checks and changes in logging

- don't set the level for submodules, it's too verbose
- etc.

README.md

@@ -3,42 +3,50 @@ 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
-
-Demo video: https://player.vimeo.com/video/840442741
+[Paper PDF](https://aclanthology.org/2023.ijcnlp-demo.3.pdf), [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
 
-Please, cite us. [Bibtex citation](http://www.afnlp.org/conferences/ijcnlp2023/proceedings/main-demo/cdrom/bib/2023.ijcnlp-demo.3.bib):
+Please, cite us. [ACL Anthology](https://aclanthology.org/2023.ijcnlp-demo.3/), [Bibtex citation](https://aclanthology.org/2023.ijcnlp-demo.3.bib):
 
 ```
-@InProceedings{machacek-dabre-bojar:2023:ijcnlp,
-  author    = {Macháček, Dominik  and  Dabre, Raj  and  Bojar, Ondřej},
-  title     = {Turning Whisper into Real-Time Transcription System},
-  booktitle      = {System Demonstrations},
-  month          = {November},
-  year           = {2023},
-  address        = {Bali, Indonesia},
-  publisher      = {Asian Federation of Natural Language Processing},
-  pages     = {17--24},
+@inproceedings{machacek-etal-2023-turning,
+    title = "Turning Whisper into Real-Time Transcription System",
+    author = "Mach{\'a}{\v{c}}ek, Dominik  and
+      Dabre, Raj  and
+      Bojar, Ond{\v{r}}ej",
+    editor = "Saha, Sriparna  and
+      Sujaini, Herry",
+    booktitle = "Proceedings of the 13th International Joint Conference on Natural Language Processing and the 3rd Conference of the Asia-Pacific Chapter of the Association for Computational Linguistics: System Demonstrations",
+    month = nov,
+    year = "2023",
+    address = "Bali, Indonesia",
+    publisher = "Association for Computational Linguistics",
+    url = "https://aclanthology.org/2023.ijcnlp-demo.3",
+    pages = "17--24",
 }
 ```
 
 ## Installation
 
-1) ``pip install librosa`` -- audio processing library
+1) ``pip install librosa soundfile`` -- audio processing library
 
 2) Whisper backend.
 
-Two alternative backends are integrated. The most recommended one is [faster-whisper](https://github.com/guillaumekln/faster-whisper) with GPU support. Follow their instructions for NVIDIA libraries -- we succeeded with CUDNN 8.5.0 and CUDA 11.7. Install with `pip install faster-whisper`.
+ Several alternative backends are integrated. The most recommended one is [faster-whisper](https://github.com/guillaumekln/faster-whisper) with GPU support. Follow their instructions for NVIDIA libraries -- we succeeded with CUDNN 8.5.0 and CUDA 11.7. Install with `pip install faster-whisper`.
 
 Alternative, less restrictive, but slower backend is [whisper-timestamped](https://github.com/linto-ai/whisper-timestamped): `pip install git+https://github.com/linto-ai/whisper-timestamped`
 
+Thirdly, it's also possible to run this software from the [OpenAI Whisper API](https://platform.openai.com/docs/api-reference/audio/createTranscription). This solution is fast and requires no GPU, just a small VM will suffice, but you will need to pay OpenAI for api access. Also note that, since each audio fragment is processed multiple times, the [price](https://openai.com/pricing) will be higher than obvious from the pricing page, so keep an eye on costs while using. Setting a higher chunk-size will reduce costs significantly. 
+Install with: `pip install openai`
+
+For running with the openai-api backend, make sure that your [OpenAI api key](https://platform.openai.com/api-keys) is set in the `OPENAI_API_KEY` environment variable. For example, before running, do: `export OPENAI_API_KEY=sk-xxx` with *sk-xxx* replaced with your api key. 
+
 The backend is loaded only when chosen. The unused one does not have to be installed.
