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https://github.com/Monadical-SAS/reflector.git
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feat: durable (#794)
* durable (no-mistakes) * hatchet no-mistake * hatchet no-mistake * hatchet no-mistake, better logging * remove conductor and add hatchet tests (no-mistakes) * self-review (no-mistakes) * hatched logs * remove shadow mode for hatchet * and add hatchet processor setting to room * . * cleanup * hatchet init db * self-review (no-mistakes) * self-review (no-mistakes) * hatchet: restore zullip report * self-review round * self-review round * self-review round * dry hatchet with celery * dry hatched with celery - 2 * self-review round * more NES instead of str * self-review wip * self-review round * self-review round * self-review round * can_replay cancelled * add forgotten file * pr autoreviewer fixes * better log webhook events * durable_started return * migration sync * latest changes feature parity * migration merge * pr review --------- Co-authored-by: Igor Loskutov <igor.loskutoff@gmail.com>
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186
server/reflector/utils/audio_padding.py
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186
server/reflector/utils/audio_padding.py
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"""
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Audio track padding utilities.
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Shared PyAV-based functions for extracting stream metadata and applying
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silence padding to audio tracks. Used by both Hatchet workflows and Celery pipelines.
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"""
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import math
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from fractions import Fraction
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import av
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from av.audio.resampler import AudioResampler
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from reflector.utils.audio_constants import (
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OPUS_DEFAULT_BIT_RATE,
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OPUS_STANDARD_SAMPLE_RATE,
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)
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def extract_stream_start_time_from_container(
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container,
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track_idx: int,
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logger=None,
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) -> float:
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"""Extract meeting-relative start time from WebM stream metadata.
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Uses PyAV to read stream.start_time from WebM container.
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More accurate than filename timestamps by ~209ms due to network/encoding delays.
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Args:
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container: PyAV container opened from audio file/URL
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track_idx: Track index for logging context
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logger: Optional logger instance (structlog or stdlib compatible)
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Returns:
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Start time in seconds (0.0 if extraction fails)
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"""
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start_time_seconds = 0.0
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try:
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audio_streams = [s for s in container.streams if s.type == "audio"]
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stream = audio_streams[0] if audio_streams else container.streams[0]
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# 1) Try stream-level start_time (most reliable for Daily.co tracks)
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if stream.start_time is not None and stream.time_base is not None:
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start_time_seconds = float(stream.start_time * stream.time_base)
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# 2) Fallback to container-level start_time (in av.time_base units)
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if (start_time_seconds <= 0) and (container.start_time is not None):
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start_time_seconds = float(container.start_time * av.time_base)
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# 3) Fallback to first packet DTS in stream.time_base
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if start_time_seconds <= 0:
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for packet in container.demux(stream):
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if packet.dts is not None:
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start_time_seconds = float(packet.dts * stream.time_base)
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break
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except Exception as e:
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if logger:
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logger.warning(
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"PyAV metadata read failed; assuming 0 start_time",
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track_idx=track_idx,
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error=str(e),
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)
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start_time_seconds = 0.0
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if logger:
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logger.info(
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f"Track {track_idx} stream metadata: start_time={start_time_seconds:.3f}s",
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track_idx=track_idx,
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)
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return start_time_seconds
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def apply_audio_padding_to_file(
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in_container,
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output_path: str,
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start_time_seconds: float,
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track_idx: int,
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logger=None,
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) -> None:
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"""Apply silence padding to audio track using PyAV filter graph.
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Uses adelay filter to prepend silence, aligning track to meeting start time.
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Output is WebM/Opus format.
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Args:
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in_container: PyAV container opened from source audio
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output_path: Path for output WebM file
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start_time_seconds: Amount of silence to prepend (in seconds)
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track_idx: Track index for logging context
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logger: Optional logger instance (structlog or stdlib compatible)
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Raises:
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Exception: If no audio stream found or PyAV processing fails
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"""
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delay_ms = math.floor(start_time_seconds * 1000)
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if logger:
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logger.info(
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f"Padding track {track_idx} with {delay_ms}ms delay using PyAV",
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track_idx=track_idx,
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delay_ms=delay_ms,
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)
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try:
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with av.open(output_path, "w", format="webm") as out_container:
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in_stream = next(
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(s for s in in_container.streams if s.type == "audio"), None
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)
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if in_stream is None:
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raise Exception("No audio stream in input")
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out_stream = out_container.add_stream(
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"libopus", rate=OPUS_STANDARD_SAMPLE_RATE
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)
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out_stream.bit_rate = OPUS_DEFAULT_BIT_RATE
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graph = av.filter.Graph()
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abuf_args = (
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f"time_base=1/{OPUS_STANDARD_SAMPLE_RATE}:"
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f"sample_rate={OPUS_STANDARD_SAMPLE_RATE}:"
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f"sample_fmt=s16:"
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f"channel_layout=stereo"
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)
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src = graph.add("abuffer", args=abuf_args, name="src")
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aresample_f = graph.add("aresample", args="async=1", name="ares")
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# adelay requires one delay value per channel separated by '|'
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delays_arg = f"{delay_ms}|{delay_ms}"
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adelay_f = graph.add(
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"adelay", args=f"delays={delays_arg}:all=1", name="delay"
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)
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sink = graph.add("abuffersink", name="sink")
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src.link_to(aresample_f)
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aresample_f.link_to(adelay_f)
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adelay_f.link_to(sink)
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graph.configure()
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resampler = AudioResampler(
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format="s16", layout="stereo", rate=OPUS_STANDARD_SAMPLE_RATE
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)
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# Decode -> resample -> push through graph -> encode Opus
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for frame in in_container.decode(in_stream):
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out_frames = resampler.resample(frame) or []
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for rframe in out_frames:
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rframe.sample_rate = OPUS_STANDARD_SAMPLE_RATE
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rframe.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
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src.push(rframe)
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while True:
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try:
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f_out = sink.pull()
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except Exception:
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break
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f_out.sample_rate = OPUS_STANDARD_SAMPLE_RATE
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f_out.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
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for packet in out_stream.encode(f_out):
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out_container.mux(packet)
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# Flush remaining frames from filter graph
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src.push(None)
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while True:
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try:
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f_out = sink.pull()
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except Exception:
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break
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f_out.sample_rate = OPUS_STANDARD_SAMPLE_RATE
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f_out.time_base = Fraction(1, OPUS_STANDARD_SAMPLE_RATE)
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for packet in out_stream.encode(f_out):
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out_container.mux(packet)
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# Flush encoder
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for packet in out_stream.encode(None):
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out_container.mux(packet)
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except Exception as e:
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if logger:
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logger.error(
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"PyAV padding failed for track",
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track_idx=track_idx,
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delay_ms=delay_ms,
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error=str(e),
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exc_info=True,
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)
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raise
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