mirror of
https://github.com/legop3/MultiRoombaRover.git
synced 2026-09-16 01:21:20 -04:00
112 lines
3.7 KiB
Bash
112 lines
3.7 KiB
Bash
#!/usr/bin/env bash
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set -euo pipefail
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set +H
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ENV_FILE="${VIDEO_ENV_FILE:-/var/lib/roverd/video.env}"
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if [[ ! -f "$ENV_FILE" ]]; then
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echo "Environment file ${ENV_FILE} missing; cannot publish audio" >&2
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exit 1
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fi
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# shellcheck disable=SC1090
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source "$ENV_FILE"
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AUDIO_ENABLE="${AUDIO_ENABLE:-0}"
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if [[ "${AUDIO_ENABLE}" -ne 1 ]]; then
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echo "Audio capture disabled; skipping audio-only publisher" >&2
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exit 0
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fi
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: "${AUDIO_PUBLISH_URL:?AUDIO_PUBLISH_URL not set in ${ENV_FILE}}"
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AUDIO_DEVICE="${AUDIO_DEVICE:-hw:0,0}"
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# The old 65,536-byte ALSA buffer represented about 171 ms before ffmpeg could
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# even publish the microphone audio:
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# 65,536 / (48,000 samples * 2 channels * 4 bytes) = 0.1707 seconds
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# Keep the defaults much smaller because this publisher feeds an interactive
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# rover stream, where late-but-smooth audio is less useful than fresher audio.
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# These remain environment-overridable so a noisy Pi or sound card can be tuned
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# in the field without changing the installed script.
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AUDIO_ALSA_BUFFER_BYTES="${AUDIO_ALSA_BUFFER_BYTES:-16384}"
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AUDIO_ALSA_PERIOD_BYTES="${AUDIO_ALSA_PERIOD_BYTES:-1024}"
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if [[ -n "${FFMPEG_BIN:-}" ]]; then
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FFMPEG_BIN_PATH="$FFMPEG_BIN"
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elif command -v ffmpeg >/dev/null 2>&1; then
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FFMPEG_BIN_PATH="$(command -v ffmpeg)"
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else
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echo "ffmpeg not found; install it via apt install ffmpeg." >&2
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exit 1
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fi
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run_pipeline() {
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local ffmpeg_args=(
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-hide_banner
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-loglevel warning
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-fflags nobuffer
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-rtbufsize 0
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-thread_queue_size 4096
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# Match the raw ALSA stream exactly so ffmpeg does not guess the pipe
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# format and so the published audio stays full-band stereo before the
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# Opus encoder sees it.
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-f s32le
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-ar 48000
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-ac 2
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-i pipe:0
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# Keep the known-required microphone boost, but remove the old
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# resample/downmix filter. That old filter threw away stereo and
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# limited the stream to narrow 16 kHz mono before encoding.
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-af "volume=20dB"
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# Opus supports up to 510 kbps. Using that ceiling keeps the stream at
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# the highest practical quality browsers and MediaMTX can carry without
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# trying to push raw PCM through the live path.
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-c:a libopus
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-b:a 510000
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-ar:a 48000
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-ac:a 2
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# Use the audio profile for quality. Latency is still controlled by the
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# 20 ms Opus frame size and the low-buffering capture/publish options
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# around the encoder.
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-application audio
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-frame_duration 20
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-compression_level 0
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# Mirror the video publisher's MPEG-TS low-latency settings. Without
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# these, ffmpeg is allowed to hold packets for mux timing, which is
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# exactly the wrong tradeoff for live rover feedback.
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-flush_packets 1
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-muxdelay 0
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-muxpreload 0
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-f mpegts
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"${AUDIO_PUBLISH_URL}"
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)
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# The Google Voice HAT microphone path is intentionally fixed instead of
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# configurable. The previous env-driven sample rate/channel knobs made it
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# easy for the rover config and the actual ffmpeg pipeline to drift apart,
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# while the hardware path we install is always 48 kHz stereo capture.
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#
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# The buffer and period are byte counts because arecord interprets -B/-F in
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# microseconds only when the value has an explicit time suffix. Keeping them
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# as byte-sized chunks gives direct control over the capture queue. The new
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# defaults are roughly 43 ms total buffer and 2.7 ms wakeup periods at
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# 48 kHz stereo S32_LE, which removes about 128 ms of avoidable capture
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# latency compared with the old 65,536-byte buffer.
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arecord -D "${AUDIO_DEVICE}" -f S32_LE -c 2 -r 48000 -B "${AUDIO_ALSA_BUFFER_BYTES}" -F "${AUDIO_ALSA_PERIOD_BYTES}" -q -t raw \
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| "${FFMPEG_BIN_PATH}" "${ffmpeg_args[@]}"
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}
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trap 'kill 0 2>/dev/null' EXIT INT TERM
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while true; do
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if run_pipeline; then
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exit 0
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fi
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echo "Audio-only publisher exited arecord=${PIPESTATUS[0]} ffmpeg=${PIPESTATUS[1]}, restarting in 2s..." >&2
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sleep 2
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done
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