 
 3) Optional, not recommended: sentence segmenter (aka sentence tokenizer) 
@@ -69,7 +77,7 @@ In case of installation issues of opus-fast-mosestokenizer, especially on Window
 
 ```
 usage: whisper_online.py [-h] [--min-chunk-size MIN_CHUNK_SIZE] [--model {tiny.en,tiny,base.en,base,small.en,small,medium.en,medium,large-v1,large-v2,large-v3,large}] [--model_cache_dir MODEL_CACHE_DIR] [--model_dir MODEL_DIR] [--lan LAN] [--task {transcribe,translate}]
-                         [--backend {faster-whisper,whisper_timestamped}] [--vad] [--buffer_trimming {sentence,segment}] [--buffer_trimming_sec BUFFER_TRIMMING_SEC] [--start_at START_AT] [--offline] [--comp_unaware]
+                         [--backend {faster-whisper,whisper_timestamped,openai-api}] [--vad] [--buffer_trimming {sentence,segment}] [--buffer_trimming_sec BUFFER_TRIMMING_SEC] [--start_at START_AT] [--offline] [--comp_unaware]
                          audio_path
 
 positional arguments:
@@ -86,10 +94,10 @@ options:
   --model_dir MODEL_DIR
                         Dir where Whisper model.bin and other files are saved. This option overrides --model and --model_cache_dir parameter.
   --lan LAN, --language LAN
-                        Language code for transcription, e.g. en,de,cs.
+                        Source language code, e.g. en,de,cs, or 'auto' for language detection.
   --task {transcribe,translate}
                         Transcribe or translate.
-  --backend {faster-whisper,whisper_timestamped}
+  --backend {faster-whisper,whisper_timestamped,openai-api}
                         Load only this backend for Whisper processing.
   --vad                 Use VAD = voice activity detection, with the default parameters.
   --buffer_trimming {sentence,segment}
@@ -147,7 +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
@@ -175,7 +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:
 
@@ -216,12 +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
 

line_packet.py

@@ -2,8 +2,6 @@
 
 """Functions for sending and receiving individual lines of text over a socket.
 
-Used by marian-server-server.py to communicate with the Marian worker.
-
 A line is transmitted using one or more fixed-size packets of UTF-8 bytes
 containing:
 
@@ -11,6 +9,7 @@ containing:
 
   - Zero or more \0 bytes as required to pad the packet to PACKET_SIZE
 
+Originally from the UEDIN team of the ELITR project. 
 """
 
 PACKET_SIZE = 65536

whisper_online.py

@@ -4,12 +4,18 @@ import numpy as np
 import librosa  
 from functools import lru_cache
 import time
+import logging
 
 
+import io
+import soundfile as sf
+import math
+
+logger = logging.getLogger(__name__)
 
 @lru_cache
 def load_audio(fname):
-    a, _ = librosa.load(fname, sr=16000)
+    a, _ = librosa.load(fname, sr=16000, dtype=np.float32)
     return a
 
 def load_audio_chunk(fname, beg, end):
@@ -30,7 +36,10 @@ class ASRBase:
         self.logfile = logfile
 
         self.transcribe_kargs = {}
-        self.original_language = lan 
+        if lan == "auto":
+            self.original_language = None
+        else:
+            self.original_language = lan
 
         self.model = self.load_model(modelsize, cache_dir, model_dir)
 
@@ -54,10 +63,11 @@ class WhisperTimestampedASR(ASRBase):
 
     def load_model(self, modelsize=None, cache_dir=None, model_dir=None):
         import whisper
+        import whisper_timestamped
         from whisper_timestamped import transcribe_timestamped
         self.transcribe_timestamped = transcribe_timestamped
         if model_dir is not None:
-            print("ignoring model_dir, not implemented",file=self.logfile)
+            logger.debug("ignoring model_dir, not implemented")
         return whisper.load_model(modelsize, download_root=cache_dir)
 
     def transcribe(self, audio, init_prompt=""):
@@ -96,8 +106,9 @@ class FasterWhisperASR(ASRBase):
 
     def load_model(self, modelsize=None, cache_dir=None, model_dir=None):
         from faster_whisper import WhisperModel
+#        logging.getLogger("faster_whisper").setLevel(logger.level)
         if model_dir is not None:
-            print(f"Loading whisper model from model_dir {model_dir}. modelsize and cache_dir parameters are not used.",file=self.logfile)
+            logger.debug(f"Loading whisper model from model_dir {model_dir}. modelsize and cache_dir parameters are not used.")
             model_size_or_path = model_dir
         elif modelsize is not None:
             model_size_or_path = modelsize
@@ -118,8 +129,11 @@ class FasterWhisperASR(ASRBase):
         return model
 
     def transcribe(self, audio, init_prompt=""):
+
         # tested: beam_size=5 is faster and better than 1 (on one 200 second document from En ESIC, min chunk 0.01)
         segments, info = self.model.transcribe(audio, language=self.original_language, initial_prompt=init_prompt, beam_size=5, word_timestamps=True, condition_on_previous_text=True, **self.transcribe_kargs)
+        #print(info)  # info contains language detection result
+
         return list(segments)
 
     def ts_words(self, segments):
@@ -142,6 +156,93 @@ class FasterWhisperASR(ASRBase):
         self.transcribe_kargs["task"] = "translate"
 
 
+class OpenaiApiASR(ASRBase):
+    """Uses OpenAI's Whisper API for audio transcription."""
