Sending entire recordings in one request can exceed provider limits. Decode audio once to a shared PCM sample clock and split uploads at a configurable maximum of 1 to 30 minutes, defaulting to 30 minutes. Prefer nearby pauses and include six seconds of overlap to protect words at audio cuts while keeping each encoded MP3 within the duration cap. Request word timestamps for chunked text and SRT output. Match overlapping words and times to retain each passage once, warn when a join falls back to timestamps, and generate SRT cues on the original recording timeline. Reuse subtitle timestamp validation without reordering regressing words. Only emit output after every request and conversion succeeds. Document the Python dependency, temporary storage, boundary limitations, and lack of automatic retry or resume. Keep single-request text formatting. Test Plan: - python3 -B -m unittest -v: all 29 tests passed, including mocked provider failures, native MP3 payload checks, boundary stitching, and SRT timing. - bash -n transcribe and shellcheck transcribe: passed. - git diff --cached --check: passed. - Locally decoded and encoded the supplied 74-minute WebM into three MP3 chunks; ffprobe confirmed each duration was below 30 minutes. - Live transcription was not verified: the configured 1Password credential was unavailable.
166 lines
6.7 KiB
Python
166 lines
6.7 KiB
Python
"""Plan bounded audio uploads and stitch their overlapping word transcripts."""
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import argparse
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import array
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import json
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import math
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from pathlib import Path
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import sys
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import wave
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from subtitles import convert, normalized_words
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CONTEXT = 3.0 # Audio on each side of a seam: six seconds shared by two uploads.
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ENCODING_MARGIN = 0.1 # MP3 encoder delay and frame padding at 16 kHz.
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PAUSE_SEARCH = 30.0
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MIN_SILENCE = 0.3
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SILENCE_PEAK = 32768 * 10 ** (-35 / 20)
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def pause_before(audio, earliest, latest):
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"""Find the last quiet interval in a small PCM window before the limit."""
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rate = audio.getframerate()
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first, last = round(earliest * rate), round(latest * rate)
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audio.setpos(first)
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samples = array.array("h", audio.readframes(last - first))
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if sys.byteorder != "little":
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samples.byteswap()
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frame = round(0.02 * rate)
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quiet_start = None
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boundary = latest
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for offset in range(0, len(samples) + frame, frame):
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block = samples[offset:offset + frame]
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quiet = bool(block) and max(map(abs, block)) <= SILENCE_PEAK
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if quiet and quiet_start is None:
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quiet_start = offset
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elif not quiet and quiet_start is not None:
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quiet_end = min(offset, len(samples))
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if (quiet_end - quiet_start) / rate >= MIN_SILENCE:
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boundary = (first + (quiet_start + quiet_end) / 2) / rate
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quiet_start = None
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return boundary
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def plan(path, max_seconds=1800):
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if not math.isfinite(max_seconds) or not 60 <= max_seconds <= 1800:
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raise ValueError("Chunk length must be between 1 and 30 minutes")
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with wave.open(str(path), "rb") as audio:
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if (audio.getnchannels(), audio.getsampwidth(), audio.getframerate()) != (1, 2, 16000):
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raise ValueError("Chunk planning requires mono 16 kHz 16-bit PCM WAV")
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duration = audio.getnframes() / audio.getframerate()
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if not duration:
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raise ValueError("Audio file contains no samples")
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chunks = []
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core_start = 0.0
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while core_start < duration:
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start = max(0.0, core_start - CONTEXT)
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if duration - start <= max_seconds - ENCODING_MARGIN:
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core_end = duration
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else:
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latest = start + max_seconds - ENCODING_MARGIN - CONTEXT
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core_end = pause_before(audio, max(core_start, latest - PAUSE_SEARCH), latest)
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end = min(duration, core_end + CONTEXT)
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chunks.append({"start": start, "end": end, "seam": core_start})
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core_start = core_end
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return chunks
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def midpoint(word):
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return (word["start"] + word["end"]) / 2
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def token(word):
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return "".join(c for c in word["word"].casefold() if c.isalnum())
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def stitch(left, right, seam):
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"""Switch transcripts at a shared word, away from either audio cut.
