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| import torch | |
| from transformers import pipeline | |
| from transformers.pipelines.audio_utils import ffmpeg_read | |
| import gradio as gr | |
| import os | |
| hugapikey=os.environ['openaikey'] | |
| #hugapikey='test' | |
| genaikey=os.environ['genaikey'] | |
| #genaikey='test' | |
| #MODEL_NAME = "seiching/whisper-small-seiching" | |
| MODEL_NAME = "openai/whisper-tiny" | |
| BATCH_SIZE = 8 | |
| DEFAULTPROMPT='你是專業的會議紀錄製作員,請根據由語音辨識軟體將會議錄音所轉錄的逐字稿,也請注意逐字稿可能有錯,請先做校正,討論內容細節請略過,請根據校正過的逐字稿撰寫會議紀錄,並要用比較正式及容易閱讀的寫法,避免口語化' | |
| # | |
| device = 0 if torch.cuda.is_available() else "cpu" | |
| pipe = pipeline( | |
| task="automatic-speech-recognition", | |
| model=MODEL_NAME, | |
| chunk_length_s=30, | |
| device=device, | |
| ) | |
| from openai import OpenAI | |
| from concurrent.futures import ThreadPoolExecutor | |
| import tiktoken | |
| def call_openai_makenote(openaiobj,transcription,usemodelname): | |
| ## 直接做會議紀錄,GPT4或GPT 3.5但小於16K | |
| response = openaiobj.chat.completions.create( | |
| #model="gpt-3.5-turbo", | |
| model=usemodelname, | |
| temperature=0, | |
| messages=[ | |
| { | |
| "role": "system", | |
| "content": "你是專業的會議紀錄製作員,請根據由語音辨識軟體將會議錄音所轉錄的逐字稿,也請注意逐字稿可能有錯,請先做校正,討論內容細節請略過,請根據校正過的逐字稿撰寫會議紀錄,並要用比較正式及容易閱讀的寫法,避免口語化" | |
| }, | |
| { | |
| "role": "user", | |
| "content": transcription | |
| } | |
| ] | |
| ) | |
| return response.choices[0].message.content | |
| def call_openai_summary(openaiobj,transcription,usemodelname): | |
| ## 分段摘要 | |
| response = openaiobj.chat.completions.create( | |
| #model="gpt-3.5-turbo", | |
| model=usemodelname, | |
| temperature=0, | |
| messages=[ | |
| { | |
| "role": "system", | |
| "content": "你是專業的會議紀錄製作員,請根據由語音辨識軟體將會議錄音所轉錄的逐字稿,也請注意逐字稿可能有錯,請先校正,再摘要會議重點內容" | |
| }, | |
| { | |
| "role": "user", | |
| "content": transcription | |
| } | |
| ] | |
| ) | |
| return response.choices[0].message.content | |
| def call_openai_summaryall(openaiobj,transcription,usemodelname): | |
| response = openaiobj.chat.completions.create( | |
| #model="gpt-3.5-turbo", | |
| model=usemodelname, | |
| temperature=0, | |
| messages=[ | |
| { | |
| "role": "system", | |
| "content": "你是專業的會議紀錄製作員,請根據分段的會議摘要,彙整成正式會議紀錄,並要用比較正式及容易閱讀的寫法,避免口語化" | |
| }, | |
| { | |
| "role": "user", | |
| "content": transcription | |
| } | |
| ] | |
| ) | |
| return response.choices[0].message.content | |
| def split_into_chunks(text,LLMmodel, tokens=15900): | |
| #encoding = tiktoken.encoding_for_model('gpt-3.5-turbo') | |
| encoding = tiktoken.encoding_for_model(LLMmodel) | |
| words = encoding.encode(text) | |
| chunks = [] | |
| for i in range(0, len(words), tokens): | |
| chunks.append(' '.join(encoding.decode(words[i:i + tokens]))) | |
| return chunks | |
| def gpt3write(openaikeystr,inputtext,LLMmodel): | |
| # openaiobj = OpenAI( | |
| # # This is the default and can be omitted | |
| # api_key=openaikeystr, | |
| # ) | |
| if hugapikey=='test': | |
| realkey=openaikeystr | |
| else: | |
| realkey=hugapikey | |
| #openaiojb =OpenAI(base_url="http://localhost:1234/v1", api_key="not-needed") | |
| openaiobj =OpenAI( api_key=realkey) | |
| text = inputtext | |
| #openaikey.set_key(openaikeystr) | |
| #print('process_chunk',openaikey.get_key()) | |
| chunks = split_into_chunks(text,LLMmodel) | |
| i=1 | |
| if len(chunks)>1: | |
| response='這是分段會議紀錄摘要\n\n' | |
| for chunk in chunks: | |
| response=response+'第' +str(i)+'段\n'+call_openai_summary(openaiobj,chunk,LLMmodel)+'\n\n' | |
| i=i+1 | |