+
+    def __init__(self, lan=None, temperature=0, logfile=sys.stderr):
+        self.logfile = logfile
+
+        self.modelname = "whisper-1"  
+        self.original_language = None if lan == "auto" else lan # ISO-639-1 language code
+        self.response_format = "verbose_json" 
+        self.temperature = temperature
+
+        self.load_model()
+
+        self.use_vad_opt = False
+
+        # reset the task in set_translate_task
+        self.task = "transcribe"
+
+    def load_model(self, *args, **kwargs):
+        from openai import OpenAI
+        self.client = OpenAI()
+
+        self.transcribed_seconds = 0  # for logging how many seconds were processed by API, to know the cost
+        
+
+    def ts_words(self, segments):
+        no_speech_segments = []
+        if self.use_vad_opt:
+            for segment in segments.segments:
+                # TODO: threshold can be set from outside
+                if segment["no_speech_prob"] > 0.8:
+                    no_speech_segments.append((segment.get("start"), segment.get("end")))
+
+        o = []
+        for word in segments.words:
+            start = word.get("start")
+            end = word.get("end")
+            if any(s[0] <= start <= s[1] for s in no_speech_segments):
+                # print("Skipping word", word.get("word"), "because it's in a no-speech segment")
+                continue
+            o.append((start, end, word.get("word")))
+        return o
+
+
+    def segments_end_ts(self, res):
+        return [s["end"] for s in res.words]
+
+    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
+
+        self.transcribed_seconds += math.ceil(len(audio_data)/16000)  # it rounds up to the whole seconds
+
+        params = {
+            "model": self.modelname,
+            "file": buffer,
+            "response_format": self.response_format,
+            "temperature": self.temperature,
+            "timestamp_granularities": ["word", "segment"]
+        }
+        if self.task != "translate" and self.original_language:
+            params["language"] = self.original_language
+        if prompt:
+            params["prompt"] = prompt
+
+        if self.task == "translate":
+            proc = self.client.audio.translations
+        else:
+            proc = self.client.audio.transcriptions
+
+        # Process transcription/translation
+        transcript = proc.create(**params)
+        logger.debug(f"OpenAI API processed accumulated {self.transcribed_seconds} seconds")
+
+        return transcript
+
+    def use_vad(self):
+        self.use_vad_opt = True
+
+    def set_translate_task(self):
+        self.task = "translate"
+
+
+
 
 class HypothesisBuffer:
 
@@ -173,9 +274,11 @@ class HypothesisBuffer:
                         c = " ".join([self.commited_in_buffer[-j][2] for j in range(1,i+1)][::-1])
                         tail = " ".join(self.new[j-1][2] for j in range(1,i+1))
                         if c == tail:
-                            print("removing last",i,"words:",file=self.logfile)
+                            words = []
                             for j in range(i):
-                                print("\t",self.new.pop(0),file=self.logfile)
+                                words.append(repr(self.new.pop(0)))
+                            words_msg = " ".join(words)
+                            logger.debug(f"removing last {i} words: {words_msg}")
                             break
 
     def flush(self):
@@ -234,9 +337,6 @@ class OnlineASRProcessor:
 
         self.transcript_buffer = HypothesisBuffer(logfile=self.logfile)
         self.commited = []
-        self.last_chunked_at = 0
-
-        self.silence_iters = 0
 
     def insert_audio_chunk(self, audio):
         self.audio_buffer = np.append(self.audio_buffer, audio)
@@ -246,7 +346,7 @@ class OnlineASRProcessor:
         "context" is the commited text that is inside the audio buffer. It is transcribed again and skipped. It is returned only for debugging and logging reasons.
