On Fri, Sep 25, 2015 at 04:26:00PM -0500, Kyle Swanson wrote: [...] > This is a feature supported by many other loudness scanners. In the > United States, this filter is used to measure loudness for much of our > distributed public radio content. Our current workaround is to first > determine channel count, and then alter our calls to FFmpeg > accordingly. This is a simple, but quite useful feature to have. > Compare the output of the following commands. The first is our > workaround, and the second is an equivalent command which also > provides an accurate measurement. The advantage to having this built > in is that the same call can be used for multi-channel inputs as well. > > ffmpeg -nostats -i mono.wav -filter_complex > "[0:0][0:0]amerge=inputs=2[aout];[aout]ebur128=framelog=verbose:peak=sample+true" > -f null - > ffmpeg -nostats -i mono.wav -filter_complex ebur128=dualmono=true > ebur128 -f null - > > See also: https://ffmpeg.org/pipermail/ffmpeg-devel/2013-March/139910.html > Also see this blog post: > http://www.producenewmedia.com/podcast-loudness-mono-vs-stereo-perception/
questions:
- what happens if you resample the input (by prefixing ebur128 with
aformat=channel_layout=stereo)?
- if I'm reading the blog correctly ("Note that In my discussions with
leading experts in the space, it has come to my attention that this
approach may not be sustainable. Many pros feel it is the responsibility
of the playback device and/or delivery system to apply the necessary
compensation.") this is not a standard behaviour at all. Do you confirm
this?
--
Clément B.
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