Merge pull request #78 from kadirnar/add-test-file

🏄‍♂️ Add test code and Runpod platform support

README.md

@@ -39,11 +39,9 @@ from transformers import BitsAndBytesConfig, HqqConfig
 url = "https://www.youtube.com/watch?v=di3rHkEZuUw"
 audio_path = download_and_convert_to_mp3(url)
 
-quant_config  = HqqConfig(
-    nbits=1,
-    group_size=64,
-    quant_zero=False,
-    quant_scale=False, axis=0) #axis=0 is used by default
+quant_config = HqqConfig(
+    nbits=1, group_size=64, quant_zero=False, quant_scale=False, axis=0
+)  # axis=0 is used by default
 
 
 bnb_config = BitsAndBytesConfig(
@@ -53,14 +51,18 @@ bnb_config = BitsAndBytesConfig(
     bnb_4bit_use_double_quant=True,
 )
 
-pipeline = SpeechToTextPipeline(model_id=distil-whisper/distil-large-v3, quant_config=quant_config) # or bnb_config
+pipeline = SpeechToTextPipeline(
+    model_id="distil-whisper/distil-large-v3", quant_config=quant_config
+)  # or bnb_config
+
 transcript = pipeline(
-    audio_path: str = "test.mp3",
-    chunk_length_s: int = 30,
-    stride_length_s: int = 5,
-    max_new_tokens: int = 128,
-    batch_size: int = 100,
-    language: str = "english",
+    audio_path="test.mp3",
+    chunk_length_s=30,
+    stride_length_s=5,
+    max_new_tokens=128,
+    batch_size=100,
+    language="english",
+    return_timestamps=False,
 )
 
 print(transcript)

requirements.txt

@@ -19,3 +19,4 @@ pre-commit==3.4.0
 autollm>=0.1.9
 speechbrain>=0.5.16
 bitsandbytes
+hqq

scripts/runpod.sh

@@ -0,0 +1,2 @@
+apt-get update
+apt install ffmpeg nvtop -y

whisperplus/pipelines/whisper.py

@@ -1,5 +1,6 @@
 import logging
 
+import torch
 from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
 
 logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(levelname)s - %(message)s')
@@ -8,7 +9,7 @@
 class SpeechToTextPipeline:
     """Class for converting audio to text using a pre-trained speech recognition model."""
 
-    def __init__(self, model_id: str = "openai/whisper-large-v3", quant_config=None):
+    def __init__(self, model_id: str = "distil-whisper/distil-large-v3", quant_config=None):
         self.model = None
         self.device = None
 
@@ -17,21 +18,16 @@ def __init__(self, model_id: str = "openai/whisper-large-v3", quant_config=None)
         else:
             logging.info("Model already loaded.")
 
-    def load_model(self, model_id: str = "openai/whisper-large-v3", quant_config=None):
-        """
-        Loads the pre-trained speech recognition model and moves it to the specified device.
-
-        Args:
-            model_id (str): Identifier of the pre-trained model to be loaded.
-        """
-        logging.info("Loading model...")
+    def load_model(self, model_id: str = "distil-whisper/distil-large-v3", quant_config=None):
         model = AutoModelForSpeechSeq2Seq.from_pretrained(
             model_id,
             quantization_config=quant_config,
             low_cpu_mem_usage=True,
             use_safetensors=True,
             attn_implementation="flash_attention_2",
-        )
+            torch_dtype=torch.float16,
+            device_map='auto',
+            max_memory={0: "24GiB"})
         logging.info("Model loaded successfully.")
 
         processor = AutoProcessor.from_pretrained(model_id)
@@ -41,14 +37,13 @@ def load_model(self, model_id: str = "openai/whisper-large-v3", quant_config=Non
 
     def __call__(
             self,
+            audio_path: str = "test.mp3",
             chunk_length_s: int = 30,
             stride_length_s: int = 5,
-            audio_path: str = "test.mp3",
             max_new_tokens: int = 128,
             batch_size: int = 100,
             language: str = "turkish",
-            return_timestamps: bool = False
-        ):
+            return_timestamps: bool = False):
         """
         Converts audio to text using the pre-trained speech recognition model.
 

whisperplus/test.py

@@ -0,0 +1,39 @@
+import time
+
+import torch
+from pipelines.whisper import SpeechToTextPipeline
+from transformers import BitsAndBytesConfig, HqqConfig
+from utils.download_utils import download_and_convert_to_mp3
+
+url = "https://www.youtube.com/watch?v=di3rHkEZuUw"
+audio_path = download_and_convert_to_mp3(url)
+
+hqq_config = HqqConfig(
+    nbits=1, group_size=64, quant_zero=False, quant_scale=False, axis=0)  # axis=0 is used by default
+
+bnb_config = BitsAndBytesConfig(
+    load_in_4bit=True,
+    bnb_4bit_quant_type="nf4",
+    bnb_4bit_compute_dtype=torch.bfloat16,
+    bnb_4bit_use_double_quant=True,
+)
+model = SpeechToTextPipeline(
+    model_id="distil-whisper/distil-large-v3", quant_config=bnb_config)  # or bnb_config
+
+start_event = torch.cuda.Event(enable_timing=True)
+end_event = torch.cuda.Event(enable_timing=True)
+
+start_event.record()
+transcript = model(
+    audio_path="testv0.mp3",
+    chunk_length_s=30,
+    stride_length_s=5,
+    max_new_tokens=128,
+    batch_size=100,
+    language="english",
+    return_timestamps=False)
+end_event.record()
+
+torch.cuda.synchronize()
+elapsed_time_ms = start_event.elapsed_time(end_event)
+print(f"Execution time: {elapsed_time_ms}ms")