ggoknar
commited on
Commit
•
da4b074
1
Parent(s):
d3d83c1
fix repo name
Browse files
app.py
CHANGED
@@ -68,7 +68,7 @@ HF_TOKEN = os.environ.get("HF_TOKEN")
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# will use api to restart space on a unrecoverable error
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api = HfApi(token=HF_TOKEN)
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-
repo_id = "
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default_system_message = """
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You are Mistral, a large language model trained and provided by Mistral, architecture of you is decoder-based LM. Your voice backend or text to speech TTS backend is provided via Coqui technology. You are right now served on Huggingface spaces.
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@@ -106,433 +106,3 @@ text_client = InferenceClient(
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"mistralai/Mistral-7B-Instruct-v0.1",
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timeout=WHISPER_TIMEOUT,
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)
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-
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-
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###### COQUI TTS FUNCTIONS ######
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def get_latents(speaker_wav):
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# create as function as we can populate here with voice cleanup/filtering
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(
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gpt_cond_latent,
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diffusion_conditioning,
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speaker_embedding,
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) = model.get_conditioning_latents(audio_path=speaker_wav)
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return gpt_cond_latent, diffusion_conditioning, speaker_embedding
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-
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-
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def format_prompt(message, history):
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prompt = (
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"<s>[INST]"
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+ system_message
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+ "[/INST] I understand, I am a Mistral chatbot with speech by Coqui team.</s>"
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)
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for user_prompt, bot_response in history:
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prompt += f"[INST] {user_prompt} [/INST]"
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prompt += f" {bot_response}</s> "
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prompt += f"[INST] {message} [/INST]"
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return prompt
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-
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-
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def generate(
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prompt,
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history,
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temperature=0.9,
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max_new_tokens=256,
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top_p=0.95,
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repetition_penalty=1.0,
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):
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temperature = float(temperature)
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if temperature < 1e-2:
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temperature = 1e-2
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top_p = float(top_p)
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-
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generate_kwargs = dict(
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temperature=temperature,
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max_new_tokens=max_new_tokens,
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top_p=top_p,
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repetition_penalty=repetition_penalty,
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do_sample=True,
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seed=42,
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)
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-
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formatted_prompt = format_prompt(prompt, history)
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-
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try:
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stream = text_client.text_generation(
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formatted_prompt,
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**generate_kwargs,
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stream=True,
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details=True,
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return_full_text=False,
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)
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output = ""
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for response in stream:
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output += response.token.text
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yield output
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-
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except Exception as e:
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if "Too Many Requests" in str(e):
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print("ERROR: Too many requests on mistral client")
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gr.Warning("Unfortunately Mistral is unable to process")
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output = "Unfortuanately I am not able to process your request now !"
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else:
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print("Unhandled Exception: ", str(e))
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gr.Warning("Unfortunately Mistral is unable to process")
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output = "I do not know what happened but I could not understand you ."
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-
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return output
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-
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-
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def transcribe(wav_path):
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try:
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# get first element from whisper_jax and strip it to delete begin and end space
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return whisper_client.predict(
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wav_path, # str (filepath or URL to file) in 'inputs' Audio component
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"transcribe", # str in 'Task' Radio component
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False, # return_timestamps=False for whisper-jax https://gist.github.com/sanchit-gandhi/781dd7003c5b201bfe16d28634c8d4cf#file-whisper_jax_endpoint-py
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api_name="/predict",
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)[0].strip()
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except:
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gr.Warning("There was a problem with Whisper endpoint, telling a joke for you.")
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return "There was a problem with my voice, tell me joke"
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-
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-
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# Chatbot demo with multimodal input (text, markdown, LaTeX, code blocks, image, audio, & video). Plus shows support for streaming text.
