* [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder
@ 2024-05-16 10:08 Lynne via ffmpeg-devel
2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays Lynne via ffmpeg-devel
` (7 more replies)
0 siblings, 8 replies; 13+ messages in thread
From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw)
To: ffmpeg-devel; +Cc: Lynne
This commit adds a decoder for the frequency-domain part of USAC.
What works:
- Mono
- Stereo (no prediction)
- Stereo (mid/side coding)
What doesn't:
- Preroll decoding (every single decoder seems faulty or weird?)
- Complex stereo prediction
Known issues:
- Spec incompliance due to the spec being flat-out wrong in 11 cases.
- Spec compliance due to the spec being woefully underspecified.
- Lack of robustness
- A single memory leak in packet fragmentation.
I've reached out to the spec authors to get a conformation.
I'm reasonably happy with the way I integrated it to send it
off for review ahead of spec issue clarification.
Lynne (8):
aacdec: move from scalefactor ranged arrays to flat arrays
aacdec: expose channel layout related functions
aacdec: expose decode_tns
aacdec_dsp: implement 768-point transform and windowing
aactab: add deemphasis tables for USAC
aactab: add tables for the new USAC arithmetic coder
aactab: add new scalefactor offset tables for 96/768pt windows
aacdec: add a decoder for AAC USAC (xHE-AAC)
libavcodec/aac/Makefile | 3 +-
libavcodec/aac/aacdec.c | 364 ++++----
libavcodec/aac/aacdec.h | 211 ++++-
libavcodec/aac/aacdec_ac.c | 224 +++++
libavcodec/aac/aacdec_ac.h | 54 ++
libavcodec/aac/aacdec_dsp_template.c | 162 ++--
libavcodec/aac/aacdec_fixed.c | 2 +
libavcodec/aac/aacdec_float.c | 4 +
libavcodec/aac/aacdec_latm.h | 14 +-
libavcodec/aac/aacdec_lpd.c | 192 ++++
libavcodec/aac/aacdec_lpd.h | 33 +
libavcodec/aac/aacdec_usac.c | 1230 ++++++++++++++++++++++++++
libavcodec/aac/aacdec_usac.h | 47 +
libavcodec/aactab.c | 523 +++++++++++
libavcodec/aactab.h | 15 +
libavcodec/sinewin_fixed_tablegen.c | 2 +
libavcodec/sinewin_fixed_tablegen.h | 4 +
17 files changed, 2850 insertions(+), 234 deletions(-)
create mode 100644 libavcodec/aac/aacdec_ac.c
create mode 100644 libavcodec/aac/aacdec_ac.h
create mode 100644 libavcodec/aac/aacdec_lpd.c
create mode 100644 libavcodec/aac/aacdec_lpd.h
create mode 100644 libavcodec/aac/aacdec_usac.c
create mode 100644 libavcodec/aac/aacdec_usac.h
--
2.43.0.381.gb435a96ce8
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^ permalink raw reply [flat|nested] 13+ messages in thread* [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 18:30 ` Michael Niedermayer 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 2/8] aacdec: expose channel layout related functions Lynne via ffmpeg-devel ` (6 subsequent siblings) 7 siblings, 1 reply; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne AAC uses an unconventional system to send scalefactors (the volume+quantization value for each band). Each window is split into either 1 or 8 blocks (long vs short), and transformed separately from one another, with the coefficients for each being also completely independent. The scalefactors slightly increase from 64 (long) to 128 (short) to accomodate better per-block-per-band volume for each window. To reduce overhead, the codec signals scalefactor sizes in an obtuse way, where each group's scalefactor types are sent via a variable length decoding, with a range. But our decoder was written in a way where those ranges were carried through the entire decoder, and to actually read them you had to use the range. Instead of having a dedicated array with a range for each scalefactor, just let the decoder directly index each scalefactor. This also switches the form of quantized scalefactors to the format the spec uses, where for intensity stereo and regular, scalefactors are stored in a scalefactor - 100 form, rather than as-is. USAC gets rid of the complex scalefactor handling. This commit permits for code sharing between both. --- libavcodec/aac/Makefile | 3 +- libavcodec/aac/aacdec.c | 100 ++++++++++++--------------- libavcodec/aac/aacdec.h | 5 +- libavcodec/aac/aacdec_dsp_template.c | 95 ++++++++++--------------- 4 files changed, 85 insertions(+), 118 deletions(-) diff --git a/libavcodec/aac/Makefile b/libavcodec/aac/Makefile index c3e525d373..8b0bfff3e5 100644 --- a/libavcodec/aac/Makefile +++ b/libavcodec/aac/Makefile @@ -2,6 +2,7 @@ clean:: $(RM) $(CLEANSUFFIXES:%=libavcodec/aac/%) OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ - aac/aacdec_float.o + aac/aacdec_float.o aac/aacdec_usac.o \ + aac/aacdec_ac.o OBJS-$(CONFIG_AAC_FIXED_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ aac/aacdec_fixed.o diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 7457fe6c97..35722f9b9b 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -1412,13 +1412,13 @@ fail: * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_band_types(AACDecContext *ac, enum BandType band_type[120], - int band_type_run_end[120], GetBitContext *gb, - IndividualChannelStream *ics) +static int decode_band_types(AACDecContext *ac, SingleChannelElement *sce, + GetBitContext *gb) { - int g, idx = 0; + IndividualChannelStream *ics = &sce->ics; const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; - for (g = 0; g < ics->num_window_groups; g++) { + + for (int g = 0; g < ics->num_window_groups; g++) { int k = 0; while (k < ics->max_sfb) { uint8_t sect_end = k; @@ -1442,10 +1442,8 @@ static int decode_band_types(AACDecContext *ac, enum BandType band_type[120], return AVERROR_INVALIDDATA; } } while (sect_len_incr == (1 << bits) - 1); - for (; k < sect_end; k++) { - band_type [idx] = sect_band_type; - band_type_run_end[idx++] = sect_end; - } + for (; k < sect_end; k++) + sce->band_type[g*ics->max_sfb + k] = sect_band_type; } } return 0; @@ -1461,69 +1459,59 @@ static int decode_band_types(AACDecContext *ac, enum BandType band_type[120], * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_scalefactors(AACDecContext *ac, int sfo[120], - GetBitContext *gb, - unsigned int global_gain, - IndividualChannelStream *ics, - enum BandType band_type[120], - int band_type_run_end[120]) +static int decode_scalefactors(AACDecContext *ac, SingleChannelElement *sce, + GetBitContext *gb, unsigned int global_gain) { - int g, i, idx = 0; + IndividualChannelStream *ics = &sce->ics; int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 }; int clipped_offset; int noise_flag = 1; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - int run_end = band_type_run_end[idx]; - switch (band_type[idx]) { + + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + switch (sce->band_type[g*ics->max_sfb + sfb]) { case ZERO_BT: - for (; i < run_end; i++, idx++) - sfo[idx] = 0; + sce->sfo[g*ics->max_sfb + sfb] = 0; break; case INTENSITY_BT: /* fallthrough */ case INTENSITY_BT2: - for (; i < run_end; i++, idx++) { - offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; - clipped_offset = av_clip(offset[2], -155, 100); - if (offset[2] != clipped_offset) { - avpriv_request_sample(ac->avctx, - "If you heard an audible artifact, there may be a bug in the decoder. " - "Clipped intensity stereo position (%d -> %d)", - offset[2], clipped_offset); - } - sfo[idx] = clipped_offset; + offset[2] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; + clipped_offset = av_clip(offset[2], -155, 100); + if (offset[2] != clipped_offset) { + avpriv_request_sample(ac->avctx, + "If you heard an audible artifact, there may be a bug in the decoder. " + "Clipped intensity stereo position (%d -> %d)", + offset[2], clipped_offset); } + sce->sfo[g*ics->max_sfb + sfb] = clipped_offset - 100; break; case NOISE_BT: - for (; i < run_end; i++, idx++) { - if (noise_flag-- > 0) - offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE; - else - offset[1] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; - clipped_offset = av_clip(offset[1], -100, 155); - if (offset[1] != clipped_offset) { - avpriv_request_sample(ac->avctx, - "If you heard an audible artifact, there may be a bug in the decoder. " - "Clipped noise gain (%d -> %d)", - offset[1], clipped_offset); - } - sfo[idx] = clipped_offset; + if (noise_flag-- > 0) + offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE; + else + offset[1] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; + clipped_offset = av_clip(offset[1], -100, 155); + if (offset[1] != clipped_offset) { + avpriv_request_sample(ac->avctx, + "If you heard an audible artifact, there may be a bug in the decoder. " + "Clipped noise gain (%d -> %d)", + offset[1], clipped_offset); } + sce->sfo[g*ics->max_sfb + sfb] = clipped_offset; break; default: - for (; i < run_end; i++, idx++) { - offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; - if (offset[0] > 255U) { - av_log(ac->avctx, AV_LOG_ERROR, - "Scalefactor (%d) out of range.\n", offset[0]); - return AVERROR_INVALIDDATA; - } - sfo[idx] = offset[0]; + offset[0] += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; + if (offset[0] > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "Scalefactor (%d) out of range.\n", offset[0]); + return AVERROR_INVALIDDATA; } + sce->sfo[g*ics->max_sfb + sfb] = offset[0] - 100; break; } } } + return 0; } @@ -1680,11 +1668,9 @@ int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, goto fail; } - if ((ret = decode_band_types(ac, sce->band_type, - sce->band_type_run_end, gb, ics)) < 0) + if ((ret = decode_band_types(ac, sce, gb)) < 0) goto fail; - if ((ret = decode_scalefactors(ac, sce->sfo, gb, global_gain, ics, - sce->band_type, sce->band_type_run_end)) < 0) + if ((ret = decode_scalefactors(ac, sce, gb, global_gain)) < 0) goto fail; ac->dsp.dequant_scalefactors(sce); diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index eed53c6c96..7585a4450b 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -146,9 +146,8 @@ typedef struct SingleChannelElement { IndividualChannelStream ics; TemporalNoiseShaping tns; enum BandType band_type[128]; ///< band types - int band_type_run_end[120]; ///< band type run end points - int sfo[120]; ///< scalefactor offsets - INTFLOAT_UNION(sf, [120]); ///< scalefactors + int sfo[128]; ///< scalefactor offsets + INTFLOAT_UNION(sf, [512]); ///< scalefactors (8 windows * 16 sfb max) INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index 621baef8ca..e69970472c 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -41,47 +41,37 @@ static void AAC_RENAME(dequant_scalefactors)(SingleChannelElement *sce) { IndividualChannelStream *ics = &sce->ics; - const enum BandType *band_type = sce->band_type; - const int *band_type_run_end = sce->band_type_run_end; const int *sfo = sce->sfo; INTFLOAT *sf = sce->AAC_RENAME(sf); - int g, i, idx = 0; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { - int run_end = band_type_run_end[idx]; - switch (band_type[idx]) { + int idx = 0; + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++, idx++) { + switch (sce->band_type[g*ics->max_sfb + sfb]) { case ZERO_BT: - for (; i < run_end; i++, idx++) - sf[idx] = FIXR(0.); + sf[idx] = FIXR(0.); break; case INTENSITY_BT: /* fallthrough */ case INTENSITY_BT2: - for (; i < run_end; i++, idx++) { #if USE_FIXED - sf[idx] = 100 - sfo[idx]; + sf[idx] = 100 - (sfo[idx] + 100); #else - sf[idx] = ff_aac_pow2sf_tab[-sfo[idx] + POW_SF2_ZERO]; + sf[idx] = ff_aac_pow2sf_tab[-sfo[idx] - 100 + POW_SF2_ZERO]; #endif /* USE_FIXED */ - } break; case NOISE_BT: - for (; i < run_end; i++, idx++) { #if USE_FIXED - sf[idx] = -(100 + sfo[idx]); + sf[idx] = -(100 + sfo[idx]); #else - sf[idx] = -ff_aac_pow2sf_tab[sfo[idx] + POW_SF2_ZERO]; + sf[idx] = -ff_aac_pow2sf_tab[sfo[idx] + POW_SF2_ZERO]; #endif /* USE_FIXED */ - } break; default: - for (; i < run_end; i++, idx++) { #if USE_FIXED - sf[idx] = -sfo[idx]; + sf[idx] = -sfo[idx] - 100; #else - sf[idx] = -ff_aac_pow2sf_tab[sfo[idx] - 100 + POW_SF2_ZERO]; + sf[idx] = -ff_aac_pow2sf_tab[sfo[idx] + POW_SF2_ZERO]; #endif /* USE_FIXED */ - } break; } } @@ -96,25 +86,23 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement const IndividualChannelStream *ics = &cpe->ch[0].ics; INTFLOAT *ch0 = cpe->ch[0].AAC_RENAME(coeffs); INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); - int g, i, group, idx = 0; const uint16_t *offsets = ics->swb_offset; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb; i++, idx++) { + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + const int idx = g*ics->max_sfb + sfb; if (cpe->ms_mask[idx] && cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { + for (int group = 0; group < ics->group_len[g]; group++) #if USE_FIXED - for (group = 0; group < ics->group_len[g]; group++) { - ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i], - ch1 + group * 128 + offsets[i], - offsets[i+1] - offsets[i]); + ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[sfb], + ch1 + group * 128 + offsets[sfb], + offsets[sfb+1] - offsets[sfb]); #else - for (group = 0; group < ics->group_len[g]; group++) { - ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i], - ch1 + group * 128 + offsets[i], - offsets[i+1] - offsets[i]); + ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[sfb], + ch1 + group * 128 + offsets[sfb], + offsets[sfb+1] - offsets[sfb]); #endif /* USE_FIXED */ - } } } ch0 += ics->group_len[g] * 128; @@ -136,37 +124,30 @@ static void AAC_RENAME(apply_intensity_stereo)(AACDecContext *ac, SingleChannelElement *sce1 = &cpe->ch[1]; INTFLOAT *coef0 = cpe->ch[0].AAC_RENAME(coeffs), *coef1 = cpe->ch[1].AAC_RENAME(coeffs); const uint16_t *offsets = ics->swb_offset; - int g, group, i, idx = 0; int c; INTFLOAT scale; - for (g = 0; g < ics->num_window_groups; g++) { - for (i = 0; i < ics->max_sfb;) { + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + const int idx = g*ics->max_sfb + sfb; if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { - const int bt_run_end = sce1->band_type_run_end[idx]; - for (; i < bt_run_end; i++, idx++) { - c = -1 + 2 * (sce1->band_type[idx] - 14); - if (ms_present) - c *= 1 - 2 * cpe->ms_mask[idx]; - scale = c * sce1->AAC_RENAME(sf)[idx]; - for (group = 0; group < ics->group_len[g]; group++) + c = -1 + 2 * (sce1->band_type[idx] - 14); + if (ms_present) + c *= 1 - 2 * cpe->ms_mask[idx]; + scale = c * sce1->AAC_RENAME(sf)[idx]; + for (int group = 0; group < ics->group_len[g]; group++) #if USE_FIXED - subband_scale(coef1 + group * 128 + offsets[i], - coef0 + group * 128 + offsets[i], - scale, - 23, - offsets[i + 1] - offsets[i] ,ac->avctx); + subband_scale(coef1 + group * 128 + offsets[sfb], + coef0 + group * 128 + offsets[sfb], + scale, + 23, + offsets[sfb + 1] - offsets[sfb], ac->avctx); #else - ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i], - coef0 + group * 128 + offsets[i], - scale, - offsets[i + 1] - offsets[i]); + ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[sfb], + coef0 + group * 128 + offsets[sfb], + scale, + offsets[sfb + 1] - offsets[sfb]); #endif /* USE_FIXED */ - } - } else { - int bt_run_end = sce1->band_type_run_end[idx]; - idx += bt_run_end - i; - i = bt_run_end; } } coef0 += ics->group_len[g] * 128; -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* Re: [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays Lynne via ffmpeg-devel @ 2024-05-16 18:30 ` Michael Niedermayer 2024-05-16 18:46 ` Lynne via ffmpeg-devel 0 siblings, 1 reply; 13+ messages in thread From: Michael Niedermayer @ 2024-05-16 18:30 UTC (permalink / raw) To: FFmpeg development discussions and patches [-- Attachment #1.1: Type: text/plain, Size: 2549 bytes --] On Thu, May 16, 2024 at 12:08:11PM +0200, Lynne via ffmpeg-devel wrote: > AAC uses an unconventional system to send scalefactors > (the volume+quantization value for each band). > Each window is split into either 1 or 8 blocks (long vs short), > and transformed separately from one another, with the coefficients > for each being also completely independent. The scalefactors > slightly increase from 64 (long) to 128 (short) to accomodate > better per-block-per-band volume for each window. > > To reduce overhead, the codec signals scalefactor sizes in an obtuse way, > where each group's scalefactor types are sent via a variable length decoding, > with a range. > But our decoder was written in a way where those ranges were carried through > the entire decoder, and to actually read them you had to use the range. > > Instead of having a dedicated array with a range for each scalefactor, > just let the decoder directly index each scalefactor. > > This also switches the form of quantized scalefactors to the format > the spec uses, where for intensity stereo and regular, scalefactors > are stored in a scalefactor - 100 form, rather than as-is. > > USAC gets rid of the complex scalefactor handling. This commit permits > for code sharing between both. > --- > libavcodec/aac/Makefile | 3 +- > libavcodec/aac/aacdec.c | 100 ++++++++++++--------------- > libavcodec/aac/aacdec.h | 5 +- > libavcodec/aac/aacdec_dsp_template.c | 95 ++++++++++--------------- > 4 files changed, 85 insertions(+), 118 deletions(-) > > diff --git a/libavcodec/aac/Makefile b/libavcodec/aac/Makefile > index c3e525d373..8b0bfff3e5 100644 > --- a/libavcodec/aac/Makefile > +++ b/libavcodec/aac/Makefile > @@ -2,6 +2,7 @@ clean:: > $(RM) $(CLEANSUFFIXES:%=libavcodec/aac/%) > > OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ > - aac/aacdec_float.o > + aac/aacdec_float.o aac/aacdec_usac.o \ > + aac/aacdec_ac.o AR libavcodec/libavcodec.a ar: libavcodec/aac/aacdec_ac.o: No such file or directory make: *** [ffbuild/library.mak:38: libavcodec/libavcodec.a] Error 1 [...] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB In fact, the RIAA has been known to suggest that students drop out of college or go to community college in order to be able to afford settlements. -- The RIAA [-- Attachment #1.2: signature.asc --] [-- Type: application/pgp-signature, Size: 195 bytes --] [-- Attachment #2: Type: text/plain, Size: 251 bytes --] _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* Re: [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays 2024-05-16 18:30 ` Michael Niedermayer @ 2024-05-16 18:46 ` Lynne via ffmpeg-devel 0 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 18:46 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne [-- Attachment #1.1.1.1: Type: text/plain, Size: 2443 bytes --] On 16/05/2024 20:30, Michael Niedermayer wrote: > On Thu, May 16, 2024 at 12:08:11PM +0200, Lynne via ffmpeg-devel wrote: >> AAC uses an unconventional system to send scalefactors >> (the volume+quantization value for each band). >> Each window is split into either 1 or 8 blocks (long vs short), >> and transformed separately from one another, with the coefficients >> for each being also completely independent. The scalefactors >> slightly increase from 64 (long) to 128 (short) to accomodate >> better per-block-per-band volume for each window. >> >> To reduce overhead, the codec signals scalefactor sizes in an obtuse way, >> where each group's scalefactor types are sent via a variable length decoding, >> with a range. >> But our decoder was written in a way where those ranges were carried through >> the entire decoder, and to actually read them you had to use the range. >> >> Instead of having a dedicated array with a range for each scalefactor, >> just let the decoder directly index each scalefactor. >> >> This also switches the form of quantized scalefactors to the format >> the spec uses, where for intensity stereo and regular, scalefactors >> are stored in a scalefactor - 100 form, rather than as-is. >> >> USAC gets rid of the complex scalefactor handling. This commit permits >> for code sharing between both. >> --- >> libavcodec/aac/Makefile | 3 +- >> libavcodec/aac/aacdec.c | 100 ++++++++++++--------------- >> libavcodec/aac/aacdec.h | 5 +- >> libavcodec/aac/aacdec_dsp_template.c | 95 ++++++++++--------------- >> 4 files changed, 85 insertions(+), 118 deletions(-) >> >> diff --git a/libavcodec/aac/Makefile b/libavcodec/aac/Makefile >> index c3e525d373..8b0bfff3e5 100644 >> --- a/libavcodec/aac/Makefile >> +++ b/libavcodec/aac/Makefile >> @@ -2,6 +2,7 @@ clean:: >> $(RM) $(CLEANSUFFIXES:%=libavcodec/aac/%) >> >> OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ >> - aac/aacdec_float.o >> + aac/aacdec_float.o aac/aacdec_usac.o \ >> + aac/aacdec_ac.o > > AR libavcodec/libavcodec.a > ar: libavcodec/aac/aacdec_ac.o: No such file or directory > make: *** [ffbuild/library.mak:38: libavcodec/libavcodec.a] Error 1 Thanks, a stray git add, fixed in my repo [-- Attachment #1.1.1.2: OpenPGP public key --] [-- Type: application/pgp-keys, Size: 637 bytes --] [-- Attachment #1.2: OpenPGP digital signature --] [-- Type: application/pgp-signature, Size: 236 bytes --] [-- Attachment #2: Type: text/plain, Size: 251 bytes --] _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 2/8] aacdec: expose channel layout related functions 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 3/8] aacdec: expose decode_tns Lynne via ffmpeg-devel ` (5 subsequent siblings) 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne --- libavcodec/aac/aacdec.c | 73 ++++++++++++++++++++--------------------- libavcodec/aac/aacdec.h | 19 +++++++++-- 2 files changed, 51 insertions(+), 41 deletions(-) diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 35722f9b9b..40554ff9e4 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -111,10 +111,6 @@ Parametric Stereo. */ -static int output_configure(AACDecContext *ac, - uint8_t layout_map[MAX_ELEM_ID*4][3], int tags, - enum OCStatus oc_type, int get_new_frame); - #define overread_err "Input buffer exhausted before END element found\n" static int count_channels(uint8_t (*layout)[3], int tags) @@ -447,8 +443,8 @@ static void pop_output_configuration(AACDecContext *ac) if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) { ac->oc[1] = ac->oc[0]; ac->avctx->ch_layout = ac->oc[1].ch_layout; - output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, - ac->oc[1].status, 0); + ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, + ac->oc[1].status, 0); } } @@ -458,7 +454,7 @@ static void pop_output_configuration(AACDecContext *ac) * * @return Returns error status. 0 - OK, !0 - error */ -static int output_configure(AACDecContext *ac, +int ff_aac_output_configure(AACDecContext *ac, uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags, enum OCStatus oc_type, int get_new_frame) { @@ -547,7 +543,7 @@ static av_cold void flush(AVCodecContext *avctx) * * @return Returns error status. 0 - OK, !0 - error */ -static int set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, +int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, uint8_t (*layout_map)[3], int *tags, int channel_config) @@ -587,7 +583,7 @@ static int set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, return 0; } -static ChannelElement *get_che(AACDecContext *ac, int type, int elem_id) +ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id) { /* For PCE based channel configurations map the channels solely based * on tags. */ @@ -603,11 +599,11 @@ static ChannelElement *get_che(AACDecContext *ac, int type, int elem_id) av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n"); - if (set_default_channel_config(ac, ac->avctx, layout_map, - &layout_map_tags, 2) < 0) + if (ff_aac_set_default_channel_config(ac, ac->avctx, layout_map, + &layout_map_tags, 2) < 0) return NULL; - if (output_configure(ac, layout_map, layout_map_tags, - OC_TRIAL_FRAME, 1) < 0) + if (ff_aac_output_configure(ac, layout_map, layout_map_tags, + OC_TRIAL_FRAME, 1) < 0) return NULL; ac->oc[1].m4ac.chan_config = 2; @@ -627,8 +623,8 @@ static ChannelElement *get_che(AACDecContext *ac, int type, int elem_id) layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT; layout_map[0][1] = 0; layout_map[1][1] = 1; - if (output_configure(ac, layout_map, layout_map_tags, - OC_TRIAL_FRAME, 1) < 0) + if (ff_aac_output_configure(ac, layout_map, layout_map_tags, + OC_TRIAL_FRAME, 1) < 0) return NULL; if (ac->oc[1].m4ac.sbr) @@ -877,8 +873,8 @@ static int decode_ga_specific_config(AACDecContext *ac, AVCodecContext *avctx, if (tags < 0) return tags; } else { - if ((ret = set_default_channel_config(ac, avctx, layout_map, - &tags, channel_config))) + if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, + &tags, channel_config))) return ret; } @@ -887,7 +883,7 @@ static int decode_ga_specific_config(AACDecContext *ac, AVCodecContext *avctx, } else if (m4ac->sbr == 1 && m4ac->ps == -1) m4ac->ps = 1; - if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) + if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) return ret; if (extension_flag) { @@ -967,11 +963,11 @@ static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx, skip_bits_long(gb, 8 * len); } - if ((ret = set_default_channel_config(ac, avctx, layout_map, - &tags, channel_config))) + if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, + &tags, channel_config))) return ret; - if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) + if (ac && (ret = ff_aac_output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0))) return ret; ep_config = get_bits(gb, 2); @@ -1206,11 +1202,12 @@ av_cold int ff_aac_decode_init(AVCodecContext *avctx) ac->oc[1].m4ac.chan_config = i; if (ac->oc[1].m4ac.chan_config) { - int ret = set_default_channel_config(ac, avctx, layout_map, - &layout_map_tags, ac->oc[1].m4ac.chan_config); + int ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, + &layout_map_tags, + ac->oc[1].m4ac.chan_config); if (!ret) - output_configure(ac, layout_map, layout_map_tags, - OC_GLOBAL_HDR, 0); + ff_aac_output_configure(ac, layout_map, layout_map_tags, + OC_GLOBAL_HDR, 0); else if (avctx->err_recognition & AV_EF_EXPLODE) return AVERROR_INVALIDDATA; } @@ -1915,8 +1912,8 @@ static int decode_extension_payload(AACDecContext *ac, GetBitContext *gb, int cn ac->oc[1].m4ac.sbr = 1; ac->oc[1].m4ac.ps = 1; ac->avctx->profile = AV_PROFILE_AAC_HE_V2; - output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, - ac->oc[1].status, 1); + ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, + ac->oc[1].status, 1); } else { ac->oc[1].m4ac.sbr = 1; ac->avctx->profile = AV_PROFILE_AAC_HE; @@ -2066,13 +2063,13 @@ static int parse_adts_frame_header(AACDecContext *ac, GetBitContext *gb) push_output_configuration(ac); if (hdr_info.chan_config) { ac->oc[1].m4ac.chan_config = hdr_info.chan_config; - if ((ret = set_default_channel_config(ac, ac->avctx, - layout_map, - &layout_map_tags, - hdr_info.chan_config)) < 0) + if ((ret = ff_aac_set_default_channel_config(ac, ac->avctx, + layout_map, + &layout_map_tags, + hdr_info.chan_config)) < 0) return ret; - if ((ret = output_configure(ac, layout_map, layout_map_tags, - FFMAX(ac->oc[1].status, + if ((ret = ff_aac_output_configure(ac, layout_map, layout_map_tags, + FFMAX(ac->oc[1].status, OC_TRIAL_FRAME), 0)) < 0) return ret; } else { @@ -2088,8 +2085,8 @@ static int parse_adts_frame_header(AACDecContext *ac, GetBitContext *gb) layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT; layout_map[0][1] = 0; layout_map[1][1] = 1; - if (output_configure(ac, layout_map, layout_map_tags, - OC_TRIAL_FRAME, 0)) + if (ff_aac_output_configure(ac, layout_map, layout_map_tags, + OC_TRIAL_FRAME, 0)) return -7; } } @@ -2142,7 +2139,7 @@ static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame, for (i = 0; i < ff_tags_per_config[chan_config]; i++) { const int elem_type = ff_aac_channel_layout_map[chan_config-1][i][0]; const int elem_id = ff_aac_channel_layout_map[chan_config-1][i][1]; - if (!