raphaelbiojout commited on
Commit ·
a60c8b9
1
Parent(s): f276e75
Encode Base64
Browse files- handler.py +55 -12
handler.py
CHANGED
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@@ -4,6 +4,7 @@ import torch
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import os
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import time
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import json
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import subprocess
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import numpy as np
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@@ -31,29 +32,22 @@ def ffmpeg_read(bpayload: bytes, sampling_rate: int) -> np.array:
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"""
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ar = f"{sampling_rate}"
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ac = "1"
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format_for_conversion = "
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ffmpeg_command = [
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"ffmpeg",
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"-nostdin",
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"-threads",
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"0",
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"-i",
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"pipe:0",
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"-f",
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format_for_conversion,
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"-ac",
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ac,
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"-ar",
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ar,
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"-
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-
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"-hide_banner",
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"-loglevel",
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"quiet",
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"pipe:1",
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]
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-
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-
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try:
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with subprocess.Popen(ffmpeg_command, stdin=subprocess.PIPE, stdout=subprocess.PIPE) as ffmpeg_process:
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@@ -71,6 +65,51 @@ def ffmpeg_read(bpayload: bytes, sampling_rate: int) -> np.array:
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return audio
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class EndpointHandler():
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def __init__(self, path=""):
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# load the model
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@@ -102,7 +141,11 @@ class EndpointHandler():
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print(f"key: {x}, value: {data[x]} ")
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# 1. process input
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audio_nparray = ffmpeg_read(inputs, SAMPLE_RATE)
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# audio_tensor= torch.from_numpy(audio_nparray)
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@@ -110,7 +153,7 @@ class EndpointHandler():
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# 2. transcribe
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device, batch_size, compute_type, whisper_model = whisper_config()
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transcription = self.model.transcribe(audio_nparray, batch_size=batch_size)
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results.append({"transcription": transcription["segments"]})
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logger.info(transcription["segments"])
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import os
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import time
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import json
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import base64
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import subprocess
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import numpy as np
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"""
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ar = f"{sampling_rate}"
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ac = "1"
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format_for_conversion = "f32le"
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ffmpeg_command = [
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"ffmpeg",
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"-i",
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"pipe:0",
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"-ac",
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ac,
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"-ar",
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ar,
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"-f",
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format_for_conversion,
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"-hide_banner",
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"-loglevel",
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"quiet",
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"pipe:1",
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]
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try:
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with subprocess.Popen(ffmpeg_command, stdin=subprocess.PIPE, stdout=subprocess.PIPE) as ffmpeg_process:
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return audio
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# FROM WHISPERX
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def load_audio(file: str, sr: int = SAMPLE_RATE):
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"""
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Open an audio file and read as mono waveform, resampling as necessary
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Parameters
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----------
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file: str
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The audio file to open
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sr: int
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The sample rate to resample the audio if necessary
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Returns
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-------
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A NumPy array containing the audio waveform, in float32 dtype.
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"""
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try:
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# Launches a subprocess to decode audio while down-mixing and resampling as necessary.
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# Requires the ffmpeg CLI to be installed.
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cmd = [
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"ffmpeg",
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"-nostdin",
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"-threads",
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"0",
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"-i",
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file,
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"-f",
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"s16le",
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"-ac",
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"1",
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"-acodec",
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"pcm_s16le",
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"-ar",
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str(sr),
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"-",
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]
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out = subprocess.run(cmd, capture_output=True, check=True).stdout
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except subprocess.CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0
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class EndpointHandler():
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def __init__(self, path=""):
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# load the model
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print(f"key: {x}, value: {data[x]} ")
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# 1. process input
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inputs_encoded = data.pop("inputs", data)
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inputs = base64.b64decode(inputs_encoded)
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print(inputs)
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audio_nparray = ffmpeg_read(inputs, SAMPLE_RATE)
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# audio_tensor= torch.from_numpy(audio_nparray)
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# 2. transcribe
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device, batch_size, compute_type, whisper_model = whisper_config()
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transcription = self.model.transcribe(audio_nparray, batch_size=batch_size,language="fr")
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results.append({"transcription": transcription["segments"]})
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logger.info(transcription["segments"])
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