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new 2312ae4d10 avcodec/ac3dec: avoid coefficient truncation in float
decoder
2312ae4d10 is described below
commit 2312ae4d10e10e315d12797b78155dbc0bc56cea
Author: Ayoub Nabil <[email protected]>
AuthorDate: Tue Aug 4 19:13:32 2026 +0200
Commit: James Almer <[email protected]>
CommitDate: Wed Aug 19 15:29:46 2026 +0000
avcodec/ac3dec: avoid coefficient truncation in float decoder
The float decoder dequantized spectral coefficients into an int32_t
buffer using mantissa >> exponent. This discarded the fractional part
before conversion to float and substantially raised the dither noise
floor for low-level coefficients.
Store decoded coefficients as INTFLOAT and apply 2^-exponent directly
in the floating-point path. Keep the fixed-point decoder unchanged, and
perform coupling, rematrixing and final scaling in the native type.
This also removes the now-unused fmtconvert dependency. Add a synthetic
FATE test that measures the corrected dither level. Existing PCM
references remain within one sample except fate-ac3-2.0, whose measured
maximum difference is two samples.
Fixes #23684.
---
configure | 2 +-
libavcodec/ac3dec.c | 53 ++++++++++++++++++++++++++++++++++-------------------
libavcodec/ac3dec.h | 4 +---
tests/fate/ac3.mak | 40 ++++++++++++++++++++++++++++++++++++++--
4 files changed, 74 insertions(+), 25 deletions(-)
diff --git a/configure b/configure
index f2896b81c7..e2be2ee397 100755
--- a/configure
+++ b/configure
@@ -3094,7 +3094,7 @@ aac_decoder_select="adts_header mpeg4audio sinewin"
aac_fixed_decoder_select="adts_header mpeg4audio"
aac_encoder_select="audio_frame_queue lpc sinewin"
aac_latm_decoder_select="aac_decoder aac_latm_parser"
-ac3_decoder_select="ac3_parser ac3dsp bswapdsp fmtconvert"
+ac3_decoder_select="ac3_parser ac3dsp bswapdsp"
ac3_fixed_decoder_select="ac3_parser ac3dsp bswapdsp"
ac3_encoder_select="ac3dsp audiodsp me_cmp"
ac3_fixed_encoder_select="ac3dsp audiodsp me_cmp"
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 9f85446bf4..79a8751154 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -51,6 +51,14 @@
/** dynamic range table. converts codes to scale factors. */
static float dynamic_range_tab[256];
float ff_ac3_heavy_dynamic_range_tab[256];
+/** scale factor for each decoded exponent: 2^-exp */
+static const float scale_factors[25] = {
+ 0x1p-0f, 0x1p-1f, 0x1p-2f, 0x1p-3f, 0x1p-4f,
+ 0x1p-5f, 0x1p-6f, 0x1p-7f, 0x1p-8f, 0x1p-9f,
+ 0x1p-10f, 0x1p-11f, 0x1p-12f, 0x1p-13f, 0x1p-14f,
+ 0x1p-15f, 0x1p-16f, 0x1p-17f, 0x1p-18f, 0x1p-19f,
+ 0x1p-20f, 0x1p-21f, 0x1p-22f, 0x1p-23f, 0x1p-24f,
+};
/*
* Initialize tables at runtime.
@@ -115,7 +123,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
#if (USE_FIXED)
s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
#else
- ff_fmt_convert_init(&s->fmt_conv);
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
#endif
if (!s->fdsp)
@@ -363,14 +370,22 @@ static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
int band_end = bin + s->cpl_band_sizes[band];
for (ch = 1; ch <= s->fbw_channels; ch++) {
if (s->channel_in_cpl[ch]) {
+#if USE_FIXED
int cpl_coord = s->cpl_coords[ch][band] << 5;
+#else
+ float cpl_coord = s->cpl_coords[ch][band] * (1.0f / (1 << 23));
+#endif
for (bin = band_start; bin < band_end; bin++) {
- s->fixed_coeffs[ch][bin] =
- MULH(s->fixed_coeffs[CPL_CH][bin] * (1 << 4),
cpl_coord);
+#if USE_FIXED
+ s->coeffs[ch][bin] =
+ MULH(s->coeffs[CPL_CH][bin] * (1 << 4), cpl_coord);
+#else
+ s->coeffs[ch][bin] = s->coeffs[CPL_CH][bin] * cpl_coord;
+#endif
}
if (ch == 2 && s->phase_flags[band]) {
for (bin = band_start; bin < band_end; bin++)
- s->fixed_coeffs[2][bin] = -s->fixed_coeffs[2][bin];
+ s->coeffs[2][bin] = -s->coeffs[2][bin];
}
}
}
@@ -390,13 +405,13 @@ typedef struct mant_groups {
int b4;
} mant_groups;
-static av_always_inline int dequantize_coeff(int mantissa, int exponent,
- int coeff_bits)
+static av_always_inline INTFLOAT dequantize_coeff(int mantissa, int exponent,
+ int coeff_bits)
{
#if USE_FIXED
return (mantissa * (1 << coeff_bits)) >> exponent;
