Shorten audio window to 10 seconds

This helps with temporal stability in long-running transcriptions, and
lets us get rid of that hack where we refuse to update old pages.

3 files changed, 4 insertions, 3 deletions

diff --git a/Images/speech_to_text_demo.gif b/Images/speech_to_text_demo.gif
index 509f26b..661d80b 100644
--- a/Images/speech_to_text_demo.gif
+++ b/Images/speech_to_text_demo.gif
diff --git a/osc_ctrl.py b/osc_ctrl.py
index ea0c145..c72aed6 100644
--- a/osc_ctrl.py
+++ b/osc_ctrl.py
@@ -250,7 +250,7 @@ def sendMessageLazy(client, msg, tx_state):
         last_cell = (len(tx_state.last_msg_encoded) / NUM_LAYERS) - 1
         last_page = floor(last_cell / (2 ** generate_utils.INDEX_BITS))
         if page < last_page:
-            continue
+            pass
 
         if cell_msg == [state.encoding[' ']] * NUM_LAYERS:
             if empty_cells_sent >= tx_state.empty_cells_to_send_per_call:
diff --git a/transcribe.py b/transcribe.py
index 91fcd54..b316014 100644
--- a/transcribe.py
+++ b/transcribe.py
@@ -33,7 +33,8 @@ class AudioState:
 
     # The maximum length that recordAudio() will put into frames before it
     # starts dropping from the start.
-    MAX_LENGTH_S = 30
+    MAX_LENGTH_S = 10
+    MAX_LENGTH_S_WHISPER = 30
     # The minimum length that recordAudio() will wait for before saving audio.
     MIN_LENGTH_S = 1
 
@@ -202,7 +203,7 @@ def transcribe(audio_state, model, filename):
     audio_state.transcribe_lock.release()
 
     audio = whisper.pad_or_trim(audio, length = audio_state.RATE *
-            audio_state.MAX_LENGTH_S)
+            audio_state.MAX_LENGTH_S_WHISPER)
     mel = whisper.log_mel_spectrogram(audio).to(model.device)
 
     result = None