         """
         k = max(0,len(self.commited)-1)
-        while k > 0 and self.commited[k-1][1] > self.last_chunked_at:
+        while k > 0 and self.commited[k-1][1] > self.buffer_time_offset:
             k -= 1
 
         p = self.commited[:k]
@@ -267,9 +367,9 @@ class OnlineASRProcessor:
         """
 
         prompt, non_prompt = self.prompt()
-        print("PROMPT:", prompt, file=self.logfile)
-        print("CONTEXT:", non_prompt, file=self.logfile)
-        print(f"transcribing {len(self.audio_buffer)/self.SAMPLING_RATE:2.2f} seconds from {self.buffer_time_offset:2.2f}",file=self.logfile)
+        logger.debug(f"PROMPT: {prompt}")
+        logger.debug(f"CONTEXT: {non_prompt}")
+        logger.debug(f"transcribing {len(self.audio_buffer)/self.SAMPLING_RATE:2.2f} seconds from {self.buffer_time_offset:2.2f}")
         res = self.asr.transcribe(self.audio_buffer, init_prompt=prompt)
 
         # transform to [(beg,end,"word1"), ...]
@@ -278,8 +378,10 @@ class OnlineASRProcessor:
         self.transcript_buffer.insert(tsw, self.buffer_time_offset)
         o = self.transcript_buffer.flush()
         self.commited.extend(o)
-        print(">>>>COMPLETE NOW:",self.to_flush(o),file=self.logfile,flush=True)
-        print("INCOMPLETE:",self.to_flush(self.transcript_buffer.complete()),file=self.logfile,flush=True)
+        completed = self.to_flush(o)
+        logger.debug(f">>>>COMPLETE NOW: {completed}")
+        the_rest = self.to_flush(self.transcript_buffer.complete())
+        logger.debug(f"INCOMPLETE: {the_rest}")
 
         # there is a newly confirmed text
 
@@ -303,18 +405,18 @@ class OnlineASRProcessor:
             #while k>0 and self.commited[k][1] > l:
             #    k -= 1
             #t = self.commited[k][1] 
-            print(f"chunking segment",file=self.logfile)
+            logger.debug("chunking segment")
             #self.chunk_at(t)
 
-        print(f"len of buffer now: {len(self.audio_buffer)/self.SAMPLING_RATE:2.2f}",file=self.logfile)
+        logger.debug(f"len of buffer now: {len(self.audio_buffer)/self.SAMPLING_RATE:2.2f}")
         return self.to_flush(o)
 
     def chunk_completed_sentence(self):
         if self.commited == []: return
-        print(self.commited,file=self.logfile)
+        logger.debug(self.commited)
         sents = self.words_to_sentences(self.commited)
         for s in sents:
-            print("\t\tSENT:",s,file=self.logfile)
+            logger.debug(f"\t\tSENT: {s}")
         if len(sents) < 2:
             return
         while len(sents) > 2:
@@ -322,7 +424,7 @@ class OnlineASRProcessor:
         # we will continue with audio processing at this timestamp
         chunk_at = sents[-2][1]
 
-        print(f"--- sentence chunked at {chunk_at:2.2f}",file=self.logfile)
+        logger.debug(f"--- sentence chunked at {chunk_at:2.2f}")
         self.chunk_at(chunk_at)
 
     def chunk_completed_segment(self, res):
@@ -339,12 +441,12 @@ class OnlineASRProcessor:
                 ends.pop(-1)
                 e = ends[-2]+self.buffer_time_offset
             if e <= t:
-                print(f"--- segment chunked at {e:2.2f}",file=self.logfile)
+                logger.debug(f"--- segment chunked at {e:2.2f}")
                 self.chunk_at(e)
             else:
-                print(f"--- last segment not within commited area",file=self.logfile)
+                logger.debug(f"--- last segment not within commited area")
         else:
-            print(f"--- not enough segments to chunk",file=self.logfile)
+            logger.debug(f"--- not enough segments to chunk")
 
 
 
@@ -357,7 +459,6 @@ class OnlineASRProcessor:
         cut_seconds = time - self.buffer_time_offset
         self.audio_buffer = self.audio_buffer[int(cut_seconds*self.SAMPLING_RATE):]
         self.buffer_time_offset = time
-        self.last_chunked_at = time
 
     def words_to_sentences(self, words):
         """Uses self.tokenizer for sentence segmentation of words.
