Reverts the previous commit's always-on measurement. Decoding audio for every
monitor that has it spends CPU on a number almost nothing reads, which is the
wrong trade even though it did solve the chicken and egg of picking a
threshold without ever seeing a level.
Measure when something is actually going to use the reading instead:
- AudioDetection is on, as before, so nothing changes for a monitor that
scores on audio; or
- somebody asked. SharedData gains audio_level_until, a wall clock second
the capture thread keeps measuring up to. The monitor editor's new level
meter pushes it forward while it is on screen and the measurement lapses a
few seconds after the page is left, so nothing has to send a stop and a
crashed browser cannot leave a monitor decoding forever.
When the reading stops being wanted the decoder is released and the published
level and peak are cleared, so a stale number is not left looking current and
an old peak does not land on the next frame row written.
audio_level_until is carved out of analysis_pad rather than appended, so
SharedData stays 888 bytes and no existing offset moves; the static_asserts,
Memory.pm and Monitor.php are updated together and all three now agree the
field is at +880.
The meter itself is on the audio settings, shown whether or not
AudioDetection is checked, because the level is what you need in order to
choose a threshold. It draws the threshold currently in the input as a mark on
the bar so a reading can be judged against it before saving, and a monitor
whose zmc is not running reads "no reading" rather than a confident 0, which
would be indistinguishable from silence.
Frames.AudioLevel is therefore 0 again on monitors that do not score on audio.
That is what the graph already treats as "no audio data", so it draws no line
rather than a flat one.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JpiSWBmtQkR5bcgpHWY4ME
Gating the measurement on AudioDetection made the graph useless for the job
it is most wanted for. AudioThreshold is a per-device number -- the floor on
one camera's mic is nothing like another's -- so it has to be measured before
it can be set, but nothing was measured until it was already set. Enabling
detection with a guessed threshold to find out what the real one should be is
backwards.
The level is now read for every monitor with decodable audio. AudioDetection
governs only whether crossing the threshold contributes a score, which is
what the setting is named for. shared_data->audio_alarm stays 0 when it is
off, so nothing downstream changes for a monitor that does not want audio
alarms.
Nothing here depends on Analysing either. The measurement is in
Monitor::Capture, which runs on whatever Analysing is set to, and frame rows
come from Event::AddFrame, which a continuously recording monitor reaches
through the RECORDING_ALWAYS path with motion detection off. So a monitor
that only records continuously still gets levels on its rows.
Since Monitor::Capture retries Open on every audio packet until it succeeds,
and that now happens for every monitor with audio rather than the handful with
detection on, AudioDetector remembers a codec it has already failed to find a
decoder for. Without it a stream ZoneMinder cannot decode logs a warning at
the audio packet rate for as long as the monitor runs. A reconnect bringing a
different codec is still tried.
The cost is one audio decode per monitor with audio, where before it was one
per monitor with detection enabled. That is small next to the video path, but
it is not nothing on a box with many cameras.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JpiSWBmtQkR5bcgpHWY4ME
zm_update-1.39.31.sql gave monitors audio detection, but the level only ever
existed in shared memory, so it was gone the moment the frame passed and there
was nothing for the event view to plot. Add Frames.AudioLevel next to Score, on
the same 0-100 dBFS-derived scale the threshold uses.
What is stored is the peak since the previous row, not the level at the instant
the row was written. Frames rows are written well below the capture rate --
only alarm, bulk and score-increasing frames get one -- so sampling at write
time would drop exactly the short loud noises worth seeing on a timeline.
AudioDetector accumulates the peak as it decodes and Event::AddFrame takes it
where the row is built, which clears it so each row covers its own interval.
The Event constructor takes and discards it once, otherwise an event's first
row reports the loudest moment since the previous event ended.
This needs no shared memory change: zma is now an offline re-analysis tool and
the live analysis runs in a thread of zmc, alongside the capture thread that
runs the decoder, so the peak can stay in the AudioDetector. SharedData keeps
its documented 888-byte layout and its fixed offsets.
The frames ajax returns the column, and Score with it. elements in
web/ajax/status.php is a whitelist that never listed Score, which is why the
event view's cue strip has been reading an undefined Score off every frame.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01JpiSWBmtQkR5bcgpHWY4ME
The tests target Catch2 v3 throughout: zm_catch2.h includes catch2/catch_all.hpp,
tests/CMakeLists.txt links Catch2::Catch2WithMain, and zm_config.cpp writes
Catch::Approx. None of those exist in v2. Two v2 spellings were left behind and
they broke the build against the version the suite actually requires.
zm_audio_detector.cpp used the unqualified Approx that v3 moved into the Catch
namespace, so four assertions failed to compile with "'Approx' was not declared
in this scope". Qualify them the way zm_config.cpp already does.
tests/main.cpp held nothing but #define CATCH_CONFIG_MAIN and the umbrella
include. In v2 that defined the entry point; in v3 the macro does nothing and
Catch2WithMain supplies main, so the file compiled to an empty translation unit.
Remove it and drop it from add_executable.
find_package(Catch2 REQUIRED) accepted a v2 install and left it to fail later as
compile errors. Ask for 3.
CI does not cover any of this because it builds with BUILD_TEST_SUITE=0.
Ran the full suite from the tests directory, where the font fixtures resolve:
144 test cases, 12480 assertions, all passing. ctest: 149/149.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
ZoneMinder has carried audio for years without ever listening to it: the
packets go into the event and nothing reads them. AudioDetector decodes
them in the capture thread and reports a 0-100 level, which Analyse()
turns into score alongside motion, ONVIF and Amcrest.
The level is dBFS-derived, not a raw amplitude ratio. Linear RMS is
unusable as a setting: ordinary speech sits at 1-3% of full scale, so
every sound worth catching would be crammed into the bottom two points of
the range and no operator could tune it. Mapping -60..0 dBFS onto 0..100
puts speech around 43 instead.
Three columns on Monitors: AudioDetection to enable it, AudioThreshold
for the level to alarm at, and AudioAlarmScore for what it contributes.
AudioAlarmScore defaults to 9, matching what an ONVIF or Amcrest alarm
already adds. A threshold of 0 means off, so enabling detection without
choosing a threshold cannot alarm on silence -- a plain level >= threshold
test would alarm on every packet in that state.
The level and the alarm flag go in SharedData's two spare bytes, renamed
from reserved1/reserved2. Offsets and sizes are unchanged, so the
888-byte cross-process layout and the Memory.pm and Monitor.php offset
tables all still agree; the readers are renamed in the same commit so the
level is available to the web UI.
The decoder is opened lazily on the first audio packet rather than at
camera setup, so a monitor with detection off never carries one and a
stream that gains audio on reconnect still gets picked up. Scoring reads
the capture thread's most recent level rather than scoring per audio
packet, because the score belongs to a video frame and audio packets do
not arrive in step with them.
Tested: 242 assertions over the pure helpers - the RMS of both sample
formats, the dB scale's monotonicity and endpoints, full-scale clamping
of decoder overshoot, and the threshold-0 case. The repo's Catch2 harness
needs v3 and only v2 is installed here, so tests/zm_audio_detector.cpp
was compiled against a v2 shim to check it, and an identical set of
assertions was run standalone.
Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01UvTCzCbvGt8xKQRNCSA7o8
