Merge branch 'main' into online-from-factory

README.md

@@ -3,14 +3,12 @@ Whisper realtime streaming for long speech-to-text transcription and translation
 
 **Turning Whisper into Real-Time Transcription System**
 
-Demonstration paper, by Dominik Macháček, Raj Dabre, Ondřej Bojar, 2023
+Demonstration paper, by [Dominik Macháček](https://ufal.mff.cuni.cz/dominik-machacek), [Raj Dabre](https://prajdabre.github.io/), [Ondřej Bojar](https://ufal.mff.cuni.cz/ondrej-bojar), 2023
 
-Abstract:    Whisper is one of the recent state-of-the-art multilingual speech recognition and translation models, however, it is not designed for real time transcription. In this paper, we build on top of Whisper and create Whisper-Streaming, an implementation of real-time speech transcription and translation of Whisper-like models. Whisper-Streaming uses local agreement policy with self-adaptive latency to enable streaming transcription. We show that Whisper-Streaming achieves high quality and 3.3 seconds latency on unsegmented long-form speech transcription test set, and we demonstrate its robustness and practical usability as a component in live transcription service at a multilingual conference. 
+Abstract:    Whisper is one of the recent state-of-the-art multilingual speech recognition and translation models, however, it is not designed for real-time transcription. In this paper, we build on top of Whisper and create Whisper-Streaming, an implementation of real-time speech transcription and translation of Whisper-like models. Whisper-Streaming uses local agreement policy with self-adaptive latency to enable streaming transcription. We show that Whisper-Streaming achieves high quality and 3.3 seconds latency on unsegmented long-form speech transcription test set, and we demonstrate its robustness and practical usability as a component in live transcription service at a multilingual conference. 
 
 
-Paper in proceedings: http://www.afnlp.org/conferences/ijcnlp2023/proceedings/main-demo/cdrom/pdf/2023.ijcnlp-demo.3.pdf
+[Paper PDF](https://aclanthology.org/2023.ijcnlp-demo.3.pdf), [Demo video](https://player.vimeo.com/video/840442741)
-
-Demo video: https://player.vimeo.com/video/840442741
 
 [Slides](http://ufallab.ms.mff.cuni.cz/~machacek/pre-prints/AACL23-2.11.2023-Turning-Whisper-oral.pdf) -- 15 minutes oral presentation at IJCNLP-AACL 2023
 
@@ -157,7 +155,7 @@ The code whisper_online.py is nicely commented, read it as the full documentatio
 
 This pseudocode describes the interface that we suggest for your implementation. You can implement any features that you need for your application.
 
-```
+```python
 from whisper_online import *
 
 src_lan = "en"  # source language
@@ -185,7 +183,7 @@ online.init()  # refresh if you're going to re-use the object for the next audio
 
 ### Server -- real-time from mic
 
-`whisper_online_server.py` has the same model options as `whisper_online.py`, plus `--host` and `--port` of the TCP connection. See help message (`-h` option).
+`whisper_online_server.py` has the same model options as `whisper_online.py`, plus `--host` and `--port` of the TCP connection and the `--warmup-file`. See the help message (`-h` option).
 
 Client example:
 
@@ -226,12 +224,20 @@ In more detail: we use the init prompt, we handle the inaccurate timestamps, we
 re-process confirmed sentence prefixes and skip them, making sure they don't
 overlap, and we limit the processing buffer window. 
 
-Contributions are welcome.
-
 ### Performance evaluation
 
 [See the paper.](http://www.afnlp.org/conferences/ijcnlp2023/proceedings/main-demo/cdrom/pdf/2023.ijcnlp-demo.3.pdf)
 
+### Contributions
+
+Contributions are welcome. We acknowledge especially:
+
+- [The GitHub contributors](https://github.com/ufal/whisper_streaming/graphs/contributors) for their pull requests with new features and bugfixes.
+- [The translation of this repo into Chinese.](https://github.com/Gloridust/whisper_streaming_CN)
+- [Ondřej Plátek](https://opla.cz/) for the paper pre-review.
+- [Peter Polák](https://ufal.mff.cuni.cz/peter-polak) for the original idea.
+- The UEDIN team of the [ELITR project](https://elitr.eu) for the original line_packet.py.
+
 
 ## Contact
 

whisper_online.py

@@ -626,7 +626,7 @@ if __name__ == "__main__":
     # load the audio into the LRU cache before we start the timer
     a = load_audio_chunk(audio_path,0,1)
 
-    # warm up the ASR, because the very first transcribe takes much more time than the other
+    # warm up the ASR because the very first transcribe takes much more time than the other
     asr.transcribe(a)
 
     beg = args.start_at

whisper_online_server.py

@@ -10,6 +10,8 @@ parser = argparse.ArgumentParser()
 # server options
 parser.add_argument("--host", type=str, default='localhost')
 parser.add_argument("--port", type=int, default=43007)
+parser.add_argument("--warmup-file", type=str, dest="warmup_file", 
+        help="The path to a speech audio wav file to warm up Whisper so that the very first chunk processing is fast. It can be e.g. https://github.com/ggerganov/whisper.cpp/raw/master/samples/jfk.wav .")
 
 
 # options from whisper_online
@@ -26,18 +28,25 @@ language = args.lan
 asr, online = asr_factory(args)
 min_chunk = args.min_chunk_size
 
-demo_audio_path = "cs-maji-2.16k.wav"
-if os.path.exists(demo_audio_path):
-    # load the audio into the LRU cache before we start the timer
-    a = load_audio_chunk(demo_audio_path,0,1)
 
-    # TODO: it should be tested whether it's meaningful
+if args.buffer_trimming == "sentence":
-    # warm up the ASR, because the very first transcribe takes much more time than the other
+    tokenizer = create_tokenizer(tgt_language)
-    asr.transcribe(a)
 else:
-    print("Whisper is not warmed up",file=sys.stderr)
+    tokenizer = None
-
+online = OnlineASRProcessor(asr,tokenizer,buffer_trimming=(args.buffer_trimming, args.buffer_trimming_sec))
 
+# warm up the ASR because the very first transcribe takes more time than the others. 
+# Test results in https://github.com/ufal/whisper_streaming/pull/81
+msg = "Whisper is not warmed up. The first chunk processing may take longer."
+if args.warmup_file:
+    if os.path.isfile(args.warmup_file):
+        a = load_audio_chunk(args.warmup_file,0,1)
+        asr.transcribe(a)
+        print("INFO: Whisper is warmed up.",file=sys.stderr)
+    else:
+        print("WARNING: The warm up file is not available. "+msg,file=sys.stderr)
+else:
+    print("WARNING: " + msg, file=sys.stderr)
 
 
 ######### Server objects