@@ -391,7 +492,7 @@ class OnlineASRProcessor:
         """
         o = self.transcript_buffer.complete()
         f = self.to_flush(o)
-        print("last, noncommited:",f,file=self.logfile)
+        logger.debug("last, noncommited: {f}")
         return f
 
 
@@ -431,7 +532,7 @@ def create_tokenizer(lan):
 
     # the following languages are in Whisper, but not in wtpsplit:
     if lan in "as ba bo br bs fo haw hr ht jw lb ln lo mi nn oc sa sd sn so su sw tk tl tt".split():
-        print(f"{lan} code is not supported by wtpsplit. Going to use None lang_code option.", file=sys.stderr)
+        logger.debug(f"{lan} code is not supported by wtpsplit. Going to use None lang_code option.")
         lan = None
 
     from wtpsplit import WtP
@@ -451,14 +552,67 @@ def add_shared_args(parser):
     parser.add_argument('--model', type=str, default='large-v2', choices="tiny.en,tiny,base.en,base,small.en,small,medium.en,medium,large-v1,large-v2,large-v3,large".split(","),help="Name size of the Whisper model to use (default: large-v2). The model is automatically downloaded from the model hub if not present in model cache dir.")
     parser.add_argument('--model_cache_dir', type=str, default=None, help="Overriding the default model cache dir where models downloaded from the hub are saved")
     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="Language code for transcription, e.g. en,de,cs.")
+    parser.add_argument('--lan', '--language', type=str, default='auto', 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.')
+    parser.add_argument("-l", "--log-level", dest="log_level", choices=['DEBUG', 'INFO', 'WARNING', 'ERROR', 'CRITICAL'], help="Set the log level", default='DEBUG')
+
+def asr_factory(args, logfile=sys.stderr):
+    """
+    Creates and configures an ASR and ASR Online instance based on the specified backend and arguments.
+    """
+    backend = args.backend
+    if backend == "openai-api":
+        logger.debug("Using OpenAI API.")
+        asr = OpenaiApiASR(lan=args.lan)
+    else:
+        if backend == "faster-whisper":
+            asr_cls = FasterWhisperASR
+        else:
+            asr_cls = WhisperTimestampedASR
+
+        # Only for FasterWhisperASR and WhisperTimestampedASR
+        size = args.model
+        t = time.time()
+        logger.info(f"Loading Whisper {size} model for {args.lan}...")
+        asr = asr_cls(modelsize=size, lan=args.lan, cache_dir=args.model_cache_dir, model_dir=args.model_dir)
+        e = time.time()
+        logger.info(f"done. It took {round(e-t,2)} seconds.")
+
+    # Apply common configurations
+    if getattr(args, 'vad', False):  # Checks if VAD argument is present and True
+        logger.info("Setting VAD filter")
+        asr.use_vad()
+
+    language = args.lan
+    if args.task == "translate":
+        asr.set_translate_task()
+        tgt_language = "en"  # Whisper translates into English
+    else:
+        tgt_language = language  # Whisper transcribes in this language
+
+    # Create the tokenizer
+    if args.buffer_trimming == "sentence":
+        tokenizer = create_tokenizer(tgt_language)
+    else:
+        tokenizer = None
+
+    # Create the OnlineASRProcessor
+    online = OnlineASRProcessor(asr,tokenizer,logfile=logfile,buffer_trimming=(args.buffer_trimming, args.buffer_trimming_sec))
+
+    return asr, online
+
+def set_logging(args,logger,other="_server"):
+    logging.basicConfig(#format='%(name)s 
+            format='%(levelname)s\t%(message)s')
+    logger.setLevel(args.log_level)
+    logging.getLogger("whisper_online"+other).setLevel(args.log_level)
+#    logging.getLogger("whisper_online_server").setLevel(args.log_level)
+
 
-## main:
 
 if __name__ == "__main__":
 
@@ -476,55 +630,28 @@ if __name__ == "__main__":
     logfile = sys.stderr
 
     if args.offline and args.comp_unaware:
-        print("No or one option from --offline and --comp_unaware are available, not both. Exiting.",file=logfile)
+        logger.error("No or one option from --offline and --comp_unaware are available, not both. Exiting.")