| finalresponse=response+'\n\n 這是根據以上分段會議紀錄彙編如下 \n\n' +call_openai_summaryall(openaiobj,response,LLMmodel) | |
| # response=response+call_openai_summary(openaiobj,chunk) | |
| else: | |
| finalresponse=call_openai_makenote(openaiobj,inputtext,LLMmodel) | |
| return finalresponse | |
| # # Processes chunks in parallel | |
| # with ThreadPoolExecutor() as executor: | |
| # responses = list(executor.map(call_openai_api, [openaiobj,chunks])) | |
| # return responses | |
| import torch | |
| from transformers import pipeline | |
| from transformers.pipelines.audio_utils import ffmpeg_read | |
| import gradio as gr | |
| transcribe_text="" | |
| device = 0 if torch.cuda.is_available() else "cpu" | |
| pipe = pipeline( | |
| task="automatic-speech-recognition", | |
| model=MODEL_NAME, | |
| chunk_length_s=30, | |
| device=device, | |
| ) | |
| # Copied from https://github.com/openai/whisper/blob/c09a7ae299c4c34c5839a76380ae407e7d785914/whisper/utils.py#L50 | |
| def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."): | |
| if seconds is not None: | |
| milliseconds = round(seconds * 1000.0) | |
| hours = milliseconds // 3_600_000 | |
| milliseconds -= hours * 3_600_000 | |
| minutes = milliseconds // 60_000 | |
| milliseconds -= minutes * 60_000 | |
| seconds = milliseconds // 1_000 | |
| milliseconds -= seconds * 1_000 | |
| hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else "" | |
| return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}" | |
| else: | |
| # we have a malformed timestamp so just return it as is | |
| return seconds | |
| def transcribe(file, return_timestamps): | |
| outputs = pipe(file, batch_size=BATCH_SIZE, generate_kwargs={"task": "transcribe","language": "chinese",}, return_timestamps=return_timestamps) | |
| text = outputs["text"] | |
| if return_timestamps: | |
| timestamps = outputs["chunks"] | |
| timestamps = [ | |
| f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}" | |
| for chunk in timestamps | |
| ] | |
| text = "\n".join(str(feature) for feature in timestamps) | |
| global transcribe_text | |
| transcribe_text=text | |
| # with open('asr_resul.txt', 'w') as f: | |
| # f.write(text) | |
| # ainotes=process_chunks(text) | |
| # with open("ainotes_result.txt", "a") as f: | |
| # f.write(ainotes) | |
| return text | |
| demo = gr.Blocks() | |
| mic_transcribe = gr.Interface( | |
| fn=transcribe, | |
| inputs=[ | |
| gr.inputs.Audio(source="microphone", type="filepath", optional=True), | |
| # gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"), | |
| gr.inputs.Checkbox(default=False, label="Return timestamps"), | |
| ], | |
| outputs="text", | |
| layout="horizontal", | |
| theme="huggingface", | |
| title="會議紀錄小幫手AINotes", | |
| description=( | |
| "可由麥克風錄音或上傳語音檔" | |
| f" 使用這個模型 [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME})如果覺得速度有點慢, 可以用(https://huggingface.co/spaces/sanchit-gandhi/whisper-jax) 先做語音辨識再做會議紀錄摘要" | |
| " 長度沒有限制" | |
| ), | |
| allow_flagging="never", | |
| ) | |
| file_transcribe = gr.Interface( | |
| fn=transcribe, | |
| inputs=[ | |
| gr.inputs.Audio(source="upload", optional=True, label="Audio file", type="filepath"), | |
| # gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"), | |
| gr.inputs.Checkbox(default=False, label="Return timestamps"), | |
| ], | |
| outputs="text", | |
| layout="horizontal", | |
| theme="huggingface", | |
| title="會議紀錄小幫手AINotes", | |
| description=( | |
| "可由麥克風錄音或上傳語音檔" | |