(che=get_che(ac, elem_type, elem_id))) { + if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); @@ -2241,7 +2238,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, } che_presence[elem_type][elem_id]++; - if (!(che=get_che(ac, elem_type, elem_id))) { + if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); err = AVERROR_INVALIDDATA; @@ -2298,7 +2295,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, "Not evaluating a further program_config_element as this construct is dubious at best.\n"); pop_output_configuration(ac); } else { - err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1); + err = ff_aac_output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1); if (!err) ac->oc[1].m4ac.chan_config = 0; pce_found = 1; diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index 7585a4450b..f54618aaa9 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -39,6 +39,7 @@ #include "libavutil/tx.h" #include "libavcodec/aac.h" +#include "libavcodec/avcodec.h" #include "libavcodec/mpeg4audio.h" typedef struct AACDecContext AACDecContext; @@ -343,10 +344,22 @@ struct AACDecContext { #define fdsp RENAME_FIXED(fdsp) #endif -int ff_aac_decode_init(struct AVCodecContext *avctx); -int ff_aac_decode_init_float(struct AVCodecContext *avctx); -int ff_aac_decode_init_fixed(struct AVCodecContext *avctx); +int ff_aac_decode_init(AVCodecContext *avctx); +int ff_aac_decode_init_float(AVCodecContext *avctx); +int ff_aac_decode_init_fixed(AVCodecContext *avctx); + int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, GetBitContext *gb, int common_window, int scale_flag); +int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, + uint8_t (*layout_map)[3], + int *tags, + int channel_config); + +int ff_aac_output_configure(AACDecContext *ac, + uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags, + enum OCStatus oc_type, int get_new_frame); + +ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id); + #endif /* AVCODEC_AAC_AACDEC_H */ -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 3/8] aacdec: expose decode_tns 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 2/8] aacdec: expose channel layout related functions Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 4/8] aacdec_dsp: implement 768-point transform and windowing Lynne via ffmpeg-devel ` (4 subsequent siblings) 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne USAC has the same syntax, with one minor change we can check for. --- libavcodec/aac/aacdec.c | 6 +++--- libavcodec/aac/aacdec.h | 3 +++ 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 40554ff9e4..a7e5b2a369 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -1542,7 +1542,7 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb, * * @return Returns error status. 0 - OK, !0 - error */ -static int decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, +int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics) { int w, filt, i, coef_len, coef_res, coef_compress; @@ -1690,7 +1690,7 @@ int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, } tns->present = get_bits1(gb); if (tns->present && !er_syntax) { - ret = decode_tns(ac, tns, gb, ics); + ret = ff_aac_decode_tns(ac, tns, gb, ics); if (ret < 0) goto fail; } @@ -1704,7 +1704,7 @@ int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, // I see no textual basis in the spec for this occurring after SSR gain // control, but this is what both reference and real implmentations do if (tns->present && er_syntax) { - ret = decode_tns(ac, tns, gb, ics); + ret = ff_aac_decode_tns(ac, tns, gb, ics); if (ret < 0) goto fail; } diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index f54618aaa9..303e285c81 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -351,6 +351,9 @@ int ff_aac_decode_init_fixed(AVCodecContext *avctx); int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce, GetBitContext *gb, int common_window, int scale_flag); +int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, + GetBitContext *gb, const IndividualChannelStream *ics); + int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx, uint8_t (*layout_map)[3], int *tags, -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 4/8] aacdec_dsp: implement 768-point transform and windowing 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel ` (2 preceding siblings ...) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 3/8] aacdec: expose decode_tns Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 5/8] aactab: add deemphasis tables for USAC Lynne via ffmpeg-devel ` (3 subsequent siblings) 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne Required for USAC --- libavcodec/aac/aacdec.c | 4 ++ libavcodec/aac/aacdec.h | 5 +++ libavcodec/aac/aacdec_dsp_template.c | 67 ++++++++++++++++++++++++++++ libavcodec/aac/aacdec_fixed.c | 2 + libavcodec/aac/aacdec_float.c | 4 ++ libavcodec/sinewin_fixed_tablegen.c | 2 + libavcodec/sinewin_fixed_tablegen.h | 4 ++ 7 files changed, 88 insertions(+) diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index a7e5b2a369..6f37ac5361 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -1113,10 +1113,12 @@ static av_cold int decode_close(AVCodecContext *avctx) } } + av_tx_uninit(&ac->mdct96); av_tx_uninit(&ac->mdct120); av_tx_uninit(&ac->mdct128); av_tx_uninit(&ac->mdct480); av_tx_uninit(&ac->mdct512); + av_tx_uninit(&ac->mdct768); av_tx_uninit(&ac->mdct960); av_tx_uninit(&ac->mdct1024); av_tx_uninit(&ac->mdct_ltp); @@ -1145,10 +1147,12 @@ static av_cold int init_dsp(AVCodecContext *avctx) if (ret < 0) \ return ret + MDCT_INIT(ac->mdct96, ac->mdct96_fn, 96, 1.0/96); MDCT_INIT(ac->mdct120, ac->mdct120_fn, 120, 1.0/120); MDCT_INIT(ac->mdct128, ac->mdct128_fn, 128, 1.0/128); MDCT_INIT(ac->mdct480, ac->mdct480_fn, 480, 1.0/480); MDCT_INIT(ac->mdct512, ac->mdct512_fn, 512, 1.0/512); + MDCT_INIT(ac->mdct768, ac->mdct768_fn, 768, 1.0/768); MDCT_INIT(ac->mdct960, ac->mdct960_fn, 960, 1.0/960); MDCT_INIT(ac->mdct1024, ac->mdct1024_fn, 1024, 1.0/1024); #undef MDCT_INIT diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index 303e285c81..20545a24d4 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -245,6 +245,7 @@ typedef struct AACDecDSP { ChannelElement *cce, int index); void (*imdct_and_windowing)(AACDecContext *ac, SingleChannelElement *sce); + void (*imdct_and_windowing_768)(AACDecContext *ac, SingleChannelElement *sce); void (*imdct_and_windowing_960)(AACDecContext *ac, SingleChannelElement *sce); void (*imdct_and_windowing_ld)(AACDecContext *ac, SingleChannelElement *sce); void (*imdct_and_windowing_eld)(AACDecContext *ac, SingleChannelElement *sce); @@ -290,18 +291,22 @@ struct AACDecContext { * @name Computed / set up during initialization * @{ */ + AVTXContext *mdct96; AVTXContext *mdct120; AVTXContext *mdct128; AVTXContext *mdct480; AVTXContext *mdct512; + AVTXContext *mdct768; AVTXContext *mdct960; AVTXContext *mdct1024; AVTXContext *mdct_ltp; + av_tx_fn mdct96_fn; av_tx_fn mdct120_fn; av_tx_fn mdct128_fn; av_tx_fn mdct480_fn; av_tx_fn mdct512_fn; + av_tx_fn mdct768_fn; av_tx_fn mdct960_fn; av_tx_fn mdct1024_fn; av_tx_fn mdct_ltp_fn; diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index e69970472c..59a69d88f3 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -383,6 +383,71 @@ static void AAC_RENAME(imdct_and_windowing)(AACDecContext *ac, SingleChannelElem } } +/** + * Conduct IMDCT and windowing for 768-point frames. + */ +static void AAC_RENAME(imdct_and_windowing_768)(AACDecContext *ac, SingleChannelElement *sce) +{ + IndividualChannelStream *ics = &sce->ics; + INTFLOAT *in = sce->AAC_RENAME(coeffs); + INTFLOAT *out = sce->AAC_RENAME(output); + INTFLOAT *saved = sce->AAC_RENAME(saved); + const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(aac_kbd_short_96) : AAC_RENAME(sine_96); + const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(aac_kbd_long_768) : AAC_RENAME(sine_768); + const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(aac_kbd_short_96) : AAC_RENAME(sine_96); + INTFLOAT *buf = ac->AAC_RENAME(buf_mdct); + INTFLOAT *temp = ac->AAC_RENAME(temp); + int i; + + // imdct + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + for (i = 0; i < 8; i++) + ac->mdct96_fn(ac->mdct96, buf + i * 96, in + i * 96, sizeof(INTFLOAT)); + } else { + ac->mdct768_fn(ac->mdct768, buf, in, sizeof(INTFLOAT)); + } + + /* window overlapping + * NOTE: To simplify the overlapping code, all 'meaningless' short to long + * and long to short transitions are considered to be short to short + * transitions. This leaves just two cases (long to long and short to short) + * with a little special sauce for EIGHT_SHORT_SEQUENCE. + */ + + if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && + (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { + ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 384); + } else { + memcpy( out, saved, 336 * sizeof(*out)); + + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ac->fdsp->vector_fmul_window(out + 336 + 0*96, saved + 336, buf + 0*96, swindow_prev, 48); + ac->fdsp->vector_fmul_window(out + 336 + 1*96, buf + 0*96 + 48, buf + 1*96, swindow, 48); + ac->fdsp->vector_fmul_window(out + 336 + 2*96, buf + 1*96 + 48, buf + 2*96, swindow, 48); + ac->fdsp->vector_fmul_window(out + 336 + 3*96, buf + 2*96 + 48, buf + 3*96, swindow, 48); + ac->fdsp->vector_fmul_window(temp, buf + 3*96 + 48, buf + 4*96, swindow, 48); + memcpy( out + 336 + 4*96, temp, 48 * sizeof(*out)); + } else { + ac->fdsp->vector_fmul_window(out + 336, saved + 336, buf, swindow_prev, 48); + memcpy( out + 432, buf + 48, 336 * sizeof(*out)); + } + } + + // buffer update + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + memcpy( saved, temp + 48, 48 * sizeof(*saved)); + ac->fdsp->vector_fmul_window(saved + 48, buf + 4*96 + 48, buf + 5*96, swindow, 48); + ac->fdsp->vector_fmul_window(saved + 144, buf + 5*96 + 48, buf + 6*96, swindow, 48); + ac->fdsp->vector_fmul_window(saved + 240, buf + 6*96 + 48, buf + 7*96, swindow, 48); + memcpy( saved + 336, buf + 7*96 + 48, 48 * sizeof(*saved)); + } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { + memcpy( saved, buf + 384, 336 * sizeof(*saved)); + memcpy( saved + 336, buf + 7*96 + 48, 48 * sizeof(*saved)); + } else { // LONG_STOP or ONLY_LONG + memcpy( saved, buf + 384, 384 * sizeof(*saved)); + } +} + /** * Conduct IMDCT and windowing. */ @@ -447,6 +512,7 @@ static void AAC_RENAME(imdct_and_windowing_960)(AACDecContext *ac, SingleChannel memcpy( saved, buf + 480, 480 * sizeof(*saved)); } } + static void AAC_RENAME(imdct_and_windowing_ld)(AACDecContext *ac, SingleChannelElement *sce) { IndividualChannelStream *ics = &sce->ics; @@ -609,6 +675,7 @@ static av_cold void AAC_RENAME(aac_dsp_init)(AACDecDSP *aac_dsp) SET(apply_prediction); SET(imdct_and_windowing); + SET(imdct_and_windowing_768); SET(imdct_and_windowing_960); SET(imdct_and_windowing_ld); SET(imdct_and_windowing_eld); diff --git a/libavcodec/aac/aacdec_fixed.c b/libavcodec/aac/aacdec_fixed.c index de90880884..89f1ea0384 100644 --- a/libavcodec/aac/aacdec_fixed.c +++ b/libavcodec/aac/aacdec_fixed.c @@ -47,6 +47,8 @@ DECLARE_ALIGNED(32, static int, aac_kbd_long_1024_fixed)[1024]; DECLARE_ALIGNED(32, static int, aac_kbd_short_128_fixed)[128]; DECLARE_ALIGNED(32, static int, aac_kbd_long_960_fixed)[960]; DECLARE_ALIGNED(32, static int, aac_kbd_short_120_fixed)[120]; +DECLARE_ALIGNED(32, static int, aac_kbd_long_768_fixed)[768]; +DECLARE_ALIGNED(32, static int, aac_kbd_short_96_fixed)[96]; static void init_tables_fixed_fn(void) { diff --git a/libavcodec/aac/aacdec_float.c b/libavcodec/aac/aacdec_float.c index 03ec264c50..14169e95d8 100644 --- a/libavcodec/aac/aacdec_float.c +++ b/libavcodec/aac/aacdec_float.c @@ -44,10 +44,14 @@ #include "libavutil/mathematics.h" #include "libavcodec/aacsbr.h" +DECLARE_ALIGNED(32, static float, sine_96)[96]; DECLARE_ALIGNED(32, static float, sine_120)[120]; +DECLARE_ALIGNED(32, static float, sine_768)[768]; DECLARE_ALIGNED(32, static float, sine_960)[960]; DECLARE_ALIGNED(32, static float, aac_kbd_long_960)[960]; DECLARE_ALIGNED(32, static float, aac_kbd_short_120)[120]; +DECLARE_ALIGNED(32, static float, aac_kbd_long_768)[768]; +DECLARE_ALIGNED(32, static float, aac_kbd_short_96)[96]; static void init_tables_float_fn(void) { diff --git a/libavcodec/sinewin_fixed_tablegen.c b/libavcodec/sinewin_fixed_tablegen.c index 15f0cc2072..86e9dfb1e7 100644 --- a/libavcodec/sinewin_fixed_tablegen.c +++ b/libavcodec/sinewin_fixed_tablegen.c @@ -35,10 +35,12 @@ int main(void) printf("SINETABLE("#size") = {\n"); \ write_int32_t_array(sine_ ## size ## _fixed, size); \ printf("};\n") + PRINT_TABLE(96); PRINT_TABLE(120); PRINT_TABLE(128); PRINT_TABLE(480); PRINT_TABLE(512); + PRINT_TABLE(768); PRINT_TABLE(960); PRINT_TABLE(1024); return 0; diff --git a/libavcodec/sinewin_fixed_tablegen.h b/libavcodec/sinewin_fixed_tablegen.h index 056735704c..660c0056b5 100644 --- a/libavcodec/sinewin_fixed_tablegen.h +++ b/libavcodec/sinewin_fixed_tablegen.h @@ -44,10 +44,12 @@ #include "libavutil/attributes.h" #define SINETABLE_CONST +SINETABLE( 96); SINETABLE( 120); SINETABLE( 128); SINETABLE( 480); SINETABLE( 512); +SINETABLE( 768); SINETABLE( 960); SINETABLE(1024); @@ -62,10 +64,12 @@ static av_cold void sine_window_init_fixed(int *window, int n) static av_cold void init_sine_windows_fixed(void) { + sine_window_init_fixed(sine_96_fixed, 96); sine_window_init_fixed(sine_120_fixed, 120); sine_window_init_fixed(sine_128_fixed, 128); sine_window_init_fixed(sine_480_fixed, 480); sine_window_init_fixed(sine_512_fixed, 512); + sine_window_init_fixed(sine_768_fixed, 768); sine_window_init_fixed(sine_960_fixed, 960); sine_window_init_fixed(sine_1024_fixed, 1024); } -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 5/8] aactab: add deemphasis tables for USAC 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel ` (3 preceding siblings ...) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 4/8] aacdec_dsp: implement 768-point transform and windowing Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 6/8] aactab: add tables for the new USAC arithmetic coder Lynne via ffmpeg-devel ` (2 subsequent siblings) 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne --- libavcodec/aactab.c | 25 +++++++++++++++++++++++++ libavcodec/aactab.h | 2 ++ 2 files changed, 27 insertions(+) diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index 3718b81a07..8ce5e43974 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -3377,3 +3377,28 @@ const DECLARE_ALIGNED(32, int, ff_aac_eld_window_480_fixed)[1800] = { 0xffecff1c, 0xffed391e, 0xffed740c, 0xffedafb1, 0xffedebe1, 0xffee287d, 0xffee654e, 0xffeea23f, }; + +/* As specified by ISO/IEC 23003 */ +#define USAC_EMPH_COEFF 0.68 + +DECLARE_ALIGNED(16, const float, ff_aac_deemph_weights)[16] = { + USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF*USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF*USAC_EMPH_COEFF*USAC_EMPH_COEFF, + + 0, + USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF*USAC_EMPH_COEFF, + + 0, + 0, + USAC_EMPH_COEFF, + USAC_EMPH_COEFF*USAC_EMPH_COEFF, + + 0, + 0, + 0, + USAC_EMPH_COEFF, +}; diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index e1a2d8b9a1..91262380d4 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -64,6 +64,8 @@ DECLARE_ALIGNED(32, extern const float, ff_aac_eld_window_480)[1800]; DECLARE_ALIGNED(32, extern const int, ff_aac_eld_window_480_fixed)[1800]; // @} +extern const float ff_aac_deemph_weights[16]; + /* Initializes data shared between float decoder and encoder. */ void ff_aac_float_common_init(void); -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 6/8] aactab: add tables for the new USAC arithmetic coder 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel ` (4 preceding siblings ...) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 5/8] aactab: add deemphasis tables for USAC Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 7/8] aactab: add new scalefactor offset tables for 96/768pt windows Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) Lynne via ffmpeg-devel 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne --- libavcodec/aactab.c | 376 ++++++++++++++++++++++++++++++++++++++++++++ libavcodec/aactab.h | 6 + 2 files changed, 382 insertions(+) diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index 8ce5e43974..dfb2dfd98d 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -1193,6 +1193,382 @@ const uint16_t *const ff_aac_codebook_vector_idx[] = { codebook_vector10_idx, }; +const uint16_t ff_aac_ac_msb_cdfs[64][17] = { + { 708, 706, 579, 569, 568, 567, 479, 469, + 297, 138, 97, 91, 72, 52, 38, 34, 0, }, + { 7619, 6917, 6519, 6412, 5514, 5003, 4683, 4563, + 3907, 3297, 3125, 3060, 2904, 2718, 2631, 2590, 0, }, + { 7263, 4888, 4810, 4803, 1889, 415, 335, 327, + 195, 72, 52, 49, 36, 20, 15, 14, 0, }, + { 3626, 2197, 2188, 2187, 582, 57, 47, 46, + 30, 12, 9, 8, 6, 4, 3, 2, 0, }, + { 7806, 5541, 5451, 5441, 2720, 834, 691, 674, + 487, 243, 179, 167, 139, 98, 77, 70, 0, }, + { 6684, 4101, 4058, 4055, 1748, 426, 368, 364, + 322, 257, 235, 232, 228, 222, 217, 215, 0, }, + { 9162, 5964, 5831, 5819, 3269, 866, 658, 638, + 535, 348, 258, 244, 234, 214, 195, 186, 0, }, + { 10638, 8491, 8365, 8351, 4418, 2067, 1859, 1834, + 1190, 601, 495, 478, 356, 217, 174, 164, 0, }, + { 13389, 10514, 10032, 9961, 7166, 3488, 2655, 2524, + 2015, 1140, 760, 672, 585, 426, 325, 283, 0, }, + { 14861, 12788, 12115, 11952, 9987, 6657, 5323, 4984, + 4324, 3001, 2205, 1943, 1764, 1394, 1115, 978, 0, }, + { 12876, 10004, 9661, 9610, 7107, 3435, 2711, 2595, + 2257, 1508, 1059, 952, 893, 753, 609, 538, 0, }, + { 15125, 13591, 13049, 12874, 11192, 8543, 7406, 7023, + 6291, 4922, 4104, 3769, 3465, 2890, 2486, 2275, 0, }, + { 14574, 13106, 12731, 12638, 10453, 7947, 7233, 7037, + 6031, 4618, 4081, 3906, 3465, 2802, 2476, 2349, 0, }, + { 15070, 13179, 12517, 12351, 10742, 7657, 6200, 5825, + 5264, 3998, 3014, 2662, 2510, 2153, 1799, 1564, 0, }, + { 15542, 14466, 14007, 13844, 12489, 10409, 9481, 9132, + 8305, 6940, 6193, 5867, 5458, 4743, 4291, 4047, 0, }, + { 15165, 14384, 14084, 13934, 12911, 11485, 10844, 10513, + 10002, 8993, 8380, 8051, 7711, 7036, 6514, 6233, 0, }, + { 15642, 14279, 13625, 13393, 12348, 9971, 8405, 7858, + 7335, 6119, 4918, 4376, 4185, 3719, 3231, 2860, 0, }, + { 13408, 13407, 11471, 11218, 11217, 11216, 9473, 9216, + 6480, 3689, 2857, 2690, 2256, 1732, 1405, 1302, 0, }, + { 16098, 15584, 15191, 14931, 14514, 13578, 12703, 12103, + 11830, 11172, 10475, 9867, 9695, 9281, 8825, 8389, 0, }, + { 15844, 14873, 14277, 13996, 13230, 11535, 10205, 9543, + 9107, 8086, 7085, 6419, 6214, 5713, 5195, 4731, 0, }, + { 16131, 15720, 15443, 15276, 14848, 13971, 13314, 12910, + 12591, 11874, 11225, 10788, 10573, 10077, 9585, 9209, 0, }, + { 16331, 16330, 12283, 11435, 11434, 11433, 8725, 8049, + 6065, 4138, 3187, 2842, 2529, 2171, 1907, 1745, 0, }, + { 16011, 15292, 14782, 14528, 14008, 12767, 11556, 10921, + 10591, 9759, 8813, 8043, 7855, 7383, 6863, 6282, 0, }, + { 16380, 16379, 15159, 14610, 14609, 14608, 12859, 12111, + 11046, 9536, 8348, 7713, 7216, 6533, 5964, 5546, 0, }, + { 16367, 16333, 16294, 16253, 16222, 16143, 16048, 15947, + 15915, 15832, 15731, 15619, 15589, 15512, 15416, 15310, 0, }, + { 15967, 15319, 14937, 14753, 14010, 12638, 11787, 11360, + 10805, 9706, 8934, 8515, 8166, 7456, 6911, 6575, 0, }, + { 4906, 3005, 2985, 2984, 875, 102, 83, 81, + 47, 17, 12, 11, 8, 5, 4, 3, 0, }, + { 7217, 4346, 4269, 4264, 1924, 428, 340, 332, + 280, 203, 179, 175, 171, 164, 159, 157, 0, }, + { 16010, 15415, 15032, 14805, 14228, 13043, 12168, 11634, + 11265, 10419, 9645, 9110, 8892, 8378, 7850, 7437, 0, }, + { 8573, 5218, 5046, 5032, 2787, 771, 555, 533, + 443, 286, 218, 205, 197, 181, 168, 162, 0, }, + { 11474, 8095, 7822, 7796, 4632, 1443, 1046, 1004, + 748, 351, 218, 194, 167, 121, 93, 83, 0, }, + { 16152, 15764, 15463, 15264, 14925, 14189, 13536, 13070, + 12846, 12314, 11763, 11277, 11131, 10777, 10383, 10011, 0, }, + { 14187, 11654, 11043, 10919, 8498, 4885, 3778, 3552, + 2947, 1835, 1283, 1134, 998, 749, 585, 514, 0, }, + { 14162, 11527, 10759, 10557, 8601, 5417, 4105, 3753, + 3286, 2353, 1708, 1473, 1370, 1148, 959, 840, 0, }, + { 16205, 15902, 15669, 15498, 15213, 14601, 14068, 13674, + 13463, 12970, 12471, 12061, 11916, 11564, 11183, 10841, 0, }, + { 15043, 12972, 12092, 11792, 10265, 7446, 5934, 5379, + 4883, 3825, 3036, 2647, 2507, 2185, 1901, 1699, 0, }, + { 15320, 13694, 12782, 12352, 11191, 8936, 7433, 6671, + 6255, 5366, 4622, 4158, 4020, 3712, 3420, 3198, 0, }, + { 16255, 16020, 15768, 15600, 15416, 14963, 14440, 14006, + 13875, 13534, 13137, 12697, 12602, 12364, 12084, 11781, 0, }, + { 15627, 14503, 13906, 13622, 12557, 10527, 9269, 8661, + 8117, 6933, 5994, 5474, 5222, 4664, 4166, 3841, 0, }, + { 16366, 16365, 14547, 14160, 14159, 14158, 11969, 11473, + 8735, 6147, 4911, 4530, 3865, 3180, 2710, 2473, 0, }, + { 16257, 16038, 15871, 15754, 15536, 15071, 14673, 14390, + 14230, 13842, 13452, 13136, 13021, 12745, 12434, 12154, 0, }, + { 15855, 14971, 14338, 13939, 13239, 11782, 10585, 9805, + 9444, 8623, 7846, 7254, 7079, 6673, 6262, 5923, 0, }, + { 9492, 6318, 6197, 6189, 3004, 652, 489, 477, + 333, 143, 96, 90, 78, 60, 50, 47, 0, }, + { 16313, 16191, 16063, 15968, 15851, 15590, 15303, 15082, + 14968, 14704, 14427, 14177, 14095, 13899, 13674, 13457, 0, }, + { 8485, 5473, 5389, 5383, 2411, 494, 386, 377, + 278, 150, 117, 112, 103, 89, 81, 78, 0, }, + { 10497, 7154, 6959, 6943, 3788, 1004, 734, 709, + 517, 238, 152, 138, 120, 90, 72, 66, 0, }, + { 16317, 16226, 16127, 16040, 15955, 15762, 15547, 15345, + 15277, 15111, 14922, 14723, 14671, 14546, 14396, 14239, 0, }, + { 16382, 16381, 15858, 15540, 15539, 15538, 14704, 14168, + 13768, 13092, 12452, 11925, 11683, 11268, 10841, 10460, 0, }, + { 5974, 3798, 3758, 3755, 1275, 205, 166, 162, + 95, 35, 26, 24, 18, 11, 8, 7, 0, }, + { 3532, 2258, 2246, 2244, 731, 135, 118, 115, + 87, 45, 36, 34, 29, 21, 17, 16, 0, }, + { 7466, 4882, 4821, 4811, 2476, 886, 788, 771, + 688, 531, 469, 457, 437, 400, 369, 361, 0, }, + { 9580, 5772, 5291, 5216, 3444, 1496, 1025, 928, + 806, 578, 433, 384, 366, 331, 296, 273, 0, }, + { 10692, 7730, 7543, 7521, 4679, 1746, 1391, 1346, + 1128, 692, 495, 458, 424, 353, 291, 268, 0, }, + { 11040, 7132, 6549, 6452, 4377, 1875, 1253, 1130, + 958, 631, 431, 370, 346, 296, 253, 227, 0, }, + { 12687, 9332, 8701, 8585, 6266, 3093, 2182, 2004, + 1683, 1072, 712, 608, 559, 458, 373, 323, 0, }, + { 13429, 9853, 8860, 8584, 6806, 4039, 2862, 2478, + 2239, 1764, 1409, 1224, 1178, 1077, 979, 903, 0, }, + { 14685, 12163, 11061, 10668, 9101, 6345, 4871, 4263, + 3908, 3200, 2668, 2368, 2285, 2106, 1942, 1819, 0, }, + { 13295, 11302, 10999, 10945, 7947, 5036, 4490, 4385, + 3391, 2185, 1836, 1757, 1424, 998, 833, 785, 0, }, + { 4992, 2993, 2972, 2970, 1269, 575, 552, 549, + 530, 505, 497, 495, 493, 489, 486, 485, 0, }, + { 15419, 13862, 13104, 12819, 11429, 8753, 7220, 6651, + 6020, 4667, 3663, 3220, 2995, 2511, 2107, 1871, 0, }, + { 12468, 9263, 8912, 8873, 5758, 2193, 1625, 1556, + 1187, 589, 371, 330, 283, 200, 149, 131, 0, }, + { 15870, 15076, 14615, 14369, 13586, 12034, 10990, 10423, + 9953, 8908, 8031, 7488, 7233, 6648, 6101, 5712, 0, }, + { 1693, 978, 976, 975, 194, 18, 16, 15, + 11, 7, 6, 5, 4, 3, 2, 1, 0, }, + { 7992, 5218, 5147, 5143, 2152, 366, 282, 276, + 173, 59, 38, 35, 27, 16, 11, 10, 0, } +}; + +const uint16_t ff_aac_ac_lsb_cdfs[3][4] = { + { 12571, 10569, 3696, 0 }, + { 12661, 5700, 3751, 0 }, + { 10827, 6884, 2929, 0 } +}; + +const uint8_t ff_aac_ac_lookup_m[742] = { + 0x01, 0x34, 0x0D, 0x13, 0x12, 0x25, 0x00, 0x3A, 0x05, 0x00, 0x21, 0x13, 0x1F, 0x1A, 0x1D, 0x36, + 0x24, 0x2B, 0x1B, 0x33, 0x37, 0x29, 0x1D, 0x33, 0x37, 0x33, 0x37, 0x33, 0x37, 0x33, 0x2C, 0x00, + 0x21, 0x13, 0x25, 0x2A, 0x00, 0x21, 0x24, 0x12, 0x2C, 0x1E, 0x37, 0x24, 0x1F, 0x35, 0x37, 0x24, + 0x35, 0x37, 0x35, 0x37, 0x38, 0x2D, 0x21, 0x29, 0x1E, 0x21, 0x13, 0x2D, 0x36, 0x38, 0x29, 0x36, + 0x37, 0x24, 0x36, 0x38, 0x37, 0x38, 0x00, 0x20, 0x23, 0x20, 0x23, 0x36, 0x38, 0x24, 0x3B, 0x24, + 0x26, 0x29, 0x1F, 0x30, 0x2D, 0x0D, 0x12, 0x3F, 0x2D, 0x21, 0x1C, 0x2A, 0x00, 0x21, 0x12, 0x1E, + 0x36, 0x38, 0x36, 0x37, 0x3F, 0x1E, 0x0D, 0x1F, 0x2A, 0x1E, 0x21, 0x24, 0x12, 0x2A, 0x3C, 0x21, + 0x24, 0x1F, 0x3C, 0x21, 0x29, 0x36, 0x38, 0x36, 0x37, 0x38, 0x21, 0x1E, 0x00, 0x3B, 0x25, 0x1E, + 0x20, 0x10, 0x1F, 0x3C, 0x20, 0x23, 0x29, 0x08, 0x23, 0x12, 0x08, 0x23, 0x21, 0x38, 0x00, 0x20, + 0x13, 0x20, 0x3B, 0x1C, 0x20, 0x3B, 0x29, 0x20, 0x23, 0x24, 0x21, 0x24, 0x21, 0x24, 0x3B, 0x13, + 0x23, 0x26, 0x23, 0x13, 0x21, 0x24, 0x26, 0x29, 0x12, 0x22, 0x2B, 0x02, 0x1E, 0x0D, 0x1F, 0x2D, + 0x00, 0x0D, 0x12, 0x00, 0x3C, 0x21, 0x29, 0x3C, 0x21, 0x2A, 0x3C, 0x3B, 0x22, 0x1E, 0x20, 0x10, + 0x1F, 0x3C, 0x0D, 0x29, 0x3C, 0x21, 0x24, 0x08, 0x23, 0x20, 0x38, 0x39, 0x3C, 0x20, 0x13, 