         sys.exit(1)
 
+#    if args.log_level:
+#        logging.basicConfig(format='whisper-%(levelname)s:%(name)s: %(message)s',
+#                            level=getattr(logging, args.log_level))
+
+    set_logging(args,logger)
+
     audio_path = args.audio_path
 
     SAMPLING_RATE = 16000
     duration = len(load_audio(audio_path))/SAMPLING_RATE
-    print("Audio duration is: %2.2f seconds" % duration, file=logfile)
+    logger.info("Audio duration is: %2.2f seconds" % duration)
 
-    size = args.model
-    language = args.lan
-
-    t = time.time()
-    print(f"Loading Whisper {size} model for {language}...",file=logfile,end=" ",flush=True)
-
-    if args.backend == "faster-whisper":
-        asr_cls = FasterWhisperASR
-    else:
-        asr_cls = WhisperTimestampedASR
-
-    asr = asr_cls(modelsize=size, lan=language, cache_dir=args.model_cache_dir, model_dir=args.model_dir)
-
-    if args.task == "translate":
-        asr.set_translate_task()
-        tgt_language = "en"  # Whisper translates into English
-    else:
-        tgt_language = language  # Whisper transcribes in this language
-
-
-    e = time.time()
-    print(f"done. It took {round(e-t,2)} seconds.",file=logfile)
-
-    if args.vad:
-        print("setting VAD filter",file=logfile)
-        asr.use_vad()
-
-    
+    asr, online = asr_factory(args, logfile=logfile)
     min_chunk = args.min_chunk_size
-    if args.buffer_trimming == "sentence":
-        tokenizer = create_tokenizer(tgt_language)
-    else:
-        tokenizer = None
-    online = OnlineASRProcessor(asr,tokenizer,logfile=logfile,buffer_trimming=(args.buffer_trimming, args.buffer_trimming_sec))
-
 
     # 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
@@ -543,16 +670,16 @@ if __name__ == "__main__":
             print("%1.4f %1.0f %1.0f %s" % (now*1000, o[0]*1000,o[1]*1000,o[2]),file=logfile,flush=True)
             print("%1.4f %1.0f %1.0f %s" % (now*1000, o[0]*1000,o[1]*1000,o[2]),flush=True)
         else:
-            print(o,file=logfile,flush=True)
+            # No text, so no output
+            pass
 
     if args.offline: ## offline mode processing (for testing/debugging)
         a = load_audio(audio_path)
         online.insert_audio_chunk(a)
         try:
             o = online.process_iter()
-        except AssertionError:
-            print("assertion error",file=logfile)
-            pass
+        except AssertionError as e:
+            log.error(f"assertion error: {repr(e)}")
         else:
             output_transcript(o)
         now = None
@@ -563,13 +690,13 @@ if __name__ == "__main__":
             online.insert_audio_chunk(a)
             try:
                 o = online.process_iter()
-            except AssertionError:
-                print("assertion error",file=logfile)
+            except AssertionError as e:
+                logger.error(f"assertion error: {repr(e)}")
                 pass
             else:
                 output_transcript(o, now=end)
 
-            print(f"## last processed {end:.2f}s",file=logfile,flush=True)
+            logger.debug(f"## last processed {end:.2f}s")
 
             if end >= duration:
                 break
@@ -595,13 +722,13 @@ if __name__ == "__main__":
 
             try:
                 o = online.process_iter()
-            except AssertionError:
-                print("assertion error",file=logfile)
+            except AssertionError as e:
+                logger.error(f"assertion error: {e}")
                 pass
             else:
                 output_transcript(o)
             now = time.time() - start
-            print(f"## last processed {end:.2f} s, now is {now:.2f}, the latency is {now-end:.2f}",file=logfile,flush=True)
+            logger.debug(f"## last processed {end:.2f} s, now is {now:.2f}, the latency is {now-end:.2f}")
 
             if end >= duration:
                 break

whisper_online_server.py

@@ -4,17 +4,24 @@ from whisper_online import *
 import sys
 import argparse
 import os
+import logging
+import numpy as np
+
+logger = logging.getLogger(__name__)
 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
 add_shared_args(parser)
 args = parser.parse_args()
 
+set_logging(args,logger,other="")
 
 # setting whisper object by args 
 
@@ -22,57 +29,22 @@ SAMPLING_RATE = 16000
 
 size = args.model
 language = args.lan
-
-t = time.time()
-print(f"Loading Whisper {size} model for {language}...",file=sys.stderr,end=" ",flush=True)
-
-if args.backend == "faster-whisper":
-    from faster_whisper import WhisperModel
-    asr_cls = FasterWhisperASR
-else:
-    import whisper
-    import whisper_timestamped
-#    from whisper_timestamped_model import WhisperTimestampedASR
-    asr_cls = WhisperTimestampedASR
-
-asr = asr_cls(modelsize=size, lan=language, cache_dir=args.model_cache_dir, model_dir=args.model_dir)
-
-if args.task == "translate":
-    asr.set_translate_task()
-    tgt_language = "en"
-else:
-    tgt_language = language
-
-e = time.time()
-print(f"done. It took {round(e-t,2)} seconds.",file=sys.stderr)
-
-if args.vad:
-    print("setting VAD filter",file=sys.stderr)
-    asr.use_vad()
-
-
+asr, online = asr_factory(args)
 min_chunk = args.min_chunk_size
 
-if args.buffer_trimming == "sentence":
-    tokenizer = create_tokenizer(tgt_language)
+# 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)
+        logger.info("Whisper is warmed up.")
