yjyoon / whisper_streaming_Deprecated star
Tijs Zwinkels 2024-01-24
OpenAI Whisper API backend
@7ed1d4581feb3d70dd5b358bd1754cbef8a434c8
7045d27
7ed1d45
whisper_online.py
--- whisper_online.py
+++ whisper_online.py
@@ -4,6 +4,8 @@
 
 import librosa  
 from functools import lru_cache
 
 import time
 
+import io
 
+import soundfile as sf
@@ -147,6 +149,76 @@
 
     def set_translate_task(self):
 
         self.transcribe_kargs["task"] = "translate"
 
 
+
 
+class OpenaiApiASR(ASRBase):
 
+    """Uses OpenAI's Whisper API for audio transcription."""
 
+
 
+    def __init__(self, modelsize=None, lan=None, cache_dir=None, model_dir=None, response_format="verbose_json", temperature=0):
 
+        self.modelname = "whisper-1"  # modelsize is not used but kept for interface consistency
 
+        self.language = lan  # ISO-639-1 language code
 
+        self.response_format = response_format
 
+        self.temperature = temperature
 
+        self.model = self.load_model(modelsize, cache_dir, model_dir)
 
+
 
+    def load_model(self, *args, **kwargs):
 
+        from openai import OpenAI
 
+        self.client = OpenAI()
 
+        # Since we're using the OpenAI API, there's no model to load locally.
 
+        print("Model configuration is set to use the OpenAI Whisper API.")
 
+
 
+    def ts_words(self, segments):
 
+        o = []
 
+        for segment in segments:
 
+            # Skip segments containing no speech
 
+            if segment["no_speech_prob"] > 0.8:
 
+                continue
 
+
 
+            # Splitting the text into words and filtering out empty strings
 
+            words = [word.strip() for word in segment["text"].split() if word.strip()]
 
+
 
+            if not words:
 
+                continue
 
+
 
+            # Assign start and end times for each word
 
+            # We only have timestamps per segment, so interpolating start and end-times
 
+            # assuming equal duration per word
 
+            segment_duration = segment["end"] - segment["start"]
 
+            duration_per_word = segment_duration / len(words)
 
+            start_time = segment["start"]
 
+            for word in words:
 
+                end_time = start_time + duration_per_word
 
+                o.append((start_time, end_time, word))
 
+                start_time = end_time
 
+
 
+        return o
 
+
 
+
 
+    def segments_end_ts(self, res):
 
+        return [s["end"] for s in res]
 
+
 
+    def transcribe(self, audio_data, prompt=None, *args, **kwargs):
 
+        # Write the audio data to a buffer
 
+        buffer = io.BytesIO()
 
+        buffer.name = "temp.wav"
 
+        sf.write(buffer, audio_data, samplerate=16000, format='WAV', subtype='PCM_16')
 
+        buffer.seek(0)  # Reset buffer's position to the beginning
 
+
 
+        # Prepare transcription parameters
 
+        transcription_params = {
 
+            "model": self.modelname,
 
+            "file": buffer,
 
+            "response_format": self.response_format,
 
+            "temperature": self.temperature
 
+        }
 
+        if self.language:
 
+            transcription_params["language"] = self.language
 
+        if prompt:
 
+            transcription_params["prompt"] = prompt
 
+
 
+        # Perform the transcription
 
+        transcript = self.client.audio.transcriptions.create(**transcription_params)
 
+
 
+        return transcript.segments
 
 
 
 class HypothesisBuffer:
@@ -459,7 +531,7 @@
 
     parser.add_argument('--model_dir', type=str, default=None, help="Dir where Whisper model.bin and other files are saved. This option overrides --model and --model_cache_dir parameter.")
 
     parser.add_argument('--lan', '--language', type=str, default='en', help="Source language code, e.g. en,de,cs, or 'auto' for language detection.")
 
     parser.add_argument('--task', type=str, default='transcribe', choices=["transcribe","translate"],help="Transcribe or translate.")
 
-    parser.add_argument('--backend', type=str, default="faster-whisper", choices=["faster-whisper", "whisper_timestamped"],help='Load only this backend for Whisper processing.')
 
+    parser.add_argument('--backend', type=str, default="faster-whisper", choices=["faster-whisper", "whisper_timestamped", "openai-api"],help='Load only this backend for Whisper processing.')
 
     parser.add_argument('--vad', action="store_true", default=False, help='Use VAD = voice activity detection, with the default parameters.')
 
     parser.add_argument('--buffer_trimming', type=str, default="segment", choices=["sentence", "segment"],help='Buffer trimming strategy -- trim completed sentences marked with punctuation mark and detected by sentence segmenter, or the completed segments returned by Whisper. Sentence segmenter must be installed for "sentence" option.')
 
     parser.add_argument('--buffer_trimming_sec', type=float, default=15, help='Buffer trimming length threshold in seconds. If buffer length is longer, trimming sentence/segment is triggered.')
@@ -499,6 +571,8 @@
 
 
     if args.backend == "faster-whisper":
 
         asr_cls = FasterWhisperASR
 
+    elif args.backend == "openai-api":
 
+        asr_cls = OpenaiApiASR
 
     else:
 
         asr_cls = WhisperTimestampedASR
7045d27
7ed1d45
whisper_online_server.py
--- whisper_online_server.py
+++ whisper_online_server.py
@@ -29,6 +29,8 @@
 
 if args.backend == "faster-whisper":
 
     from faster_whisper import WhisperModel
 
     asr_cls = FasterWhisperASR
 
+elif args.backend == "openai-api":
 
+    asr_cls = OpenaiApiASR
 
 else:
 
     import whisper
 
     import whisper_timestamped
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