0x3C, + 0x00, 0x0D, 0x13, 0x1F, 0x3C, 0x09, 0x26, 0x1F, 0x08, 0x09, 0x26, 0x12, 0x08, 0x23, 0x29, 0x20, + 0x23, 0x21, 0x24, 0x20, 0x13, 0x20, 0x3B, 0x16, 0x20, 0x3B, 0x29, 0x20, 0x3B, 0x29, 0x20, 0x3B, + 0x13, 0x21, 0x24, 0x29, 0x0B, 0x13, 0x09, 0x3B, 0x13, 0x09, 0x3B, 0x13, 0x21, 0x3B, 0x13, 0x0D, + 0x26, 0x29, 0x26, 0x29, 0x3D, 0x12, 0x22, 0x28, 0x2E, 0x04, 0x08, 0x13, 0x3C, 0x3B, 0x3C, 0x20, + 0x10, 0x3C, 0x21, 0x07, 0x08, 0x10, 0x00, 0x08, 0x0D, 0x29, 0x08, 0x0D, 0x29, 0x08, 0x09, 0x13, + 0x20, 0x23, 0x39, 0x08, 0x09, 0x13, 0x08, 0x09, 0x16, 0x08, 0x09, 0x10, 0x12, 0x20, 0x3B, 0x3D, + 0x09, 0x26, 0x20, 0x3B, 0x24, 0x39, 0x09, 0x26, 0x20, 0x0D, 0x13, 0x00, 0x09, 0x13, 0x20, 0x0D, + 0x26, 0x12, 0x20, 0x3B, 0x13, 0x21, 0x26, 0x0B, 0x12, 0x09, 0x3B, 0x16, 0x09, 0x3B, 0x3D, 0x09, + 0x26, 0x0D, 0x13, 0x26, 0x3D, 0x1C, 0x12, 0x1F, 0x28, 0x2E, 0x07, 0x0B, 0x08, 0x09, 0x00, 0x39, + 0x0B, 0x08, 0x26, 0x08, 0x09, 0x13, 0x20, 0x0B, 0x39, 0x10, 0x39, 0x0D, 0x13, 0x20, 0x10, 0x12, + 0x09, 0x13, 0x20, 0x3B, 0x13, 0x09, 0x26, 0x0B, 0x09, 0x3B, 0x1C, 0x09, 0x3B, 0x13, 0x20, 0x3B, + 0x13, 0x09, 0x26, 0x0B, 0x16, 0x0D, 0x13, 0x09, 0x13, 0x09, 0x13, 0x26, 0x3D, 0x1C, 0x1F, 0x28, + 0x2E, 0x07, 0x10, 0x39, 0x0B, 0x39, 0x39, 0x13, 0x39, 0x0B, 0x39, 0x0B, 0x39, 0x26, 0x39, 0x10, + 0x20, 0x3B, 0x16, 0x20, 0x10, 0x09, 0x26, 0x0B, 0x13, 0x09, 0x13, 0x26, 0x1C, 0x0B, 0x3D, 0x1C, + 0x1F, 0x28, 0x2B, 0x07, 0x0C, 0x39, 0x0B, 0x39, 0x0B, 0x0C, 0x0B, 0x26, 0x0B, 0x26, 0x3D, 0x0D, + 0x1C, 0x14, 0x28, 0x2B, 0x39, 0x0B, 0x0C, 0x0E, 0x3D, 0x1C, 0x0D, 0x12, 0x22, 0x2B, 0x07, 0x0C, + 0x0E, 0x3D, 0x1C, 0x10, 0x1F, 0x2B, 0x0C, 0x0E, 0x19, 0x14, 0x10, 0x1F, 0x28, 0x0C, 0x0E, 0x19, + 0x14, 0x26, 0x22, 0x2B, 0x0C, 0x0E, 0x19, 0x14, 0x26, 0x28, 0x0E, 0x19, 0x14, 0x26, 0x28, 0x0E, + 0x19, 0x14, 0x28, 0x0E, 0x19, 0x14, 0x22, 0x28, 0x2B, 0x0E, 0x14, 0x2B, 0x31, 0x00, 0x3A, 0x3A, + 0x05, 0x05, 0x1B, 0x1D, 0x33, 0x06, 0x35, 0x35, 0x20, 0x21, 0x37, 0x21, 0x24, 0x05, 0x1B, 0x2C, + 0x2C, 0x2C, 0x06, 0x34, 0x1E, 0x34, 0x00, 0x08, 0x36, 0x09, 0x21, 0x26, 0x1C, 0x2C, 0x00, 0x02, + 0x02, 0x02, 0x3F, 0x04, 0x04, 0x04, 0x34, 0x39, 0x20, 0x0A, 0x0C, 0x39, 0x0B, 0x0F, 0x07, 0x07, + 0x07, 0x07, 0x34, 0x39, 0x39, 0x0A, 0x0C, 0x39, 0x0C, 0x0F, 0x07, 0x07, 0x07, 0x00, 0x39, 0x39, + 0x0C, 0x0F, 0x07, 0x07, 0x39, 0x0C, 0x0F, 0x07, 0x39, 0x0C, 0x0F, 0x39, 0x39, 0x0C, 0x0F, 0x39, + 0x0C, 0x39, 0x0C, 0x0F, 0x00, 0x11, 0x27, 0x17, 0x2F, 0x27, 0x00, 0x27, 0x17, 0x00, 0x11, 0x17, + 0x00, 0x11, 0x17, 0x11, 0x00, 0x27, 0x15, 0x11, 0x17, 0x01, 0x15, 0x11, 0x15, 0x11, 0x15, 0x15, + 0x17, 0x00, 0x27, 0x01, 0x27, 0x27, 0x15, 0x00, 0x27, 0x11, 0x27, 0x15, 0x15, 0x15, 0x27, 0x15, + 0x15, 0x15, 0x15, 0x17, 0x2F, 0x11, 0x17, 0x27, 0x27, 0x27, 0x11, 0x27, 0x15, 0x27, 0x27, 0x15, + 0x15, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, + 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, + 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x27, 0x17, 0x2F, 0x17, 0x2F, 0x2B, 0x00, 0x27, 0x00, 0x00, 0x11, + 0x15, 0x00, 0x11, 0x11, 0x27, 0x27, 0x15, 0x17, 0x15, 0x17, 0x15, 0x17, 0x27, 0x17, 0x27, 0x17, + 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, 0x27, 0x17, + 0x27, 0x15, 0x27, 0x27, 0x15, 0x27 +}; + +const uint32_t ff_aac_ac_hash_m[742] = { + 0x00000104, 0x0000030A, 0x00000510, 0x00000716, + 0x00000A1F, 0x00000F2E, 0x00011100, 0x00111103, + 0x00111306, 0x00111436, 0x00111623, 0x00111929, + 0x00111F2E, 0x0011221B, 0x00112435, 0x00112621, + 0x00112D12, 0x00113130, 0x0011331D, 0x00113535, + 0x00113938, 0x0011411B, 0x00114433, 0x00114635, + 0x00114F29, 0x00116635, 0x00116F24, 0x00117433, + 0x0011FF0F, 0x00121102, 0x0012132D, 0x00121436, + 0x00121623, 0x00121912, 0x0012213F, 0x0012232D, + 0x00122436, 0x00122638, 0x00122A29, 0x00122F2B, + 0x0012322D, 0x00123436, 0x00123738, 0x00123B29, + 0x0012411D, 0x00124536, 0x00124938, 0x00124F12, + 0x00125535, 0x00125F29, 0x00126535, 0x0012B837, + 0x0013112A, 0x0013131E, 0x0013163B, 0x0013212D, + 0x0013233C, 0x00132623, 0x00132F2E, 0x0013321E, + 0x00133521, 0x00133824, 0x0013411E, 0x00134336, + 0x00134838, 0x00135135, 0x00135537, 0x00135F12, + 0x00137637, 0x0013FF29, 0x00140024, 0x00142321, + 0x00143136, 0x00143321, 0x00143F25, 0x00144321, + 0x00148638, 0x0014FF29, 0x00154323, 0x0015FF12, + 0x0016F20C, 0x0018A529, 0x00210031, 0x0021122C, + 0x00211408, 0x00211713, 0x00211F2E, 0x0021222A, + 0x00212408, 0x00212710, 0x00212F2E, 0x0021331E, + 0x00213436, 0x00213824, 0x0021412D, 0x0021431E, + 0x00214536, 0x00214F1F, 0x00216637, 0x00220004, + 0x0022122A, 0x00221420, 0x00221829, 0x00221F2E, + 0x0022222D, 0x00222408, 0x00222623, 0x00222929, + 0x00222F2B, 0x0022321E, 0x00223408, 0x00223724, + 0x00223A29, 0x0022411E, 0x00224436, 0x00224823, + 0x00225134, 0x00225621, 0x00225F12, 0x00226336, + 0x00227637, 0x0022FF29, 0x0023112D, 0x0023133C, + 0x00231420, 0x00231916, 0x0023212D, 0x0023233C, + 0x00232509, 0x00232929, 0x0023312D, 0x00233308, + 0x00233509, 0x00233724, 0x0023413C, 0x00234421, + 0x00234A13, 0x0023513C, 0x00235421, 0x00235F1F, + 0x00236421, 0x0023FF29, 0x00240024, 0x0024153B, + 0x00242108, 0x00242409, 0x00242726, 0x00243108, + 0x00243409, 0x00243610, 0x00244136, 0x00244321, + 0x00244523, 0x00244F1F, 0x00245423, 0x0024610A, + 0x00246423, 0x0024FF29, 0x00252510, 0x00253121, + 0x0025343B, 0x00254121, 0x00254510, 0x00254F25, + 0x00255221, 0x0025FF12, 0x00266513, 0x0027F529, + 0x0029F101, 0x002CF224, 0x00310030, 0x0031122A, + 0x00311420, 0x00311816, 0x0031212C, 0x0031231E, + 0x00312408, 0x00312710, 0x0031312A, 0x0031321E, + 0x00313408, 0x00313623, 0x0031411E, 0x0031433C, + 0x00320007, 0x0032122D, 0x00321420, 0x00321816, + 0x0032212D, 0x0032233C, 0x00322509, 0x00322916, + 0x0032312D, 0x00323420, 0x00323710, 0x00323F2B, + 0x00324308, 0x00324623, 0x00324F25, 0x00325421, + 0x00325F1F, 0x00326421, 0x0032FF29, 0x00331107, + 0x00331308, 0x0033150D, 0x0033211E, 0x00332308, + 0x00332420, 0x00332610, 0x00332929, 0x0033311E, + 0x00333308, 0x0033363B, 0x00333A29, 0x0033413C, + 0x00334320, 0x0033463B, 0x00334A29, 0x0033510A, + 0x00335320, 0x00335824, 0x0033610A, 0x00336321, + 0x00336F12, 0x00337623, 0x00341139, 0x0034153B, + 0x00342108, 0x00342409, 0x00342610, 0x00343108, + 0x00343409, 0x00343610, 0x00344108, 0x0034440D, + 0x00344610, 0x0034510A, 0x00345309, 0x0034553B, + 0x0034610A, 0x00346309, 0x0034F824, 0x00350029, + 0x00352510, 0x00353120, 0x0035330D, 0x00353510, + 0x00354120, 0x0035430D, 0x00354510, 0x00354F28, + 0x0035530D, 0x00355510, 0x00355F1F, 0x00356410, + 0x00359626, 0x0035FF12, 0x00366426, 0x0036FF12, + 0x0037F426, 0x0039D712, 0x003BF612, 0x003DF81F, + 0x00410004, 0x00411207, 0x0041150D, 0x0041212A, + 0x00412420, 0x0041311E, 0x00413308, 0x00413509, + 0x00413F2B, 0x00414208, 0x00420007, 0x0042123C, + 0x00421409, 0x00422107, 0x0042223C, 0x00422409, + 0x00422610, 0x0042313C, 0x00423409, 0x0042363B, + 0x0042413C, 0x00424320, 0x0042463B, 0x00425108, + 0x00425409, 0x0042FF29, 0x00431107, 0x00431320, + 0x0043153B, 0x0043213C, 0x00432320, 0x00432610, + 0x0043313C, 0x00433320, 0x0043353B, 0x00433813, + 0x00434108, 0x00434409, 0x00434610, 0x00435108, + 0x0043553B, 0x00435F25, 0x00436309, 0x0043753B, + 0x0043FF29, 0x00441239, 0x0044143B, 0x00442139, + 0x00442309, 0x0044253B, 0x00443108, 0x00443220, + 0x0044353B, 0x0044410A, 0x00444309, 0x0044453B, + 0x00444813, 0x0044510A, 0x00445309, 0x00445510, + 0x00445F25, 0x0044630D, 0x00450026, 0x00452713, + 0x00453120, 0x0045330D, 0x00453510, 0x00454120, + 0x0045430D, 0x00454510, 0x00455120, 0x0045530D, + 0x00456209, 0x00456410, 0x0045FF12, 0x00466513, + 0x0047FF22, 0x0048FF25, 0x0049F43D, 0x004BFB25, + 0x004EF825, 0x004FFF18, 0x00511339, 0x00512107, + 0x00513409, 0x00520007, 0x00521107, 0x00521320, + 0x00522107, 0x00522409, 0x0052313C, 0x00523320, + 0x0052353B, 0x00524108, 0x00524320, 0x00531139, + 0x00531309, 0x00532139, 0x00532309, 0x0053253B, + 0x00533108, 0x0053340D, 0x00533713, 0x00534108, + 0x0053453B, 0x00534F2B, 0x00535309, 0x00535610, + 0x00535F25, 0x0053643B, 0x00541139, 0x00542139, + 0x00542309, 0x00542613, 0x00543139, 0x00543309, + 0x00543510, 0x00543F2B, 0x00544309, 0x00544510, + 0x00544F28, 0x0054530D, 0x0054FF12, 0x00553613, + 0x00553F2B, 0x00554410, 0x0055510A, 0x0055543B, + 0x00555F25, 0x0055633B, 0x0055FF12, 0x00566513, + 0x00577413, 0x0059FF28, 0x005CC33D, 0x005EFB28, + 0x005FFF18, 0x00611339, 0x00612107, 0x00613320, + 0x0061A724, 0x00621107, 0x0062140B, 0x00622107, + 0x00622320, 0x00623139, 0x00623320, 0x00631139, + 0x0063130C, 0x00632139, 0x00632309, 0x00633139, + 0x00633309, 0x00633626, 0x00633F2B, 0x00634309, + 0x00634F2B, 0x0063543B, 0x0063FF12, 0x0064343B, + 0x00643F2B, 0x0064443B, 0x00645209, 0x00665513, + 0x0066610A, 0x00666526, 0x0067A616, 0x0069843D, + 0x006CF612, 0x006EF326, 0x006FFF18, 0x0071130C, + 0x00721107, 0x00722239, 0x0072291C, 0x0072340B, + 0x00731139, 0x00732239, 0x0073630B, 0x0073FF12, + 0x0074430B, 0x00755426, 0x00776F28, 0x00777410, + 0x0078843D, 0x007CF416, 0x007EF326, 0x007FFF18, + 0x00822239, 0x00831139, 0x0083430B, 0x0084530B, + 0x0087561C, 0x00887F25, 0x00888426, 0x008AF61C, + 0x008F0018, 0x008FFF18, 0x00911107, 0x0093230B, + 0x0094530B, 0x0097743D, 0x00998C25, 0x00999616, + 0x009EF825, 0x009FFF18, 0x00A3430B, 0x00A4530B, + 0x00A7743D, 0x00AA9F2B, 0x00AAA616, 0x00ABD61F, + 0x00AFFF18, 0x00B3330B, 0x00B44426, 0x00B7643D, + 0x00BB971F, 0x00BBB53D, 0x00BEF512, 0x00BFFF18, + 0x00C22139, 0x00C5330E, 0x00C7633D, 0x00CCAF2E, + 0x00CCC616, 0x00CFFF18, 0x00D4440E, 0x00D6420E, + 0x00DDCF2E, 0x00DDD516, 0x00DFFF18, 0x00E4330E, + 0x00E6841C, 0x00EEE61C, 0x00EFFF18, 0x00F3320E, + 0x00F55319, 0x00F8F41C, 0x00FAFF2E, 0x00FF002E, + 0x00FFF10C, 0x00FFF33D, 0x00FFF722, 0x00FFFF18, + 0x01000232, 0x0111113E, 0x01112103, 0x0111311A, + 0x0112111A, 0x01122130, 0x01123130, 0x0112411D, + 0x01131102, 0x01132102, 0x01133102, 0x01141108, + 0x01142136, 0x01143136, 0x01144135, 0x0115223B, + 0x01211103, 0x0121211A, 0x01213130, 0x01221130, + 0x01222130, 0x01223102, 0x01231104, 0x01232104, + 0x01233104, 0x01241139, 0x01241220, 0x01242220, + 0x01251109, 0x0125223B, 0x0125810A, 0x01283212, + 0x0131111A, 0x01312130, 0x0131222C, 0x0131322A, + 0x0132122A, 0x0132222D, 0x0132322D, 0x01331207, + 0x01332234, 0x01333234, 0x01341139, 0x01343134, + 0x01344134, 0x01348134, 0x0135220B, 0x0136110B, + 0x01365224, 0x01411102, 0x01412104, 0x01431239, + 0x01432239, 0x0143320A, 0x01435134, 0x01443107, + 0x01444134, 0x01446134, 0x0145220E, 0x01455134, + 0x0147110E, 0x01511102, 0x01521239, 0x01531239, + 0x01532239, 0x01533107, 0x0155220E, 0x01555134, + 0x0157110E, 0x01611107, 0x01621239, 0x01631239, + 0x01661139, 0x01666134, 0x01711107, 0x01721239, + 0x01745107, 0x0177110C, 0x01811107, 0x01821107, + 0x0185110C, 0x0188210C, 0x01911107, 0x01933139, + 0x01A11107, 0x01A31139, 0x01F5220E, 0x02000001, + 0x02000127, 0x02000427, 0x02000727, 0x02000E2F, + 0x02110000, 0x02111200, 0x02111411, 0x02111827, + 0x02111F2F, 0x02112411, 0x02112715, 0x02113200, + 0x02113411, 0x02113715, 0x02114200, 0x02121200, + 0x02121301, 0x02121F2F, 0x02122200, 0x02122615, + 0x02122F2F, 0x02123311, 0x02123F2F, 0x02124411, + 0x02131211, 0x02132311, 0x02133211, 0x02184415, + 0x02211200, 0x02211311, 0x02211F2F, 0x02212311, + 0x02212F2F, 0x02213211, 0x02221201, 0x02221311, + 0x02221F2F, 0x02222311, 0x02222F2F, 0x02223211, + 0x02223F2F, 0x02231211, 0x02232211, 0x02232F2F, + 0x02233211, 0x02233F2F, 0x02287515, 0x022DAB17, + 0x02311211, 0x02311527, 0x02312211, 0x02321211, + 0x02322211, 0x02322F2F, 0x02323311, 0x02323F2F, + 0x02331211, 0x02332211, 0x02332F2F, 0x02333F2F, + 0x0237FF17, 0x02385615, 0x023D9517, 0x02410027, + 0x02487827, 0x024E3117, 0x024FFF2F, 0x02598627, + 0x025DFF2F, 0x025FFF2F, 0x02687827, 0x026DFA17, + 0x026FFF2F, 0x02796427, 0x027E4217, 0x027FFF2F, + 0x02888727, 0x028EFF2F, 0x028FFF2F, 0x02984327, + 0x029F112F, 0x029FFF2F, 0x02A76527, 0x02AEF717, + 0x02AFFF2F, 0x02B7C827, 0x02BEF917, 0x02BFFF2F, + 0x02C66527, 0x02CD5517, 0x02CFFF2F, 0x02D63227, + 0x02DDD527, 0x02DFFF2B, 0x02E84717, 0x02EEE327, + 0x02EFFF2F, 0x02F54527, 0x02FCF817, 0x02FFEF2B, + 0x02FFFA2F, 0x02FFFE2F, 0x03000127, 0x03000201, + 0x03111200, 0x03122115, 0x03123200, 0x03133211, + 0x03211200, 0x03213127, 0x03221200, 0x03345215, + 0x04000F17, 0x04122F17, 0x043F6515, 0x043FFF17, + 0x044F5527, 0x044FFF17, 0x045F0017, 0x045FFF17, + 0x046F6517, 0x04710027, 0x047F4427, 0x04810027, + 0x048EFA15, 0x048FFF2F, 0x049F4427, 0x049FFF2F, + 0x04AEA727, 0x04AFFF2F, 0x04BE9C15, 0x04BFFF2F, + 0x04CE5427, 0x04CFFF2F, 0x04DE3527, 0x04DFFF17, + 0x04EE4627, 0x04EFFF17, 0x04FEF327, 0x04FFFF2F, + 0x06000F27, 0x069FFF17, 0x06FFFF17, 0x08110017, + 0x08EFFF15, 0xFFFFFF00 +}; + /* @name swb_offsets * Sample offset into the window indicating the beginning of a scalefactor * window band diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index 91262380d4..9d584ebbe5 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -93,6 +93,12 @@ extern const float *const ff_aac_codebook_vectors[]; extern const float *const ff_aac_codebook_vector_vals[]; extern const uint16_t *const ff_aac_codebook_vector_idx[]; +extern const uint16_t ff_aac_ac_msb_cdfs[64][17]; +extern const uint16_t ff_aac_ac_lsb_cdfs[3][4]; +extern const uint8_t ff_aac_ac_lookup_m[742]; +extern const uint32_t ff_aac_ac_hash_m[742]; +extern const uint16_t ff_aac_ac_cf_m[64][17]; + extern const uint16_t * const ff_swb_offset_1024[13]; extern const uint16_t * const ff_swb_offset_960 [13]; extern const uint16_t * const ff_swb_offset_512 [13]; -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 7/8] aactab: add new scalefactor offset tables for 96/768pt windows 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel ` (5 preceding siblings ...) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 6/8] aactab: add tables for the new USAC arithmetic coder Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) Lynne via ffmpeg-devel 7 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne --- libavcodec/aactab.c | 117 ++++++++++++++++++++++++++++++++++++++++++++ libavcodec/aactab.h | 4 ++ 2 files changed, 121 insertions(+) diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index dfb2dfd98d..18afa69bad 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -154,6 +154,10 @@ const uint8_t ff_aac_num_swb_960[] = { 40, 40, 46, 49, 49, 49, 46, 46, 42, 42, 42, 40, 40 }; +const uint8_t ff_aac_num_swb_768[] = { + 37, 37, 41, 43, 43, 43, 43, 43, 39, 39, 39, 37, 37 +}; + const uint8_t ff_aac_num_swb_512[] = { 0, 0, 0, 36, 36, 37, 31, 31, 0, 0, 0, 0, 0 }; @@ -170,6 +174,10 @@ const uint8_t ff_aac_num_swb_120[] = { 12, 12, 12, 14, 14, 14, 15, 15, 15, 15, 15, 15, 15 }; +const uint8_t ff_aac_num_swb_96[] = { + 12, 12, 12, 12, 12, 12, 14, 14, 14, 14, 14, 14, 14 +}; + const uint8_t ff_aac_pred_sfb_max[] = { 33, 33, 38, 40, 40, 40, 41, 41, 37, 37, 37, 34, 34 }; @@ -1806,6 +1814,99 @@ static const uint16_t swb_offset_120_8[] = 0, 4, 8, 12, 16, 20, 24, 28, 36, 44, 52, 60, 72, 88, 108, 120 }; +static const uint16_t swb_offset_768_96[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, + 40, 44, 48, 52, 56, 64, 72, 80, 88, 96, + 108, 120, 132, 144, 156, 172, 188, 212, 240, 276, + 320, 384, 448, 512, 576, 640, 704, 768 +}; + +static const uint16_t swb_offset_768_64[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, + 44, 48, 52, 56, 64, 72, 80, 88, 100, 112, 124, + 140, 156, 172, 192, 216, 240, 268, 304, 344, 384, 424, + 464, 504, 544, 584, 624, 664, 704, 744, 768 +}; + +static const uint16_t swb_offset_768_48[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, + 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, + 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, + 544, 576, 608, 640, 672, 704, 736, 768 +}; + +static const uint16_t swb_offset_768_32[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 48, + 56, 64, 72, 80, 88, 96, 108, 120, 132, 144, 160, 176, + 196, 216, 240, 264, 292, 320, 352, 384, 416, 448, 480, 512, + 544, 576, 608, 640, 672, 704, 736, 768 +}; + +static const uint16_t swb_offset_768_24[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, + 52, 60, 68, 76, 84, 92, 100, 108, 116, 124, 136, 148, + 160, 172, 188, 204, 220, 240, 260, 284, 308, 336, 364, 396, + 432, 468, 508, 552, 600, 652, 704, 768 +}; + +static const uint16_t swb_offset_768_16[] = +{ + 0, 8, 16, 24, 32, 40, 48, 56, 64, + 72, 80, 88, 100, 112, 124, 136, 148, 160, + 172, 184, 196, 212, 228, 244, 260, 280, 300, + 320, 344, 368, 396, 424, 456, 492, 532, 572, + 616, 664, 716, 768 +}; + +static const uint16_t swb_offset_768_8[] = +{ + 0, 12, 24, 36, 48, 60, 72, 84, 96, 108, + 120, 132, 144, 156, 172, 188, 204, 220, 236, 252, + 268, 288, 308, 328, 348, 372, 396, 420, 448, 476, + 508, 544, 580, 620, 664, 712, 764, 768 +}; + +static const uint16_t swb_offset_96_96[] = +{ + 0, 4, 8, 12, 16, 20, 24, + 32, 40, 48, 64, 92, 96 +}; + +static const uint16_t swb_offset_96_64[] = +{ + 0, 4, 8, 12, 16, 20, 24, + 32, 40, 48, 64, 92, 96 +}; + +static const uint16_t swb_offset_96_48[] = +{ + 0, 4, 8, 12, 16, 20, 28, + 36, 44, 56, 68, 80, 96 +}; + +static const uint16_t swb_offset_96_24[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, + 36, 44, 52, 64, 76, 92, 96 +}; + +static const uint16_t swb_offset_96_16[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, + 32, 40, 48, 60, 72, 88, 96 +}; + +static const uint16_t swb_offset_96_8[] = +{ + 0, 4, 8, 12, 16, 20, 24, 28, + 36, 44, 52, 60, 72, 88, 96 +}; + const uint16_t * const ff_swb_offset_1024[] = { swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64, swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32, @@ -1822,6 +1923,14 @@ const uint16_t * const ff_swb_offset_960[] = { swb_offset_960_8 }; +const uint16_t * const ff_swb_offset_768[] = { + swb_offset_768_96, swb_offset_768_96, swb_offset_768_64, + swb_offset_768_48, swb_offset_768_48, swb_offset_768_32, + swb_offset_768_24, swb_offset_768_24, swb_offset_768_16, + swb_offset_768_16, swb_offset_768_16, swb_offset_768_8, + swb_offset_768_8 +}; + const uint16_t * const ff_swb_offset_512[] = { NULL, NULL, NULL, swb_offset_512_48, swb_offset_512_48, swb_offset_512_32, @@ -1856,6 +1965,14 @@ const uint16_t * const ff_swb_offset_120[] = { swb_offset_120_8 }; +const uint16_t * const ff_swb_offset_96[] = { + swb_offset_96_96, swb_offset_96_96, swb_offset_96_96, + swb_offset_96_48, swb_offset_96_48, swb_offset_96_48, + swb_offset_96_24, swb_offset_96_24, swb_offset_96_16, + swb_offset_96_16, swb_offset_96_16, swb_offset_96_8, + swb_offset_96_8 +}; + // @} /* @name ff_tns_max_bands diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index 9d584ebbe5..481fc57d93 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -74,10 +74,12 @@ void ff_aac_float_common_init(void); */ extern const uint8_t ff_aac_num_swb_1024[]; extern const uint8_t ff_aac_num_swb_960 []; +extern const uint8_t ff_aac_num_swb_768 []; extern const uint8_t ff_aac_num_swb_512 []; extern const uint8_t ff_aac_num_swb_480 []; extern const uint8_t ff_aac_num_swb_128 []; extern const uint8_t ff_aac_num_swb_120 []; +extern const uint8_t ff_aac_num_swb_96 []; // @} extern const uint8_t ff_aac_pred_sfb_max []; @@ -101,10 +103,12 @@ extern const uint16_t ff_aac_ac_cf_m[64][17]; extern const uint16_t * const ff_swb_offset_1024[13]; extern const uint16_t * const ff_swb_offset_960 [13]; +extern const uint16_t * const ff_swb_offset_768 [13]; extern const uint16_t * const ff_swb_offset_512 [13]; extern const uint16_t * const ff_swb_offset_480 [13]; extern const uint16_t * const ff_swb_offset_128 [13]; extern const uint16_t * const ff_swb_offset_120 [13]; +extern const uint16_t * const ff_swb_offset_96 [13]; extern const uint8_t ff_tns_max_bands_1024[13]; extern const uint8_t ff_tns_max_bands_512 [13]; -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel ` (6 preceding siblings ...) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 7/8] aactab: add new scalefactor offset tables for 96/768pt windows Lynne via ffmpeg-devel @ 2024-05-16 10:08 ` Lynne via ffmpeg-devel 2024-05-16 10:26 ` Andreas Rheinhardt 7 siblings, 1 reply; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 10:08 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne This commit adds a decoder for the frequency-domain part of USAC. What works: - Mono - Stereo (no prediction) - Stereo (mid/side coding) What doesn't: - Preroll decoding (every single decoder seems faulty or weird?) - Complex stereo prediction Known issues: - Spec incompliance (noise synthesis in particular) - Lack of robustness --- libavcodec/aac/Makefile | 2 +- libavcodec/aac/aacdec.c | 181 ++-- libavcodec/aac/aacdec.h | 179 ++++ libavcodec/aac/aacdec_ac.c | 224 +++++ libavcodec/aac/aacdec_ac.h | 54 ++ libavcodec/aac/aacdec_dsp_template.c | 4 +- libavcodec/aac/aacdec_latm.h | 14 +- libavcodec/aac/aacdec_lpd.c | 192 ++++ libavcodec/aac/aacdec_lpd.h | 33 + libavcodec/aac/aacdec_usac.c | 1230 ++++++++++++++++++++++++++ libavcodec/aac/aacdec_usac.h | 47 + libavcodec/aactab.c | 5 + libavcodec/aactab.h | 3 + 13 files changed, 2093 insertions(+), 75 deletions(-) create mode 100644 libavcodec/aac/aacdec_ac.c create mode 100644 libavcodec/aac/aacdec_ac.h create mode 100644 libavcodec/aac/aacdec_lpd.c create mode 100644 libavcodec/aac/aacdec_lpd.h create mode 100644 libavcodec/aac/aacdec_usac.c create mode 100644 libavcodec/aac/aacdec_usac.h diff --git a/libavcodec/aac/Makefile b/libavcodec/aac/Makefile index 8b0bfff3e5..70b1dca274 100644 --- a/libavcodec/aac/Makefile +++ b/libavcodec/aac/Makefile @@ -3,6 +3,6 @@ clean:: OBJS-$(CONFIG_AAC_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ aac/aacdec_float.o aac/aacdec_usac.o \ - aac/aacdec_ac.o + aac/aacdec_ac.o aac/aacdec_lpd.o OBJS-$(CONFIG_AAC_FIXED_DECODER) += aac/aacdec.o aac/aacdec_tab.o \ aac/aacdec_fixed.o diff --git a/libavcodec/aac/aacdec.c b/libavcodec/aac/aacdec.c index 6f37ac5361..5dcbe46fb9 100644 --- a/libavcodec/aac/aacdec.c +++ b/libavcodec/aac/aacdec.c @@ -40,6 +40,7 @@ #include "aacdec.h" #include "aacdec_tab.h" +#include "aacdec_usac.h" #include "libavcodec/aac.h" #include "libavcodec/aac_defines.h" @@ -535,6 +536,8 @@ static av_cold void flush(AVCodecContext *avctx) } } } + + ff_aac_usac_reset_state(ac, &ac->oc[1]); } /** @@ -993,13 +996,14 @@ static int decode_eld_specific_config(AACDecContext *ac, AVCodecContext *avctx, */ static int decode_audio_specific_config_gb(AACDecContext *ac, AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, + OutputConfiguration *oc, GetBitContext *gb, int get_bit_alignment, int sync_extension) { int i, ret; GetBitContext gbc = *gb; + MPEG4AudioConfig *m4ac = &oc->m4ac; MPEG4AudioConfig m4ac_bak = *m4ac; if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) { @@ -1033,14 +1037,22 @@ static int decode_audio_specific_config_gb(AACDecContext *ac, case AOT_ER_AAC_LC: case AOT_ER_AAC_LD: if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment, - m4ac, m4ac->chan_config)) < 0) + &oc->m4ac, m4ac->chan_config)) < 0) return ret; break; case AOT_ER_AAC_ELD: if ((ret = decode_eld_specific_config(ac, avctx, gb, - m4ac, m4ac->chan_config)) < 0) + &oc->m4ac, m4ac->chan_config)) < 0) + return ret; + break; +#if CONFIG_AAC_DECODER + case AOT_USAC_NOSBR: /* fallthrough */ + case AOT_USAC: + if ((ret = ff_aac_usac_config_decode(ac, avctx, gb, + oc, m4ac->chan_config)) < 0) return ret; break; +#endif default: avpriv_report_missing_feature(avctx, "Audio object type %s%d", @@ -1060,7 +1072,7 @@ static int decode_audio_specific_config_gb(AACDecContext *ac, static int decode_audio_specific_config(AACDecContext *ac, AVCodecContext *avctx, - MPEG4AudioConfig *m4ac, + OutputConfiguration *oc, const uint8_t *data, int64_t bit_size, int sync_extension) { @@ -1080,7 +1092,7 @@ static int decode_audio_specific_config(AACDecContext *ac, if ((ret = init_get_bits(&gb, data, bit_size)) < 0) return ret; - return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0, + return decode_audio_specific_config_gb(ac, avctx, oc, &gb, 0, sync_extension); } @@ -1104,6 +1116,15 @@ static av_cold int decode_close(AVCodecContext *avctx) { AACDecContext *ac = avctx->priv_data; + for (int i = 0; i < 2; i++) { + OutputConfiguration *oc = &ac->oc[i]; + AACUSACConfig *usac = &oc->usac; + for (int j = 0; j < usac->nb_elems; j++) { + AACUsacElemConfig *ec = &usac->elems[i]; + av_freep(&ec->ext.pl_data); + } + } + for (int type = 0; type < FF_ARRAY_ELEMS(ac->che); type++) { for (int i = 0; i < MAX_ELEM_ID; i++) { if (ac->che[type][i]) { @@ -1181,7 +1202,7 @@ av_cold int ff_aac_decode_init(AVCodecContext *avctx) ac->oc[1].m4ac.sample_rate = avctx->sample_rate; if (avctx->extradata_size > 0) { - if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, + if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], avctx->extradata, avctx->extradata_size * 8LL, 1)) < 0) @@ -1549,6 +1570,8 @@ static int decode_pulses(Pulse *pulse, GetBitContext *gb, int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics) { + const int is_usac = ac->oc[1].m4ac.object_type == AOT_USAC || + ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR; int w, filt, i, coef_len, coef_res, coef_compress; const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; @@ -1560,7 +1583,12 @@ int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns, int tmp2_idx; tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); - if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { + if (is_usac) + tns->order[w][filt] = get_bits(gb, 4 - is8); + else + tns->order[w][filt] = get_bits(gb, 5 - (2 * is8)); + + if (tns->order[w][filt] > tns_max_order) { av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n", tns->order[w][filt], tns_max_order); @@ -1598,6 +1626,7 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, { int idx; int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; + cpe->max_sfb_ste = cpe->ch[0].ics.max_sfb; if (ms_present == 1) { for (idx = 0; idx < max_idx; idx++) cpe->ms_mask[idx] = get_bits1(gb); @@ -2182,42 +2211,19 @@ static int aac_decode_er_frame(AVCodecContext *avctx, AVFrame *frame, return 0; } -static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, - int *got_frame_ptr, GetBitContext *gb, - const AVPacket *avpkt) +static int decode_frame_ga(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr) { - AACDecContext *ac = avctx->priv_data; - ChannelElement *che = NULL, *che_prev = NULL; + int err; + int is_dmono; + int elem_id; enum RawDataBlockType elem_type, che_prev_type = TYPE_END; - int err, elem_id; - int samples = 0, multiplier, audio_found = 0, pce_found = 0; - int is_dmono, sce_count = 0; - int payload_alignment; uint8_t che_presence[4][MAX_ELEM_ID] = {{0}}; + ChannelElement *che = NULL, *che_prev = NULL; + int samples = 0, multiplier, audio_found = 0, pce_found = 0, sce_count = 0; + AVFrame *frame = ac->frame; - ac->frame = frame; - - if (show_bits(gb, 12) == 0xfff) { - if ((err = parse_adts_frame_header(ac, gb)) < 0) { - av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); - goto fail; - } - if (ac->oc[1].m4ac.sampling_index > 12) { - av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index); - err = AVERROR_INVALIDDATA; - goto fail; - } - } - - if ((err = frame_configure_elements(avctx)) < 0) - goto fail; - - // The AV_PROFILE_AAC_* defines are all object_type - 1 - // This may lead to an undefined profile being signaled - ac->avctx->profile = ac->oc[1].m4ac.object_type - 1; - - payload_alignment = get_bits_count(gb); - ac->tags_mapped = 0; + int payload_alignment = get_bits_count(gb); // parse while ((elem_type = get_bits(gb, 3)) != TYPE_END) { elem_id = get_bits(gb, 4); @@ -2225,28 +2231,23 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, if (avctx->debug & FF_DEBUG_STARTCODE) av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id); - if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE) { - err = AVERROR_INVALIDDATA; - goto fail; - } + if (!avctx->ch_layout.nb_channels && elem_type != TYPE_PCE) + return AVERROR_INVALIDDATA; if (elem_type < TYPE_DSE) { if (che_presence[elem_type][elem_id]) { int error = che_presence[elem_type][elem_id] > 1; av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n", elem_type, elem_id); - if (error) { - err = AVERROR_INVALIDDATA; - goto fail; - } + if (error) + return AVERROR_INVALIDDATA; } che_presence[elem_type][elem_id]++; if (!(che=ff_aac_get_che(ac, elem_type, elem_id))) { av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024; che->present = 1; @@ -2283,10 +2284,8 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, int tags; int pushed = push_output_configuration(ac); - if (pce_found && !pushed) { - err = AVERROR_INVALIDDATA; - goto fail; - } + if (pce_found && !pushed) + return AVERROR_INVALIDDATA; tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb, payload_alignment); @@ -2312,8 +2311,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, elem_id += get_bits(gb, 8) - 1; if (get_bits_left(gb) < 8 * elem_id) { av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } err = 0; while (elem_id > 0) { @@ -2337,19 +2335,16 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, } if (err) - goto fail; + return err; if (get_bits_left(gb) < 3) { av_log(avctx, AV_LOG_ERROR, overread_err); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } } - if (!avctx->ch_layout.nb_channels) { - *got_frame_ptr = 0; + if (!avctx->ch_layout.nb_channels) return 0; - } multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0; samples <<= multiplier; @@ -2364,16 +2359,17 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, if (!ac->frame->data[0] && samples) { av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); - err = AVERROR_INVALIDDATA; - goto fail; + return AVERROR_INVALIDDATA; } if (samples) { ac->frame->nb_samples = samples; ac->frame->sample_rate = avctx->sample_rate; - } else + *got_frame_ptr = 1; + } else { av_frame_unref(ac->frame); - *got_frame_ptr = !!samples; + *got_frame_ptr = 0; + } /* for dual-mono audio (SCE + SCE) */ is_dmono = ac->dmono_mode && sce_count == 2 && @@ -2387,6 +2383,59 @@ static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, } return 0; +} + +static int aac_decode_frame_int(AVCodecContext *avctx, AVFrame *frame, + int *got_frame_ptr, GetBitContext *gb, + const AVPacket *avpkt) +{ + int err; + AACDecContext *ac = avctx->priv_data; + + ac->frame = frame; + *got_frame_ptr = 0; + + if (show_bits(gb, 12) == 0xfff) { + if ((err = parse_adts_frame_header(ac, gb)) < 0) { + av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); + goto fail; + } + if (ac->oc[1].m4ac.sampling_index > 12) { + av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index); + err = AVERROR_INVALIDDATA; + goto fail; + } + } + + if ((err = frame_configure_elements(avctx)) < 0) + goto fail; + + // The AV_PROFILE_AAC_* defines are all object_type - 1 + // This may lead to an undefined profile being signaled + ac->avctx->profile = ac->oc[1].m4ac.object_type - 1; + + ac->tags_mapped = 0; + + if ((ac->oc[1].m4ac.object_type == AOT_USAC) || + (ac->oc[1].m4ac.object_type == AOT_USAC_NOSBR)) { + if (ac->is_fixed) { + avpriv_report_missing_feature(ac->avctx, + "AAC USAC fixed-point decoding"); + return AVERROR_PATCHWELCOME; + } +#if CONFIG_AAC_DECODER + err = ff_aac_usac_decode_frame(avctx, ac, gb, got_frame_ptr); + if (err < 0) + goto fail; +#endif + } else { + err = decode_frame_ga(avctx, ac, gb, got_frame_ptr); + if (err < 0) + goto fail; + } + + return err; + fail: pop_output_configuration(ac); return err; @@ -2414,7 +2463,7 @@ static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, if (new_extradata) { /* discard previous configuration */ ac->oc[1].status = OC_NONE; - err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac, + err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1], new_extradata, new_extradata_size * 8LL, 1); if (err < 0) { diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h index 20545a24d4..3e6592cf0e 100644 --- a/libavcodec/aac/aacdec.h +++ b/libavcodec/aac/aacdec.h @@ -42,6 +42,8 @@ #include "libavcodec/avcodec.h" #include "libavcodec/mpeg4audio.h" +#include "aacdec_ac.h" + typedef struct AACDecContext AACDecContext; /** @@ -69,6 +71,32 @@ enum CouplingPoint { AFTER_IMDCT = 3, }; +enum AACUsacElem { + ID_USAC_SCE = 0, + ID_USAC_CPE = 1, + ID_USAC_LFE = 2, + ID_USAC_EXT = 3, +}; + +enum ExtensionHeaderType { + ID_CONFIG_EXT_FILL = 0, + ID_CONFIG_EXT_LOUDNESS_INFO = 2, + ID_CONFIG_EXT_STREAM_ID = 7, +}; + +enum AACUsacExtension { + ID_EXT_ELE_FILL, + ID_EXT_ELE_MPEGS, + ID_EXT_ELE_SAOC, + ID_EXT_ELE_AUDIOPREROLL, + ID_EXT_ELE_UNI_DRC, +}; + +enum AACUSACLoudnessExt { + UNIDRCLOUDEXT_TERM = 0x0, + UNIDRCLOUDEXT_EQ = 0x1, +}; + // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. #define RENAME_FIXED(name) name ## _fixed @@ -93,6 +121,40 @@ typedef struct LongTermPrediction { int8_t used[MAX_LTP_LONG_SFB]; } LongTermPrediction; +/* Per channel core mode */ +typedef struct AACUsacElemData { + uint8_t core_mode; + uint8_t scale_factor_grouping; + + /* Timewarping ratio */ +#define NUM_TW_NODES 16 + uint8_t tw_ratio[NUM_TW_NODES]; + + struct { + uint8_t acelp_core_mode : 3; + uint8_t lpd_mode : 5; + + uint8_t bpf_control_info : 1; + uint8_t core_mode_last : 1; + uint8_t fac_data_present : 1; + + int last_lpd_mode; + } ldp; + + struct { + unsigned int seed; + uint8_t level : 3; + uint8_t offset : 5; + } noise; + + struct { + uint8_t gain; + uint32_t kv[8 /* (1024 / 16) / 8 */][8]; + } fac; + + AACArithState ac; +} AACUsacElemData; + /** * Individual Channel Stream */ @@ -145,6 +207,7 @@ typedef struct ChannelCoupling { */ typedef struct SingleChannelElement { IndividualChannelStream ics; + AACUsacElemData ue; ///< USAC element data TemporalNoiseShaping tns; enum BandType band_type[128]; ///< band types int sfo[128]; ///< scalefactor offsets @@ -163,25 +226,141 @@ typedef struct SingleChannelElement { }; } SingleChannelElement; +typedef struct AACUsacStereo { + uint8_t common_window; + uint8_t common_tw; + + uint8_t ms_mask_mode; + uint8_t config_idx; + + struct { + uint8_t use_prev_frame; + uint8_t pred_dir; + uint8_t delta_code_time; + uint8_t pred_used[8][64]; + + AVComplexFloat pred[8][64]; + } cplx; +} AACUsacStereo; + /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct ChannelElement { int present; // CPE specific + uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band // shared SingleChannelElement ch[2]; // CCE specific ChannelCoupling coup; + // USAC stereo coupling data + AACUsacStereo us; } ChannelElement; +typedef struct AACUSACLoudnessInfo { + uint8_t drc_set_id : 6; + uint8_t downmix_id : 7; + struct { + uint16_t lvl : 12; + uint8_t present : 1; + } sample_peak; + + struct { + uint16_t lvl : 12; + uint8_t measurement : 4; + uint8_t reliability : 2; + uint8_t present : 1; + } true_peak; + + uint8_t nb_measurements : 4; + struct { + uint8_t method_def : 4; + uint8_t method_val; + uint8_t measurement : 4; + uint8_t reliability : 2; + } measurements[16]; +} AACUSACLoudnessInfo; + +typedef struct AACUsacElemConfig { + enum AACUsacElem type; + + uint8_t tw_mdct : 1; + uint8_t noise_fill : 1; + + uint8_t stereo_config_index; + + struct { + int ratio; + + uint8_t harmonic_sbr : 1; /* harmonicSBR */ + uint8_t bs_intertes : 1; /* bs_interTes */ + uint8_t bs_pvc : 1; /* bs_pvc */ + + struct { + uint8_t start_freq; /* dflt_start_freq */ + uint8_t stop_freq; /* dflt_stop_freq */ + + uint8_t freq_scale; /* dflt_freq_scale */ + uint8_t alter_scale : 1; /* dflt_alter_scale */ + uint8_t noise_scale; /* dflt_noise_scale */ + + uint8_t limiter_bands; /* dflt_limiter_bands */ + uint8_t limiter_gains; /* dflt_limiter_gains */ + uint8_t interpol_freq : 1; /* dflt_interpol_freq */ + uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ + } dflt; + } sbr; + + struct { + uint8_t freq_res; /* bsFreqRes */ + uint8_t fixed_gain; /* bsFixedGainDMX */ + uint8_t temp_shape_config; /* bsTempShapeConfig */ + uint8_t decorr_config; /* bsDecorrConfig */ + uint8_t high_rate_mode : 1; /* bsHighRateMode */ + uint8_t phase_coding : 1; /* bsPhaseCoding */ + + uint8_t otts_bands_phase; /* bsOttBandsPhase */ + uint8_t residual_coding; /* bsResidualCoding */ + uint8_t residual_bands; /* bsResidualBands */ + uint8_t pseudo_lr : 1; /* bsPseudoLr */ + uint8_t env_quant_mode : 1; /* bsEnvQuantMode */ + } mps; + + struct { + enum AACUsacExtension type; + uint8_t payload_frag; + uint32_t default_len; + uint32_t pl_data_offset; + uint8_t *pl_data; + } ext; +} AACUsacElemConfig; + +typedef struct AACUSACConfig { + uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ + uint8_t rate_idx; + uint16_t core_frame_len; + uint16_t stream_identifier; + + AACUsacElemConfig elems[64]; + int nb_elems; + + struct { + uint8_t nb_album; + AACUSACLoudnessInfo album_info[64]; + uint8_t nb_info; + AACUSACLoudnessInfo info[64]; + } loudness; +} AACUSACConfig; + typedef struct OutputConfiguration { MPEG4AudioConfig m4ac; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; AVChannelLayout ch_layout; enum OCStatus status; + AACUSACConfig usac; } OutputConfiguration; /** diff --git a/libavcodec/aac/aacdec_ac.c b/libavcodec/aac/aacdec_ac.c new file mode 100644 index 0000000000..326d716bd3 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.c @@ -0,0 +1,224 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavcodec/aactab.h" +#include "aacdec_ac.h" + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N) +{ + float ratio; + if (reset) { + memset(state->last, 0, sizeof(state->last)); + state->last_len = N; + memset(state->cur, 0, sizeof(state->cur)); + state->cur[3] = 0; + state->cur[2] = 0; + state->cur[1] = 0; + state->cur[0] = 1; + state->last[0] = 0 << 12; + state->state_pre = 0; + return 0; + } else if (state->last_len != N) { + int i; + uint8_t last[512 /* 2048 / 4 */]; + memcpy(last, state->last, sizeof(last)); + + ratio = state->last_len / (float)N; + for (i = 0; i < N/2; i++) { + int k = (int)(i * ratio); + state->last[i] = last[k]; + } + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; + + state->last_len = N; + } + + memset(state->cur, 0, sizeof(state->cur)); + state->cur[3] = 0; + state->cur[2] = 0; + state->cur[1] = 0; + state->cur[0] = 1; + + state->state_pre = state->last[0] << 12; + return state->last[0] << 12; +} + +extern int ec_debug; + +int trig = 0; + +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N) +{ + c = state->state_pre >> 8; + c = c + (state->last[i + 1] << 8); + c = (c << 4); + c += state->cur[1]; + + state->state_pre = c; + + if (i > 3 && + ((state->cur[3] + state->cur[2] + state->cur[1]) < 5)) + return c + 0x10000; + + return c; +} + +uint32_t ff_aac_ac_get_pk(uint32_t c) +{ + int i_min = -1; + int i, j; + int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1; + while ((i_max - i_min) > 1) { + i = i_min + ((i_max - i_min) / 2); + j = ff_aac_ac_hash_m[i]; + if (c < (j >> 8)) + i_max = i; + else if (c > (j >> 8)) + i_min = i; + else + return (j & 0xFF); + } + return ff_aac_ac_lookup_m[i_max]; +} + +void ff_aac_ac_update_context(AACArithState *state, int idx, + uint16_t a, uint16_t b) +{ + state->cur[0] = a + b + 1; + if (state->cur[0] > 0xF) + state->cur[0] = 0xF; + + state->cur[3] = state->cur[2]; + state->cur[2] = state->cur[1]; + state->cur[1] = state->cur[0]; + + state->last[idx] = state->cur[0]; +} + +/* Initialize AC */ +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb) +{ + ac->low = 0; + ac->high = UINT16_MAX; + ac->val = get_bits(gb, 