com/songquanpeng/one-api) compatible gateways that aggregate multiple LLM backends behind a unified OpenAI-compatible `/v1` endpoint. ### Features - **All model types**: Chat, Embedding, Rerank, Image2Text, TTS, Speech2Text - **List Models discovery**: `NewAPI(OpenAIAPICompatible)` class in `model_meta.py` queries the gateway's `/v1/models` to auto-discover available models via the native `GET /api/v1/providers/<name>/models` endpoint - **Model parameter editing**: Pencil icon on each discovered model row to edit `model_type`, `max_tokens`, and `features` (e.g. tool call support) before submitting - **Custom model addition**: "Add Custom Model" button at the bottom of the List Models dropdown for models not returned by the API - **Gear icon settings**: Enabled the Settings gear button on provider instances to manage models on existing instances (viewMode) - **viewMode credential passthrough**: Fixed List Models in viewMode — merges `initialValues` credentials when `api_key`/`base_url` fields are hidden by `hideWhenInstanceExists` ### Changes **Backend** (8 files): - `rag/llm/chat_model.py` — `NewAPIChat(Base)` class - `rag/llm/embedding_model.py` — `NewAPIEmbed(OpenAIEmbed)` class (no auto `/v1` append) - `rag/llm/rerank_model.py` — `NewAPIRerank(Base)` class (uses `/rerank` endpoint) - `rag/llm/cv_model.py` — `NewAPICv(GptV4)` class - `rag/llm/tts_model.py` — `NewAPITTS(OpenAITTS)` class - `rag/llm/sequence2txt_model.py` — `NewAPISeq2txt(GPTSeq2txt)` class - `rag/llm/model_meta.py` — `NewAPI(OpenAIAPICompatible)` class for List Models discovery - `conf/llm_factories.json` — New API factory entry with all model type tags **Frontend** (8 files + 1 new SVG): - `web/src/assets/svg/llm/new-api.svg` — New API logo icon - `web/src/constants/llm.ts` — `LLMFactory.NewAPI` enum + `IconMap` entry - `web/src/components/svg-icon.tsx` — `NewAPI` added to `svgIcons` - `web/src/pages/user-setting/setting-model/modal/provider-modal/field-config/local-llm-configs.ts` — New API `buildLocalConfig` - `web/src/pages/user-setting/setting-model/modal/provider-modal/constants.ts` — `LIST_MODEL_PROVIDERS` includes NewAPI - `web/src/pages/user-setting/setting-model/components/used-model.tsx` — Enable Settings gear button - `web/src/pages/user-setting/setting-model/modal/provider-modal/hooks/use-list-models-picker.ts` — viewMode credential merge + model editing state/handlers - `web/src/pages/user-setting/setting-model/modal/provider-modal/hooks/use-list-models-options.tsx` — Pencil edit icon per model row - `web/src/pages/user-setting/setting-model/modal/provider-modal/index.tsx` — `AddCustomModelDialog` import + edit dialog rendering **Note on Go implementation**: A Go model driver (`NewAPIModel` delegating to `OpenAIModel`) has been prepared but is deferred until the Go runtime is enabled in a future release (current v0.26.0 images use `API_PROXY_SCHEME=python` and do not compile Go binaries). Will submit as a follow-up PR. ## Related - Depends on: #15996 (provider instance API improvements — server-side credential lookup, idempotent `add_model`, security fixes — required for viewMode gear icon and batch model submission) ## Test plan - [ ] Add New API provider with api_key and base_url pointing to an OpenAI-compatible gateway - [ ] Click "List Models" — should discover and display available models from `/v1/models` - [ ] Click pencil icon on a model — should open edit dialog to change model_type, max_tokens, features - [ ] Select multiple models and click OK — should add all selected models - [ ] Click gear icon on the added instance — should open viewMode with List Models working - [ ] In viewMode, select new models including pre-existing ones, click OK — should succeed (requires #15996) - [ ] Verify all model types work: create a Chat assistant, Embedding KB, Rerank setting 🤖 Generated with [Claude Code](https://claude.com/claude-code) --------- Co-authored-by: Tim Wang <wanghualoong@users.noreply.github.com> Co-authored-by: Claude Opus 4.6 <noreply@anthropic.com>
2026-06-26 18:47:20 +08:00
class NewAPISeq2txt(GPTSeq2txt):
_FACTORY_NAME = "New API"
def __init__(self, key, model_name="whisper-1", base_url="", **kwargs):
if not base_url:
raise ValueError("url cannot be None")
model_name = model_name.split("___")[0]
super().__init__(key, model_name=model_name, base_url=base_url, **kwargs)
feat(stt): add FunASR / SenseVoice provider (#16473) ### Summary Adds FunASR as a self-hosted speech-to-text provider through its OpenAI-compatible `/v1/audio/transcriptions` endpoint. This is a focused replacement for #15526 by @Rene0422 and relates to #15448. The unrelated Markdown parser changes from the previous branch are intentionally removed so this PR contains only the FunASR provider integration. - register FunASR as a `SPEECH2TEXT` factory; - add `FunASRSeq2txt` with `sensevoice` and `http://localhost:8000/v1` defaults, an optional API key, URL normalization, and inherited transcription handling; - wire FunASR into the current local-provider schema with a prefilled local URL and official documentation link; - discover the server's `/v1/models` dynamically and expose every returned model as speech-to-text in the model picker; - use RAGFlow's existing default provider icon fallback instead of referencing a missing `funasr` asset; - list FunASR in the supported-provider documentation; - add focused backend and frontend regression tests. ### Validation - focused backend pytest suite -> `7 passed` - real CPU `funasr-server` + RAGFlow provider smoke test -> discovered `fun-asr-nano`, `sensevoice`, and `paraformer`; transcribed a real WAV as `我现在在录一段测试音频` (`10` tokens, `0.504s`) - `ruff check` and `ruff format --check` on the changed Python files - `python3 -m py_compile` on the provider and its test - JSON parse and a semantic assertion for exactly one enabled FunASR `SPEECH2TEXT` factory - focused frontend Jest test -> `2 passed` - ESLint and Prettier on all changed TypeScript files - `npm run build` -> production build succeeded (`14,181` modules transformed) - `git diff --check` ### Deployment Run FunASR separately and point the RAGFlow provider at it: ```bash pip install funasr funasr-server --device cuda --model sensevoice ``` The API key remains optional because the stock local server does not require authentication. A key can still be supplied when the endpoint is protected by a gateway. --------- Signed-off-by: LauraGPT <LauraGPT@users.noreply.github.com> Co-authored-by: LauraGPT <LauraGPT@users.noreply.github.com>
2026-07-15 19:02:05 +08:00
class FunASRSeq2txt(GPTSeq2txt):
"""FunASR speech-to-text provider for its OpenAI-compatible API."""
_FACTORY_NAME = "FunASR"
def __init__(self, key, model_name="sensevoice", base_url="http://localhost:8000/v1", **kwargs):
"""Initialize a client for a FunASR OpenAI-compatible endpoint."""
if not base_url:
base_url = "http://localhost:8000/v1"
super().__init__(key=key or "funasr", model_name=model_name, base_url=base_url, **kwargs)
logging.info("[FunASR] Speech2Text initialized with model %s at %s", model_name, self.base_url)