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-
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-
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def add_text(history, text):
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history = [] if history is None else history
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history = history + [(text, None)]
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return history, gr.update(value="", interactive=False)
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-
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-
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def add_file(history, file):
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history = [] if history is None else history
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-
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try:
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text = transcribe(file)
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print("Transcribed text:", text)
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except Exception as e:
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print(str(e))
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gr.Warning("There was an issue with transcription, please try writing for now")
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# Apply a null text on error
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text = "Transcription seems failed, please tell me a joke about chickens"
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-
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history = history + [(text, None)]
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return history, gr.update(value="", interactive=False)
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-
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-
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##NOTE: not using this as it yields a chacter each time while we need to feed history to TTS
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def bot(history, system_prompt=""):
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history = [] if history is None else history
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-
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if system_prompt == "":
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system_prompt = system_message
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-
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history[-1][1] = ""
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for character in generate(history[-1][0], history[:-1]):
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history[-1][1] = character
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yield history
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-
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-
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def get_latents(speaker_wav):
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# Generate speaker embedding and latents for TTS
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(
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gpt_cond_latent,
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diffusion_conditioning,
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speaker_embedding,
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) = model.get_conditioning_latents(audio_path=speaker_wav)
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return gpt_cond_latent, diffusion_conditioning, speaker_embedding
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-
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-
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latent_map = {}
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latent_map["Female_Voice"] = get_latents("examples/female.wav")
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-
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-
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def get_voice(prompt, language, latent_tuple, suffix="0"):
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gpt_cond_latent, diffusion_conditioning, speaker_embedding = latent_tuple
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# Direct version
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t0 = time.time()
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out = model.inference(
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prompt, language, gpt_cond_latent, speaker_embedding, diffusion_conditioning
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)
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inference_time = time.time() - t0
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print(f"I: Time to generate audio: {round(inference_time*1000)} milliseconds")
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real_time_factor = (time.time() - t0) / out["wav"].shape[-1] * 24000
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print(f"Real-time factor (RTF): {real_time_factor}")
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wav_filename = f"output_{suffix}.wav"
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torchaudio.save(wav_filename, torch.tensor(out["wav"]).unsqueeze(0), 24000)
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return wav_filename
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-
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-
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def wave_header_chunk(frame_input=b"", channels=1, sample_width=2, sample_rate=24000):
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# This will create a wave header then append the frame input
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# It should be first on a streaming wav file
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# Other frames better should not have it (else you will hear some artifacts each chunk start)
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wav_buf = io.BytesIO()
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with wave.open(wav_buf, "wb") as vfout:
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vfout.setnchannels(channels)
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vfout.setsampwidth(sample_width)
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vfout.setframerate(sample_rate)
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vfout.writeframes(frame_input)
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wav_buf.seek(0)
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return wav_buf.read()
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-
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-
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def get_voice_streaming(prompt, language, latent_tuple, suffix="0"):
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gpt_cond_latent, diffusion_conditioning, speaker_embedding = latent_tuple
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try:
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t0 = time.time()
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chunks = model.inference_stream(
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prompt,
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language,
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gpt_cond_latent,
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speaker_embedding,
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)
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-
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first_chunk = True
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for i, chunk in enumerate(chunks):
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if first_chunk:
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first_chunk_time = time.time() - t0
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metrics_text = f"Latency to first audio chunk: {round(first_chunk_time*1000)} milliseconds\n"
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first_chunk = False
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print(f"Received chunk {i} of audio length {chunk.shape[-1]}")
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-
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# In case output is required to be multiple voice files
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# out_file = f'{char}_{i}.wav'
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# write(out_file, 24000, chunk.detach().cpu().numpy().squeeze())
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# audio = AudioSegment.from_file(out_file)
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# audio.export(out_file, format='wav')
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# return out_file
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# directly return chunk as bytes for streaming
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chunk = chunk.detach().cpu().numpy().squeeze()
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chunk = (chunk * 32767).astype(np.int16)
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-
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yield chunk.tobytes()
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-
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except RuntimeError as e:
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if "device-side assert" in str(e):
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# cannot do anything on cuda device side error, need tor estart
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print(
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f"Exit due to: Unrecoverable exception caused by prompt:{sentence}",
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flush=True,
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)
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gr.Warning("Unhandled Exception encounter, please retry in a minute")
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print("Cuda device-assert Runtime encountered need restart")
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# HF Space specific.. This error is unrecoverable need to restart space
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api.restart_space(repo_id=repo_id)
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else:
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print("RuntimeError: non device-side assert error:", str(e))
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gr.Warning("Unhandled Exception encounter, please retry in a minute")
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return None
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return None
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except:
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return None
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-
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-
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def get_sentence(history, system_prompt=""):
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history = [] if history is None else history
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if system_prompt == "":
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system_prompt = system_message
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history[-1][1] = ""
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mistral_start = time.time()
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print("Mistral start")
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sentence_list = []
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sentence_hash_list = []
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text_to_generate = ""
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for character in generate(history[-1][0], history[:-1]):
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history[-1][1] = character
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# It is coming word by word
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text_to_generate = nltk.sent_tokenize(history[-1][1].replace("\n", " ").strip())
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-
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if len(text_to_generate) > 1:
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dif = len(text_to_generate) - len(sentence_list)
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-
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if dif == 1 and len(sentence_list) != 0:
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continue
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sentence = text_to_generate[len(sentence_list)]
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# This is expensive replace with hashing!
