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LibAudio: Prevent overflows during prediction
Saturating arithmetic leads to less screwed up audio in these cases.
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0347d04289
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2 changed files with 30 additions and 4 deletions
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@ -797,16 +797,16 @@ ErrorOr<void, LoaderError> FlacLoaderPlugin::decode_custom_lpc(Vector<i64>& deco
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// approximate the waveform with the predictor
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for (size_t i = subframe.order; i < m_current_frame->sample_count; ++i) {
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// (see below)
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i64 sample = 0;
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Checked<i64> sample = 0;
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for (size_t t = 0; t < subframe.order; ++t) {
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// It's really important that we compute in 64-bit land here.
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// Even though FLAC operates at a maximum bit depth of 32 bits, modern encoders use super-large coefficients for maximum compression.
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// These will easily overflow 32 bits and cause strange white noise that abruptly stops intermittently (at the end of a frame).
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// The simple fix of course is to do intermediate computations in 64 bits.
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// The simple fix of course is to do intermediate computations in 64 bits, but we additionally use saturating arithmetic.
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// These considerations are not in the original FLAC spec, but have been added to the IETF standard: https://datatracker.ietf.org/doc/html/draft-ietf-cellar-flac-03#appendix-A.3
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sample += static_cast<i64>(coefficients[t]) * static_cast<i64>(decoded[i - t - 1]);
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sample.saturating_add(Checked<i64>::saturating_mul(static_cast<i64>(coefficients[t]), static_cast<i64>(decoded[i - t - 1])));
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}
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decoded[i] += sample >> lpc_shift;
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decoded[i] += sample.value() >> lpc_shift;
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}
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return {};
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