| f" 使用這個模型 [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) 如果覺得速度有點慢, 可以用(https://huggingface.co/spaces/sanchit-gandhi/whisper-jax),先做語音辨識再做會議紀錄摘要" | |
| " 長度沒有限制" | |
| ), | |
| # examples=[ | |
| # ["./example.flac", "transcribe", False], | |
| # ["./example.flac", "transcribe", True], | |
| # ], | |
| cache_examples=True, | |
| allow_flagging="never", | |
| ) | |
| import google.generativeai as genai | |
| def gpt4write(openaikeystr,transcribe_text,LLMmodel): | |
| # openaiobj = OpenAI( | |
| # # This is the default and can be omitted | |
| # api_key=openaikeystr, | |
| # ) | |
| if hugapikey=='test': | |
| realkey=openaikeystr | |
| else: | |
| realkey=hugapikey | |
| #openaiojb =OpenAI(base_url="http://localhost:1234/v1", api_key="not-needed") | |
| openaiobj =OpenAI( api_key=realkey) | |
| #text = inputtext | |
| #openaikey.set_key(openaikeystr) | |
| #print('process_chunk',openaikey.get_key()) | |
| #chunks = split_into_chunks(text) | |
| #response='這是分段會議紀錄結果\n\n' | |
| finalresponse=call_openai_makenote(openaiobj,transcribe_text,LLMmodel) | |
| # response=response+call_openai_summary(openaiobj,chunk) | |
| return finalresponse | |
| return 'ok' | |
| def gewritenote(prompt,inputscript): | |
| api_key = genaikey | |
| genai.configure(api_key = api_key) | |
| model = genai.GenerativeModel('gemini-pro') | |
| #genprompt='你是專業的會議紀錄製作員,請根據由語音辨識軟體將會議錄音所轉錄的逐字稿,也請注意逐字稿可能有錯,請先做校正,討論內容細節請略過,請根據校正過的逐字稿撰寫會議紀錄,並要用比較正式及容易閱讀的寫法,避免口語化' | |
| genprompt=prompt+'#'+inputscript+'#' | |
| response = model.generate_content( genprompt) | |
| return response.text | |
| def writenotes( LLMmodel,apikeystr,prompt,inputscript): | |
| #text=transcribe_text | |
| #openaikey.set_key(inputkey) | |
| #openaikey = OpenAIKeyClass(inputkey) | |
| if(len(prompt))<10: | |
| prompt=DEFAULTPROMPT | |
| global transcribe_text | |
| print('ok') | |
| if len(inputscript)>10: #有資料表示不是來自語音辨識結果 | |
| transcribe_text=inputscript | |
| if LLMmodel=="gpt-3.5-turbo": | |
| ainotestext=gpt3write(apikeystr,transcribe_text,LLMmodel) | |
| elif LLMmodel=="gpt-4-0125-preview": | |
| ainotestext=gpt4write(apikeystr,transcribe_text,LLMmodel) | |
| elif LLMmodel=='gemini': | |
| ainotestext=gewritenote(prompt,transcribe_text) | |
| # ainotestext=inputscript | |
| #ainotestext="" | |
| # with open('asr_resul.txt', 'w') as f: | |
| # #print(transcribe_text) | |
| # # f.write(inputkey) | |
| # f.write(transcribe_text) | |
| # with open('ainotes.txt','w') as f: | |
| # f.write(ainotestext) | |
| return ainotestext | |
| ainotes = gr.Interface( | |
| fn=writenotes, | |
| inputs=[ gr.inputs.Radio(["gemini","gpt-3.5-turbo", "gpt-4-0125-preview"], label="LLMmodel", default="gemini"),gr.Textbox(label="使用GPT請輸入OPEN AI API KEY",placeholder="請輸入sk..."),gr.Textbox(label="自訂提示詞(prompt)若無會用以下預設值",info=DEFAULTPROMPT),gr.Textbox(label="逐字稿",placeholder="若沒有做語音辨識,請輸入逐字稿")], | |
| outputs="text", | |
| layout="horizontal", | |
| theme="huggingface", | |
| title="會議紀錄小幫手AINotes", | |
| description=( | |
| "可由麥克風錄音或上傳語音檔,並將本逐字稿欄位清空,若有逐字稿可以直接貼在逐字稿" | |
| f" 使用這個模型 [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) 如果覺得速度有點慢, 可以用(https://huggingface.co/spaces/sanchit-gandhi/whisper-jax), 做完語音辨識再貼過來做會議紀錄摘要" | |
| " 長度沒有限制" | |
| ), | |
| # examples=[ | |
| # ["./example.flac", "transcribe", False], | |
| # ["./example.flac", "transcribe", True], | |
| # ], | |
| cache_examples=True, | |
| allow_flagging="never", | |
| ) | |
| with demo: | |
| gr.TabbedInterface([file_transcribe,mic_transcribe,ainotes], ["語音檔辨識","麥克風語音檔辨識","產生會議紀錄" ]) | |
| demo.launch(enable_queue=True) |