+    else:
+        logger.critical("The warm up file is not available. "+msg)
+        sys.exit(1)
 else:
-    tokenizer = None
-online = OnlineASRProcessor(asr,tokenizer,buffer_trimming=(args.buffer_trimming, args.buffer_trimming_sec))
-
-
-
-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
-    # warm up the ASR, because the very first transcribe takes much more time than the other
-    asr.transcribe(a)
-else:
-    print("Whisper is not warmed up",file=sys.stderr)
-
-
+    logger.warning(msg)
 
 
 ######### Server objects
@@ -80,9 +52,6 @@ else:
 import line_packet
 import socket
 
-import logging
-
-
 class Connection:
     '''it wraps conn object'''
     PACKET_SIZE = 65536
@@ -130,12 +99,10 @@ class ServerProcessor:
         out = []
         while sum(len(x) for x in out) < self.min_chunk*SAMPLING_RATE:
             raw_bytes = self.connection.non_blocking_receive_audio()
-            print(raw_bytes[:10])
-            print(len(raw_bytes))
             if not raw_bytes:
                 break
             sf = soundfile.SoundFile(io.BytesIO(raw_bytes), channels=1,endian="LITTLE",samplerate=SAMPLING_RATE, subtype="PCM_16",format="RAW")
-            audio, _ = librosa.load(sf,sr=SAMPLING_RATE)
+            audio, _ = librosa.load(sf,sr=SAMPLING_RATE,dtype=np.float32)
             out.append(audio)
         if not out:
             return None
@@ -162,7 +129,7 @@ class ServerProcessor:
             print("%1.0f %1.0f %s" % (beg,end,o[2]),flush=True,file=sys.stderr)
             return "%1.0f %1.0f %s" % (beg,end,o[2])
         else:
-            print(o,file=sys.stderr,flush=True)
+            logger.debug("No text in this segment")
             return None
 
     def send_result(self, o):
@@ -176,14 +143,13 @@ class ServerProcessor:
         while True:
             a = self.receive_audio_chunk()
             if a is None:
-                print("break here",file=sys.stderr)
                 break
             self.online_asr_proc.insert_audio_chunk(a)
             o = online.process_iter()
             try:
                 self.send_result(o)
             except BrokenPipeError:
-                print("broken pipe -- connection closed?",file=sys.stderr)
+                logger.info("broken pipe -- connection closed?")
                 break
 
 #        o = online.finish()  # this should be working
@@ -191,24 +157,19 @@ class ServerProcessor:
 
 
 
-
-# Start logging.
-level = logging.INFO
-logging.basicConfig(level=level, format='whisper-server-%(levelname)s: %(message)s')
-
 # server loop
 
 with socket.socket(socket.AF_INET, socket.SOCK_STREAM) as s:
     s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
     s.bind((args.host, args.port))
     s.listen(1)
-    logging.info('INFO: Listening on'+str((args.host, args.port)))
+    logger.info('Listening on'+str((args.host, args.port)))
     while True:
         conn, addr = s.accept()
-        logging.info('INFO: Connected to client on {}'.format(addr))
+        logger.info('Connected to client on {}'.format(addr))
         connection = Connection(conn)
         proc = ServerProcessor(connection, online, min_chunk)
         proc.process()
         conn.close()
-        logging.info('INFO: Connection to client closed')
-logging.info('INFO: Connection closed, terminating.')
+        logger.info('Connection to client closed')
+logger.info('Connection closed, terminating.')