#else
- return mantissa >> exponent;
+ return mantissa * scale_factors[exponent];
#endif
}
@@ -420,7 +435,7 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext
*s, int ch_index, ma
int end_freq = s->end_freq[ch_index];
uint8_t *baps = s->bap[ch_index];
int8_t *exps = s->dexps[ch_index];
- int32_t *coeffs = s->fixed_coeffs[ch_index];
+ INTFLOAT *coeffs = s->coeffs[ch_index];
int dither = (ch_index == CPL_CH) || s->dither_flag[ch_index];
#if USE_FIXED
int coeff_bits = fixed_coeff_bits(s);
@@ -516,7 +531,7 @@ static void remove_dithering(AC3DecodeContext *s) {
if (!s->dither_flag[ch] && s->channel_in_cpl[ch]) {
for (i = s->start_freq[CPL_CH]; i < s->end_freq[CPL_CH]; i++) {
if (!s->bap[CPL_CH][i])
- s->fixed_coeffs[ch][i] = 0;
+ s->coeffs[ch][i] = 0;
}
}
}
@@ -534,7 +549,7 @@ static inline void
decode_transform_coeffs_ch(AC3DecodeContext *s, int blk,
if (CONFIG_EAC3_DECODER && !blk)
ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
- s->fixed_coeffs[ch][bin] = dequantize_coeff(
+ s->coeffs[ch][bin] = dequantize_coeff(
s->pre_mantissa[ch][bin][blk], s->dexps[ch][bin], 0);
}
}
@@ -567,7 +582,7 @@ static inline void decode_transform_coeffs(AC3DecodeContext
*s, int blk)
end = s->end_freq[ch];
}
do
- s->fixed_coeffs[ch][end] = 0;
+ s->coeffs[ch][end] = 0;
while (++end < 256);
}
@@ -590,9 +605,9 @@ static void do_rematrixing(AC3DecodeContext *s)
if (s->rematrixing_flags[bnd]) {
bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd + 1]);
for (i = ff_ac3_rematrix_band_tab[bnd]; i < bndend; i++) {
- int tmp0 = s->fixed_coeffs[1][i];
- s->fixed_coeffs[1][i] += s->fixed_coeffs[2][i];
- s->fixed_coeffs[2][i] = tmp0 - s->fixed_coeffs[2][i];
+ INTFLOAT tmp0 = s->coeffs[1][i];
+ s->coeffs[1][i] += s->coeffs[2][i];
+ s->coeffs[2][i] = tmp0 - s->coeffs[2][i];
}
}
}
@@ -1326,16 +1341,16 @@ static int decode_audio_block(AC3DecodeContext *s, int
blk, int offset)
#if USE_FIXED
if (fixed_coeff_bits(s))
- scale_coefs_q2(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain,
+ scale_coefs_q2(s->transform_coeffs[ch], s->coeffs[ch], gain,
256);
else
- scale_coefs(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain,
256);
+ scale_coefs(s->transform_coeffs[ch], s->coeffs[ch], gain, 256);
#else
if (s->target_level != 0)
gain = gain * s->level_gain[audio_channel];
- gain *= 1.0 / 4194304.0f;
- s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch],
- s->fixed_coeffs[ch], gain, 256);
+ gain *= 1.0f / 4194304.0f;
+ s->fdsp->vector_fmul_scalar(s->transform_coeffs[ch], s->coeffs[ch],
+ gain, 256);
#endif
}
diff --git a/libavcodec/ac3dec.h b/libavcodec/ac3dec.h
index c0e6f683ef..25ce1b1c90 100644
--- a/libavcodec/ac3dec.h
+++ b/libavcodec/ac3dec.h
@@ -61,7 +61,6 @@
#include "avcodec.h"
#include "bswapdsp.h"
#include "get_bits.h"
-#include "fmtconvert.h"
#define AC3_OUTPUT_LFEON 8
@@ -83,7 +82,6 @@ typedef struct AC3DecodeContext {
AVFloatDSPContext *fdsp;
#endif
AC3DSPContext ac3dsp;
- FmtConvertContext fmt_conv; ///< optimized conversion functions
///@}
AVTXContext *tx_128, *tx_256;
@@ -249,7 +247,7 @@ typedef struct AC3DecodeContext {
SHORTFLOAT *outptr[AC3_MAX_CHANNELS];
///@name Aligned arrays
- DECLARE_ALIGNED(16, int, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];
///< fixed-point transform coefficients
+ DECLARE_ALIGNED(32, INTFLOAT, coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];
///< decoded transform coefficients
DECLARE_ALIGNED(32, INTFLOAT,
transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< transform
coefficients
DECLARE_ALIGNED(32, INTFLOAT, delay)[EAC3_MAX_CHANNELS][AC3_BLOCK_SIZE];
///< delay - added to the next block
#if USE_FIXED
diff --git a/tests/fate/ac3.mak b/tests/fate/ac3.mak
index 70aa4f4568..c579b65e69 100644
--- a/tests/fate/ac3.mak
+++ b/tests/fate/ac3.mak
@@ -71,6 +71,14 @@ fate-eac3-fixed-dependent-substream: REF =
$(SAMPLES)/eac3/the_great_wall_7.1.pc
$(FATE_AC3) $(FATE_EAC3) $(FATE_EAC3_FIXED): CMP = oneoff
+# the references were generated with the truncating dequantization
+# (mantissa >> exp) that the float decoder used to share with ac3_fixed.
