unnecessary resampling?
Posted: 03 Mar 2016 09:38
Why does VLC use an audio resampler even when the input and output have the same sample rate?
(To prevent excessive CPU usage, I usually keep the 'Audio resampler' preference set to 'Disable', which works fine, so obviously resampling wasn't needed. But just recently I was doing something more complicated than simply playing a music file, which actually needed resampling, so I had to turn it back on. That reminded me of the issue, and I decided to finally ask about it.)
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core debug: selecting program id=0
core debug: looking for decoder module matching "any": 43 candidates
core debug: using decoder module "flac"
core debug: Buffering 0%
flac debug: decode STREAMINFO
flac debug: channels:2 samplerate:96000 bitspersamples:24
flac debug: STREAMINFO decoded
core debug: Buffering 14%
core debug: Buffering 28%
core debug: reusing audio output
directsound debug: Opening DirectSound Audio Output
core debug: Buffering 42%
core debug: Buffering 56%
core debug: Buffering 71%
core debug: Buffering 85%
core debug: Buffering 99%
qt4 debug: IM: Setting an input
core debug: Stream buffering done (341 ms in 2 ms)
directsound debug: Windows speaker config: Stereo and stream has 2 channels, using 2 channels
core debug: output 'f32l' 96000 Hz Stereo frame=1 samples/8 bytes
core debug: looking for audio volume module matching "any": 2 candidates
core debug: using audio volume module "float_mixer"
core debug: input 's32l' 96000 Hz Stereo frame=1 samples/8 bytes
core debug: conversion: 's32l'->'f32l' 96000 Hz->96000 Hz Stereo->Stereo
core debug: looking for audio converter module matching "any": 12 candidates
audio_format debug: s32l->f32l, bits per sample: 32->32
core debug: using audio converter module "audio_format"
core debug: conversion pipeline complete
core debug: looking for audio resampler module matching "any": 3 candidates
core debug: using audio resampler module "samplerate"
core debug: End of audio preroll
core debug: Decoder wait done in 20 ms
(To prevent excessive CPU usage, I usually keep the 'Audio resampler' preference set to 'Disable', which works fine, so obviously resampling wasn't needed. But just recently I was doing something more complicated than simply playing a music file, which actually needed resampling, so I had to turn it back on. That reminded me of the issue, and I decided to finally ask about it.)
-----------------------------------
core debug: selecting program id=0
core debug: looking for decoder module matching "any": 43 candidates
core debug: using decoder module "flac"
core debug: Buffering 0%
flac debug: decode STREAMINFO
flac debug: channels:2 samplerate:96000 bitspersamples:24
flac debug: STREAMINFO decoded
core debug: Buffering 14%
core debug: Buffering 28%
core debug: reusing audio output
directsound debug: Opening DirectSound Audio Output
core debug: Buffering 42%
core debug: Buffering 56%
core debug: Buffering 71%
core debug: Buffering 85%
core debug: Buffering 99%
qt4 debug: IM: Setting an input
core debug: Stream buffering done (341 ms in 2 ms)
directsound debug: Windows speaker config: Stereo and stream has 2 channels, using 2 channels
core debug: output 'f32l' 96000 Hz Stereo frame=1 samples/8 bytes
core debug: looking for audio volume module matching "any": 2 candidates
core debug: using audio volume module "float_mixer"
core debug: input 's32l' 96000 Hz Stereo frame=1 samples/8 bytes
core debug: conversion: 's32l'->'f32l' 96000 Hz->96000 Hz Stereo->Stereo
core debug: looking for audio converter module matching "any": 12 candidates
audio_format debug: s32l->f32l, bits per sample: 32->32
core debug: using audio converter module "audio_format"
core debug: conversion pipeline complete
core debug: looking for audio resampler module matching "any": 3 candidates
core debug: using audio resampler module "samplerate"
core debug: End of audio preroll
core debug: Decoder wait done in 20 ms