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sentence_hash = hash(sentence)
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if sentence_hash not in sentence_hash_list:
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sentence_hash_list.append(sentence_hash)
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sentence_list.append(sentence)
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print("New Sentence: ", sentence)
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yield (sentence, history)
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-
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# return that final sentence token
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# TODO need a counter that one may be replica as before
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last_sentence = nltk.sent_tokenize(history[-1][1].replace("\n", " ").strip())[-1]
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sentence_hash = hash(last_sentence)
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if sentence_hash not in sentence_hash_list:
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sentence_hash_list.append(sentence_hash)
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sentence_list.append(last_sentence)
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print("New Sentence: ", last_sentence)
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-
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yield (last_sentence, history)
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-
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-
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def generate_speech(history):
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language = "en"
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-
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wav_list = []
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for sentence, history in get_sentence(history):
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print(sentence)
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# Sometimes prompt </s> coming on output remove it
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sentence = sentence.replace("</s>", "")
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# A fast fix for last chacter, may produce weird sounds if it is with text
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if sentence[-1] in ["!", "?", ".", ","]:
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# just add a space
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sentence = sentence[:-1] + " " + sentence[-1]
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print("Sentence for speech:", sentence)
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-
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try:
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# generate speech using precomputed latents
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# This is not streaming but it will be fast
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# wav = get_voice(sentence,language, latent_map["Female_Voice"], suffix=len(wav_list))
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audio_stream = get_voice_streaming(
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sentence, language, latent_map["Female_Voice"], suffix=len(wav_list)
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)
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wav_chunks = wave_header_chunk()
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frame_length = 0
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for chunk in audio_stream:
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try:
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wav_chunks += chunk
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frame_length += len(chunk)
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except:
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# hack to continue on playing. sometimes last chunk is empty , will be fixed on next TTS
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continue
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-
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wav_list.append(wav_chunks)
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yield (gr.Audio.update(value=wav_chunks, autoplay=True), history)
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-
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# Streaming wait time calculation
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# audio_length = frame_length / sample_width/ frame_rate
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wait_time = frame_length / 2 / 24000 + 0.5 # plus 500ms
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-
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# for non streaming
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# wait_time= librosa.get_duration(path=wav)
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-
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wait_time = AUDIO_WAIT_MODIFIER * wait_time
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print("Sleeping till audio end")
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time.sleep(wait_time)
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except RuntimeError as e:
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427 |
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if "device-side assert" in str(e):
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# cannot do anything on cuda device side error, need tor estart
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429 |
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print(
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f"Exit due to: Unrecoverable exception caused by prompt:{sentence}",
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flush=True,
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)
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gr.Warning("Unhandled Exception encounter, please retry in a minute")
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434 |
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print("Cuda device-assert Runtime encountered need restart")
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435 |
-
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# HF Space specific.. This error is unrecoverable need to restart space
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api.restart_space(repo_id=repo_id)
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else:
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439 |
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print("RuntimeError: non device-side assert error:", str(e))
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raise e
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441 |
-
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442 |
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# Spoken on autoplay everysencen now produce a concataned one at the one
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443 |
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# requires pip install ffmpeg-python
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444 |
-
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445 |
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# files_to_concat= [ffmpeg.input(w) for w in wav_list]
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# combined_file_name="combined.wav"
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447 |
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# ffmpeg.concat(*files_to_concat,v=0, a=1).output(combined_file_name).run(overwrite_output=True)
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448 |
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# final_audio.update(value=combined_file_name, visible=True)
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449 |
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# yield (combined_file_name, history)
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450 |
-
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451 |
-
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452 |
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css = """