+# the float decoder now dequantizes exactly (mantissa * 2^-exp), which
+# moves its output by up to 1 s16 unit on these streams, except a measured
+# MAXDIFF of 2 on fate-ac3-2.0 and fate-ac3-5.1.
+fate-ac3-2.0: FUZZ = 2
+fate-ac3-5.1: FUZZ = 2
+
FATE_AC3-$(call PCM, AC3, AC3 AC3_FIXED, PCM_S16LE_MUXER ARESAMPLE_FILTER)
+= $(FATE_AC3)
FATE_EAC3-$(call PCM, EAC3, EAC3, PCM_S16LE_MUXER ARESAMPLE_FILTER)
+= $(FATE_EAC3)
FATE_EAC3-$(call PCM, EAC3, EAC3 AC3_FIXED, PCM_S16LE_MUXER ARESAMPLE_FILTER)
+= $(FATE_EAC3_FIXED)
@@ -131,12 +139,40 @@ fate-ac3-fixed-dexp24: CMD = framecrc
-auto_conversion_filters -c ac3_fixed \
-i
$(TARGET_PATH)/tests/data/fate/ac3-fixed-dexp24.ac3 \
-af atrim=start_sample=264192
+# digital silence encoded with the bitexact fixed-point encoder must decode
+# to pure dither noise at the level mandated by the spec (mantissa * 2^-exp).
+# The former truncating dequantization (mantissa >> exp) collapsed the dither
+# mantissas to coarse {-1, 0} steps, raising the decoded noise floor by a
+# factor of ~7 (stddev 18.22 instead of 2.69, tiny_psnr f32 units).
+tests/data/fate/ac3-silence.ac3: TAG = GEN
+tests/data/fate/ac3-silence.ac3: ffmpeg$(PROGSSUF)$(EXESUF) | tests/data/fate
+ $(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
+ -f lavfi -i anullsrc=r=44100:cl=stereo -t 1 \
+ -c:a ac3_fixed -b:a 192k -flags +bitexact -f ac3 -y $(TARGET_PATH)/$@
2>/dev/null
+
+tests/data/fate/ac3-silence.f32: TAG = GEN
+tests/data/fate/ac3-silence.f32: ffmpeg$(PROGSSUF)$(EXESUF) | tests/data/fate
+ $(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
+ -f lavfi -i anullsrc=r=44100:cl=stereo -af atrim=end_sample=44100 \
+ -f f32le -y $(TARGET_PATH)/$@ 2>/dev/null
+
+FATE_AC3_DITHER-$(call ALLYES, FFMPEG LAVFI_INDEV ANULLSRC_FILTER ATRIM_FILTER
\
+ ARESAMPLE_FILTER AC3_FIXED_ENCODER AC3_MUXER \
+ AC3_DEMUXER AC3_DECODER PCM_F32LE_ENCODER \
+ PCM_F32LE_MUXER FILE_PROTOCOL PIPE_PROTOCOL) +=
fate-ac3-float-dither
+fate-ac3-float-dither: tests/data/fate/ac3-silence.ac3
tests/data/fate/ac3-silence.f32
+fate-ac3-float-dither: CMD = ffmpeg -auto_conversion_filters -cons_noisegen 1
-i $(TARGET_PATH)/tests/data/fate/ac3-silence.ac3 -af atrim=end_sample=44100 -f
f32le -
+fate-ac3-float-dither: CMP = stddev
+fate-ac3-float-dither: CMP_UNIT = f32
+fate-ac3-float-dither: REF = tests/data/fate/ac3-silence.f32
+fate-ac3-float-dither: CMP_TARGET = 2.69
+
FATE_EAC3-$(call ALLYES, EAC3_DEMUXER EAC3_MUXER EAC3_CORE_BSF) +=
fate-eac3-core-bsf
fate-eac3-core-bsf: CMD = md5pipe -i
$(TARGET_SAMPLES)/eac3/the_great_wall_7.1.eac3 -c:a copy -bsf:a eac3_core
-fflags +bitexact -f eac3
fate-eac3-core-bsf: CMP = oneline
fate-eac3-core-bsf: REF = b704bf851e99b7442e9bed368b60e6ca
FATE_SAMPLES_AVCONV += $(FATE_AC3-yes) $(FATE_EAC3-yes)
-FATE_FFMPEG += $(FATE_AC3_FIXED_DEXP24-yes)
+FATE_FFMPEG += $(FATE_AC3_DITHER-yes) $(FATE_AC3_FIXED_DEXP24-yes)
-fate-ac3: $(FATE_AC3-yes) $(FATE_EAC3-yes) $(FATE_AC3_FIXED_DEXP24-yes)
+fate-ac3: $(FATE_AC3-yes) $(FATE_EAC3-yes) $(FATE_AC3_DITHER-yes)
$(FATE_AC3_FIXED_DEXP24-yes)
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