Spaces:
Sleeping
Sleeping
Commit
·
a1dd728
1
Parent(s):
92d093b
update space
Browse files
app.py
CHANGED
@@ -35,32 +35,35 @@ SAMPLE_RATE = feature_extractor.sampling_rate
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SEED = 42
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def
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#
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accumulated_audio = accumulated_audio.astype(np.int16)
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# Create an audio segment from the complete numpy array
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audio_segment = AudioSegment(
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frame_rate=sampling_rate,
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sample_width=
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channels=1
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)
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# Export to MP3
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mp3_io = io.BytesIO()
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audio_segment.export(mp3_io, format="mp3", bitrate="320k")
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mp3_bytes = mp3_io.getvalue()
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mp3_io.close()
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return mp3_bytes
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def generate_response(audio):
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gr.Info("Transcribing Audio", duration=5)
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question = client.automatic_speech_recognition(audio).text
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@@ -76,7 +79,9 @@ def generate_response(audio):
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@spaces.GPU
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def read_response(answer):
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play_steps = int(frame_rate * play_steps_in_s)
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description = "Jenny speaks at an average pace with a calm delivery in a very confined sounding environment with clear audio quality."
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@@ -97,23 +102,10 @@ def read_response(answer):
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set_seed(SEED)
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thread = Thread(target=model.generate, kwargs=generation_kwargs)
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thread.start()
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# Accumulate audio chunks
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audio_chunks = []
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start = time.time()
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for new_audio in streamer:
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print(f"Sample of length: {round(new_audio.shape[0] / sampling_rate, 2)} seconds after {time.time() - start} seconds")
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# Yield None for the audio to maintain the streaming interface
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yield answer, None
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# Convert the accumulated audio to MP3 at the end
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if audio_chunks:
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final_mp3 = accumulate_and_convert_to_mp3(audio_chunks, sampling_rate)
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yield answer, final_mp3
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sampling_rate = model.audio_encoder.config.sampling_rate
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frame_rate = model.audio_encoder.config.frame_rate
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with gr.Blocks() as block:
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SEED = 42
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def numpy_to_mp3(audio_array, sampling_rate):
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# Normalize audio_array if it's floating-point
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if np.issubdtype(audio_array.dtype, np.floating):
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max_val = np.max(np.abs(audio_array))
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audio_array = (audio_array / max_val) * 32767 # Normalize to 16-bit range
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audio_array = audio_array.astype(np.int16)
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# Create an audio segment from the numpy array
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audio_segment = AudioSegment(
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audio_array.tobytes(),
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frame_rate=sampling_rate,
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sample_width=audio_array.dtype.itemsize,
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channels=1
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)
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# Export the audio segment to MP3 bytes - use a high bitrate to maximise quality
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mp3_io = io.BytesIO()
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audio_segment.export(mp3_io, format="mp3", bitrate="320k")
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# Get the MP3 bytes
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mp3_bytes = mp3_io.getvalue()
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mp3_io.close()
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return mp3_bytes
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sampling_rate = model.audio_encoder.config.sampling_rate
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frame_rate = model.audio_encoder.config.frame_rate
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def generate_response(audio):
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gr.Info("Transcribing Audio", duration=5)
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question = client.automatic_speech_recognition(audio).text
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@spaces.GPU
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def read_response(answer):
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play_steps_in_s = 6.0
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play_steps = int(frame_rate * play_steps_in_s)
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description = "Jenny speaks at an average pace with a calm delivery in a very confined sounding environment with clear audio quality."
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set_seed(SEED)
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thread = Thread(target=model.generate, kwargs=generation_kwargs)
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thread.start()
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start = time.time()
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for new_audio in streamer:
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print(f"Sample of length: {round(new_audio.shape[0] / sampling_rate, 2)} seconds after {time.time() - start} seconds")
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yield answer, numpy_to_mp3(new_audio, sampling_rate=sampling_rate)
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with gr.Blocks() as block:
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