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Match text AND time to avoid deleting intentional repetitions. Prefer an
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anchor with a matching neighbour; accept a single shared word when speech
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is sparse. Timestamp ownership is the fallback if hypotheses disagree.
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"""
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radius = CONTEXT - 0.5
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left_near = [(i, word, token(word)) for i, word in enumerate(left)
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if abs(midpoint(word) - seam) <= radius]
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right_near = [(i, word, token(word)) for i, word in enumerate(right)
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if abs(midpoint(word) - seam) <= radius]
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matches = {}
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for i, a, a_token in left_near:
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for j, b, b_token in right_near:
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difference = abs(midpoint(a) - midpoint(b))
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if a_token and a_token == b_token and difference <= 1.0:
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matches[i, j] = (difference, abs((midpoint(a) + midpoint(b)) / 2 - seam))
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if matches:
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def score(pair):
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i, j = pair
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run = [matches[pair][0]]
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for direction in (-1, 1):
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step = direction
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while (i + step, j + step) in matches:
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run.append(matches[i + step, j + step][0])
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step += direction
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# A whole matching phrase disambiguates "yes yes yes" better than
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# a single word that just happens to be closer to the seam.
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return (-len(run), round(sum(run) / len(run), 3), matches[pair][1])
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i, j = min(matches, key=score)
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return left[:i + 1] + right[j + 1:]
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if left_near or right_near:
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print(
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f"Warning: no shared words near {seam:.2f}s; joining by timestamps. "
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"Review the transcript at this boundary.", file=sys.stderr,
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)
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return ([word for word in left if midpoint(word) < seam] +
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[word for word in right if midpoint(word) >= seam])
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def merge(chunks, responses):
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if len(chunks) != len(responses) or not chunks:
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raise ValueError("Expected one transcription response per chunk")
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merged = []
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for index, (chunk, response) in enumerate(zip(chunks, responses)):
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try:
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words = normalized_words(response, "Chunk stitching")
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for word in words:
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if word["end"] > chunk["end"] - chunk["start"] + 1.0:
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raise ValueError("Word timestamp exceeds the chunk duration")
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word["start"] += chunk["start"]
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word["end"] += chunk["start"]
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if word.get("speaker") is not None:
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# Speaker IDs belong to individual requests, not the file.
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word["speaker"] = (index, word["speaker"])
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except ValueError as error:
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raise ValueError(f"Chunk {index + 1}: {error}") from error
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merged = stitch(merged, words, chunk["seam"]) if index else words
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return {"text": " ".join(word["word"] for word in merged), "words": merged}
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def main():
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parser = argparse.ArgumentParser(description=__doc__)
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commands = parser.add_subparsers(dest="command", required=True)
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planner = commands.add_parser("plan")
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planner.add_argument("audio")
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planner.add_argument("minutes", type=int)
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merger = commands.add_parser("merge")
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merger.add_argument("manifest", type=Path)
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merger.add_argument("--srt", action="store_true")
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args = parser.parse_args()
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try:
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if args.command == "plan":
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print(json.dumps(plan(args.audio, args.minutes * 60)))
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else:
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chunks = json.loads(args.manifest.read_text(encoding="utf-8"))
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responses = [json.loads(
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(args.manifest.parent / f"response-{i}.json").read_text(encoding="utf-8")
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) for i in range(len(chunks))]
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response = merge(chunks, responses)
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sys.stdout.write(convert(response) if args.srt else response["text"] + "\n")
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except (ValueError, OSError, wave.Error) as error:
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parser.exit(1, f"Audio chunking failed: {error}\n")
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if __name__ == "__main__":
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main()
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