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453 |
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.bot .chatbot p {
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overflow: hidden; /* Ensures the content is not revealed until the animation */
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455 |
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//border-right: .15em solid orange; /* The typwriter cursor */
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white-space: nowrap; /* Keeps the content on a single line */
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margin: 0 auto; /* Gives that scrolling effect as the typing happens */
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458 |
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letter-spacing: .15em; /* Adjust as needed */
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animation:
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460 |
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typing 3.5s steps(40, end);
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461 |
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blink-caret .75s step-end infinite;
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462 |
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}
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463 |
-
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464 |
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/* The typing effect */
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465 |
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@keyframes typing {
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466 |
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from { width: 0 }
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to { width: 100% }
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}
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469 |
-
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470 |
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/* The typewriter cursor effect */
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@keyframes blink-caret {
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from, to { border-color: transparent }
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50% { border-color: orange; }
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474 |
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}
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475 |
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"""
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476 |
-
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477 |
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with gr.Blocks(title=title) as demo:
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478 |
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gr.Markdown(DESCRIPTION)
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479 |
-
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480 |
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chatbot = gr.Chatbot(
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481 |
-
[],
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482 |
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elem_id="chatbot",
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483 |
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avatar_images=("examples/lama.jpeg", "examples/lama2.jpeg"),
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484 |
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bubble_full_width=False,
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)
|
486 |
-
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487 |
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with gr.Row():
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488 |
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txt = gr.Textbox(
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489 |
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scale=3,
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490 |
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show_label=False,
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491 |
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placeholder="Enter text and press enter, or speak to your microphone",
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492 |
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container=False,
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493 |
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)
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494 |
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txt_btn = gr.Button(value="Submit text", scale=1)
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495 |
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btn = gr.Audio(source="microphone", type="filepath", scale=4)
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496 |
-
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497 |
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with gr.Row():
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498 |
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audio = gr.Audio(
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499 |
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label="Generated audio response",
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500 |
-
streaming=False,
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501 |
-
autoplay=False,
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502 |
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interactive=True,
|
503 |
-
show_label=True,
|
504 |
-
)
|
505 |
-
# TODO add a second audio that plays whole sentences (for mobile especially)
|
506 |
-
# final_audio = gr.Audio(label="Final audio response", streaming=False, autoplay=False, interactive=False,show_label=True, visible=False)
|
507 |
-
|
508 |
-
clear_btn = gr.ClearButton([chatbot, audio])
|
509 |
-
|
510 |
-
txt_msg = txt_btn.click(add_text, [chatbot, txt], [chatbot, txt], queue=False).then(
|
511 |
-
generate_speech, chatbot, [audio, chatbot]
|
512 |
-
)
|
513 |
-
|
514 |
-
txt_msg.then(lambda: gr.update(interactive=True), None, [txt], queue=False)
|
515 |
-
|
516 |
-
txt_msg = txt.submit(add_text, [chatbot, txt], [chatbot, txt], queue=False).then(
|
517 |
-
generate_speech, chatbot, [audio, chatbot]
|
518 |
-
)
|
519 |
-
|
520 |
-
txt_msg.then(lambda: gr.update(interactive=True), None, [txt], queue=False)
|
521 |
-
|
522 |
-
file_msg = btn.stop_recording(
|
523 |
-
add_file, [chatbot, btn], [chatbot, txt], queue=False
|
524 |
-
).then(generate_speech, chatbot, [audio, chatbot])
|
525 |
-
|
526 |
-
gr.Markdown(
|
527 |
-
"""
|
528 |
-
This Space demonstrates how to speak to a chatbot, based solely on open-source models.
|
529 |
-
It relies on 3 models:
|
530 |
-
1. [Whisper-large-v2](https://huggingface.co/spaces/sanchit-gandhi/whisper-jax) as an ASR model, to transcribe recorded audio to text. It is called through a [gradio client](https://www.gradio.app/docs/client).
|
531 |
-
2. [Mistral-7b-instruct](https://huggingface.co/spaces/osanseviero/mistral-super-fast) as the chat model, the actual chat model. It is called from [huggingface_hub](https://huggingface.co/docs/huggingface_hub/guides/inference).
|
532 |
-
3. [Coqui's XTTS](https://huggingface.co/spaces/coqui/xtts) as a TTS model, to generate the chatbot answers. This time, the model is hosted locally.
|
533 |
-
|
534 |
-
Note:
|
535 |
-
- By using this demo you agree to the terms of the Coqui Public Model License at https://coqui.ai/cpml"""
|
536 |
-
)
|
537 |
-
demo.queue()
|
538 |
-
demo.launch(debug=True)
|
|
|
68 |
# will use api to restart space on a unrecoverable error
|
69 |
api = HfApi(token=HF_TOKEN)
|
70 |
|
71 |
+
repo_id = "ylacombe/voice-chat-with-mistral"
|
72 |
|
73 |
default_system_message = """
|
74 |
You are Mistral, a large language model trained and provided by Mistral, architecture of you is decoder-based LM. Your voice backend or text to speech TTS backend is provided via Coqui technology. You are right now served on Huggingface spaces.
|
|
|
106 |
"mistralai/Mistral-7B-Instruct-v0.1",
|
107 |
timeout=WHISPER_TIMEOUT,
|
108 |
)
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