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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 07:11:50 PM »It's been suggested to me by someone at Audacity that I use foobar2000 for conversions. There IS an AC3 decoder avaiable for it.
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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 04:25:35 PM »I AM targetting high-end playback equipment and want best possible input. Source is 48k AC3, but have to have 44k WAV. So I guess the question is open.
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Windows / Re: Audio Resample algorithm question
« Last post by eumagga0x2a on December 14, 2018, 03:59:27 PM »It is good enough for practical use cases. For audiophiles, 16 bit at 44100 Hz (the audio CD quality) is a disaster. It is still good enough unless you use high end audio equipment :-)
(I had greatest difficulty to discern 24 bit / 192 kHz audio from 16 bit / 44.1 kHz on studio equipment worth many K €.)
(I had greatest difficulty to discern 24 bit / 192 kHz audio from 16 bit / 44.1 kHz on studio equipment worth many K €.)
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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 03:44:45 PM »So my question remains (please forgive if I'm not "getting it"); is this decoding of the AC3 stream by Avidemux the best possible (or reasonably best possible) quality?
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Windows / Re: Audio Resample algorithm question
« Last post by eumagga0x2a on December 14, 2018, 03:42:21 PM »Avidemux requests float from the decoder so that the bit depth of the source audio doesn't matter. Other programs may handle audio sample format differently.
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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 03:31:25 PM »I"m confused. Media info shows me that the AC3 stream in my video is 16-bit. So apparently you are saying the AviD changes to 32float in the process of outputting 16-bit PCM. Are you saying some other AC3-to-WAV tool might not do that? (I don't know enough about AC3 to intuitively know.)
(I can't use the original AC3 for where I need the audio to go. Plus I have to do editing and volume modifications along the way, therefore the need to get it to a simple .wav format.)
(I can't use the original AC3 for where I need the audio to go. Plus I have to do editing and volume modifications along the way, therefore the need to get it to a simple .wav format.)
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Windows / Re: Audio Resample algorithm question
« Last post by eumagga0x2a on December 14, 2018, 03:17:02 PM »Avidemux uses 32 bit float internally for audio. It is always converted to 16 bit LE when saving (decoded) PCM audio or passing it to the sound system, so bit depth is modified. You can avoid it only by doing all audio conversion steps (why not keeping the original AC3 stream?) in a different program.
I have no idea whether the benefit justifies the effort.
I have no idea whether the benefit justifies the effort.
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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 03:01:12 PM »I have found (and am waiting on confirmation now) that supposedly dithering is used only when changing the bit-depth, not the sample rate. I'm using 16-bit all the way.QuoteI'm hoping to hear that "all AC3 decoding gives the exact same results no matter who does it"At least as long you don't apply dithering, necessary for sample format conversion. But I really don't know, I assume the real-life answer is "No".
So assuming 16-bit all the way, what should we look for when hunting for the best possible AC3-to-WAV conversion? (And can we trust that Avidemux does it?)
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Windows / Re: Audio Resample algorithm question
« Last post by TCmullet on December 14, 2018, 02:56:30 PM »https://github.com/erikd/libsamplerate/blob/master/src/src_sinc.c#L120121 db signal to noise ratio? I've never heard of one that high. But I was assuming that "quality" is defined in more terms than merely SNR.
You get 121 dB SNR with the medium preset vs 144 dB SNR with the best quality one. Is 121 dB SNR not good enough?
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Windows / Re: Audio Resample algorithm question
« Last post by eumagga0x2a on December 14, 2018, 02:47:57 PM »https://github.com/erikd/libsamplerate/blob/master/src/src_sinc.c#L120
You get 121 dB SNR with the medium preset vs 144 dB SNR with the best quality one. Is 121 dB SNR not good enough?
For AC3, Avidemux uses the internal or an external a52dec (32 bit float mode) and converts the samples to 16 bit low endian when it outputs or saves audio. For E-AC3, Avidemux uses the internal FFmpeg (libavcodec).
At least as long you don't apply dithering, necessary for sample format conversion. But I really don't know, I assume the real-life answer is "No".
You get 121 dB SNR with the medium preset vs 144 dB SNR with the best quality one. Is 121 dB SNR not good enough?
For AC3, Avidemux uses the internal or an external a52dec (32 bit float mode) and converts the samples to 16 bit low endian when it outputs or saves audio. For E-AC3, Avidemux uses the internal FFmpeg (libavcodec).
Quote
I'm hoping to hear that "all AC3 decoding gives the exact same results no matter who does it"
At least as long you don't apply dithering, necessary for sample format conversion. But I really don't know, I assume the real-life answer is "No".