16); +} + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len) +{ + int val = ac->val; + int low = ac->low; + int high = ac->high; + + int rng = high - low + 1; + int c = ((((int)(val - low + 1)) << 14) - ((int)1)); + + /* Note: this could be done faster via heuristics, the total number of + * configurations is low */ + const uint16_t *p = cdf - 1; + const uint16_t *q; + + switch (cdf_len) { + case 2: + if ((p[1] * rng) > c) + p += 1; + break; + case 4: + if ((p[2] * rng) > c) + p += 2; + if ((p[1] * rng) > c) + p += 1; + break; + case 17: + /* First check if the current probability is even met at all */ + if ((p[1] * rng) <= c) + break; + p += 1; + for (int i = 8; i >= 1; i >>= 1) + if ((p[i] * rng) > c) + p += i; + break; + case 27: + const uint16_t *p_24 = p + 24; + + if ((p[16] * rng) > c) + p += 16; + if ((p[8] * rng) > c) + p += 8; + if (p != p_24) + if ((p[4] * rng) > c) + p += 4; + if ((p[2] * rng) > c) + p += 2; + + if (p != &p_24[2]) + if ((p[1] * rng) > c) + p += 1; + break; + default: + /* This should never happen */ + av_assert2(0); + } + + int sym = (int)((ptrdiff_t)(p - cdf)) + 1; + if (sym) + high = low + ((rng * cdf[sym - 1]) >> 14) - 1; + low += (rng * cdf[sym]) >> 14; + + /* This loop could be done faster */ + while (1) { + if (high < 32768) { + ; + } else if (low >= 32768) { + val -= 32768; + low -= 32768; + high -= 32768; + } else if (low >= 16384 && high < 49152) { + val -= 16384; + low -= 16384; + high -= 16384; + } else { + break; + } + low += low; + high += high + 1; + val = (val << 1) | get_bits1(gb); + }; + + ac->low = low; + ac->high = high; + ac->val = val; + + return sym; +} + +void ff_aac_ac_finish(AACArithState *state, int offset, int N) +{ + int i; + + for (i = offset; i < N/2; i++) + state->last[i] = 1; + + for (; i < FF_ARRAY_ELEMS(state->last); i++) + state->last[i] = 0; +} diff --git a/libavcodec/aac/aacdec_ac.h b/libavcodec/aac/aacdec_ac.h new file mode 100644 index 0000000000..ef96bed770 --- /dev/null +++ b/libavcodec/aac/aacdec_ac.h @@ -0,0 +1,54 @@ +/* + * AAC definitions and structures + * Copyright (c) 2024 Lynne + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AACDEC_AC_H +#define AVCODEC_AACDEC_AC_H + +#include "libavcodec/get_bits.h" + +typedef struct AACArithState { + uint8_t last[512 /* 2048 / 4 */]; + int last_len; + uint8_t cur[4]; + uint16_t state_pre; +} AACArithState; + +typedef struct AACArith { + uint16_t low; + uint16_t high; + uint16_t val; +} AACArith; + +#define FF_AAC_AC_ESCAPE 16 + +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len); +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len); +uint32_t ff_aac_ac_get_pk(uint32_t c); + +void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b); +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb); + +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, + const uint16_t *cdf, uint16_t cdf_len); + +void ff_aac_ac_finish(AACArithState *state, int offset, int nb); + +#endif /* AVCODEC_AACDEC_AC_H */ diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c index 59a69d88f3..8d31af22f8 100644 --- a/libavcodec/aac/aacdec_dsp_template.c +++ b/libavcodec/aac/aacdec_dsp_template.c @@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); const uint16_t *offsets = ics->swb_offset; for (int g = 0; g < ics->num_window_groups; g++) { - for (int sfb = 0; sfb < ics->max_sfb; sfb++) { - const int idx = g*ics->max_sfb + sfb; + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + const int idx = g*cpe->max_sfb_ste + sfb; if (cpe->ms_mask[idx] && cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h index e40a2fe1a7..047c11e0fb 100644 --- a/libavcodec/aac/aacdec_latm.h +++ b/libavcodec/aac/aacdec_latm.h @@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, { AACDecContext *ac = &latmctx->aac_ctx; AVCodecContext *avctx = ac->avctx; - MPEG4AudioConfig m4ac = { 0 }; + OutputConfiguration oc = { 0 }; + MPEG4AudioConfig *m4ac = &oc.m4ac; GetBitContext gbc; int config_start_bit = get_bits_count(gb); int sync_extension = 0; @@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, if (get_bits_left(gb) <= 0) return AVERROR_INVALIDDATA; - bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, + bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc, &gbc, config_start_bit, sync_extension); @@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, asclen = bits_consumed; if (!latmctx->initialized || - ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || - ac->oc[1].m4ac.chan_config != m4ac.chan_config) { + ac->oc[1].m4ac.sample_rate != m4ac->sample_rate || + ac->oc[1].m4ac.chan_config != m4ac->chan_config) { if (latmctx->initialized) { - av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); + av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", + m4ac->sample_rate, m4ac->chan_config); } else { av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); } @@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, } else { push_output_configuration(&latmctx->aac_ctx); if ((err = decode_audio_specific_config( - &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, + &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1], avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { pop_output_configuration(&latmctx->aac_ctx); return err; diff --git a/libavcodec/aac/aacdec_lpd.c b/libavcodec/aac/aacdec_lpd.c new file mode 100644 index 0000000000..be39e2c175 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.c @@ -0,0 +1,192 @@ +/* + * Copyright (c) 2024 Lynne <dev@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aacdec_lpd.h" +#include "aacdec_usac.h" + +const uint8_t ff_aac_lpd_mode_tab[32][4] = { + { 0, 0, 0, 0 }, + { 1, 0, 0, 0 }, + { 0, 1, 0, 0 }, + { 1, 1, 0, 0 }, + { 0, 0, 1, 0 }, + { 1, 0, 1, 0 }, + { 0, 1, 1, 0 }, + { 1, 1, 1, 0 }, + { 0, 0, 0, 1 }, + { 1, 0, 0, 1 }, + { 0, 1, 0, 1 }, + { 1, 1, 0, 1 }, + { 0, 0, 1, 1 }, + { 1, 0, 1, 1 }, + { 0, 1, 1, 1 }, + { 1, 1, 1, 1 }, + { 2, 2, 0, 0 }, + { 2, 2, 1, 0 }, + { 2, 2, 0, 1 }, + { 2, 2, 1, 1 }, + { 0, 0, 2, 2 }, + { 1, 0, 2, 2 }, + { 0, 1, 2, 2 }, + { 1, 1, 2, 2 }, + { 2, 2, 2, 2 }, + { 3, 3, 3, 3 }, + /* Larger values are reserved, but permit them for resilience */ + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, + { 0, 0, 0, 0 }, +}; + +static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn) +{ + if (nk_mode == 1) { + for (int k = 0; k < no_qn; k++) { + qn[k] = ff_aac_get_vlclbf(gb); + if (qn[k]) + qn[k]++; + } + return; + } + + for (int k = 0; k < no_qn; k++) + qn[k] = get_bits(gb, 2) + 2; + + if (nk_mode == 2) { + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + qn[k] = ff_aac_get_vlclbf(gb); + if (qn[k]) + qn[k] += 4; + } + } + return; + } + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + int qn_ext = ff_aac_get_vlclbf(gb); + switch (qn_ext) { + case 0: qn[k] = 5; break; + case 1: qn[k] = 6; break; + case 2: qn[k] = 0; break; + default: qn[k] = qn_ext + 4; break; + } + } + } +} + +static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv, + int nk_mode, int no_qn) +{ + int n, nk; + + int qn[2]; + parse_qn(gb, qn, nk_mode, no_qn); + + for (int k = 0; k < no_qn; k++) { + if (qn[k] > 4) { + nk = (qn[k] - 3) / 2; + n = qn[k] - nk*2; + } else { + nk = 0; + n = qn[k]; + } + } + + int idx = get_bits(gb, 4*n); + + if (nk > 0) + for (int i = 0; i < 8; i++) + kv[i] = get_bits(gb, nk); + + return 0; +} + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len) +{ + int ret; + if (use_gain) + ce->fac.gain = get_bits(gb, 7); + + for (int i = 0; i < len/8; i++) { + ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1); + if (ret < 0) + return ret; + } + + return 0; +} + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb) +{ + ce->ldp.acelp_core_mode = get_bits(gb, 3); + ce->ldp.lpd_mode = get_bits(gb, 5); + + ce->ldp.bpf_control_info = get_bits1(gb); + ce->ldp.core_mode_last = get_bits1(gb); + ce->ldp.fac_data_present = get_bits1(gb); + + const uint8_t *mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode]; + + int first_ldp_flag = !ce->ldp.core_mode_last; + int first_tcx_flag = 1; + if (first_ldp_flag) + ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */ + + int k = 0; + while (k < 0) { + if (!k) { + if (ce->ldp.core_mode_last && ce->ldp.fac_data_present) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } else { + if (!ce->ldp.last_lpd_mode && mod[k] > 0 || + ce->ldp.last_lpd_mode && !mod[k]) + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); + } + if (!mod[k]) { +// parse_acelp_coding(); + ce->ldp.last_lpd_mode = 0; + k++; + } else { +// parse_tcx_coding(); + ce->ldp.last_lpd_mode = mod[k]; + k += (1 << (mod[k] - 1)); + first_tcx_flag = 0; + } + } + +// parse_lpc_data(first_lpd_flag); + + if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) { + uint16_t len_8 = usac->core_frame_len / 8; + uint16_t len_16 = usac->core_frame_len / 16; + uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16; + int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len); + if (ret < 0) + return ret; + } + + return 0; +} diff --git a/libavcodec/aac/aacdec_lpd.h b/libavcodec/aac/aacdec_lpd.h new file mode 100644 index 0000000000..924ff75e52 --- /dev/null +++ b/libavcodec/aac/aacdec_lpd.h @@ -0,0 +1,33 @@ +/* + * Copyright (c) 2024 Lynne <dev@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AAC_AACDEC_LPD_H +#define AVCODEC_AAC_AACDEC_LPD_H + +#include "aacdec.h" +#include "libavcodec/get_bits.h" + +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, + int use_gain, int len); + +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemData *ce, GetBitContext *gb); + +#endif /* AVCODEC_AAC_AACDEC_LPD_H */ diff --git a/libavcodec/aac/aacdec_usac.c b/libavcodec/aac/aacdec_usac.c new file mode 100644 index 0000000000..4b48c4d6ca --- /dev/null +++ b/libavcodec/aac/aacdec_usac.c @@ -0,0 +1,1230 @@ +/* + * Copyright (c) 2024 Lynne <dev@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "aacdec_usac.h" +#include "aacdec_tab.h" +#include "aacdec_lpd.h" +#include "aacdec_ac.h" + +#include "libavcodec/opusdsp.h" +#include "libavcodec/aactab.h" +#include "libavutil/mem.h" +#include "libavcodec/mpeg4audio.h" + +/* Number of scalefactor bands per complex prediction band, equal to 2. */ +#define SFB_PER_PRED_BAND 2 + +static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3) +{ + uint32_t val = get_bits(gb, nb1); + if (val < ((1 << nb1) - 1)) + return val; + + val += get_bits(gb, nb2); + if (val == ((1 << nb2) - 1)) + val += get_bits(gb, nb3); + + return val; +} + +static int aac_usac_samplerate[] = { + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, + 16000, 12000, 11025, 8000, 7350, -1, -1, 57600, 51200, + 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, -1, -1, -1, -1, +}; + +/* Finish later */ +static const enum AVChannel usac_ch_pos_to_av[64] = { + [0] = AV_CHAN_FRONT_LEFT, + [1] = AV_CHAN_FRONT_RIGHT, + [2] = AV_CHAN_FRONT_CENTER, + [3] = AV_CHAN_LOW_FREQUENCY, + [4] = AV_CHAN_BACK_LEFT, + [5] = AV_CHAN_BACK_RIGHT, + [6] = AV_CHAN_FRONT_LEFT_OF_CENTER, + [7] = AV_CHAN_FRONT_RIGHT_OF_CENTER, + [8] = 0, +}; + +static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info, + GetBitContext *gb) +{ + info->drc_set_id = get_bits(gb, 6); + info->downmix_id = get_bits(gb, 7); + + if ((info->sample_peak.present = get_bits1(gb))) /* samplePeakLevelPresent */ + info->sample_peak.lvl = get_bits(gb, 12); + + if ((info->true_peak.present = get_bits1(gb))) { /* truePeakLevelPresent */ + info->true_peak.lvl = get_bits(gb, 12); + info->true_peak.measurement = get_bits(gb, 4); + info->true_peak.reliability = get_bits(gb, 2); + } + + info->nb_measurements = get_bits(gb, 4); + for (int i = 0; i < info->nb_measurements; i++) { + info->measurements[i].method_def = get_bits(gb, 4); + info->measurements[i].method_val = ff_aac_get_vlclbf(gb); + info->measurements[i].measurement = get_bits(gb, 4); + info->measurements[i].reliability = get_bits(gb, 2); + } + + return 0; +} + +static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac, + GetBitContext *gb) +{ + int ret; + + usac->loudness.nb_album = get_bits(gb, 6); /* loudnessInfoAlbumCount */ + usac->loudness.nb_info = get_bits(gb, 6); /* loudnessInfoCount */ + + for (int i = 0; i < usac->loudness.nb_album; i++) { + ret = decode_loudness_info(ac, &usac->loudness.album_info[i], gb); + if (ret < 0) + return ret; + } + + for (int i = 0; i < usac->loudness.nb_info; i++) { + ret = decode_loudness_info(ac, &usac->loudness.info[i], gb); + if (ret < 0) + return ret; + } + + if (get_bits1(gb)) { /* loudnessInfoSetExtPresent */ + enum AACUSACLoudnessExt type; + while ((type = get_bits(gb, 4)) != UNIDRCLOUDEXT_TERM) { + uint8_t size_bits = get_bits(gb, 4) + 4; + uint8_t bit_size = get_bits(gb, size_bits) + 1; + switch (type) { + case UNIDRCLOUDEXT_EQ: + avpriv_report_missing_feature(ac->avctx, "loudnessInfoV1"); + return AVERROR_PATCHWELCOME; + default: + for (int i = 0; i < bit_size; i++) + skip_bits1(gb); + } + } + } + + return 0; +} + +static void decode_usac_sbr_data(AACUsacElemConfig *e, GetBitContext *gb) +{ + uint8_t header_extra1; + uint8_t header_extra2; + + e->sbr.harmonic_sbr = get_bits1(gb); /* harmonicSBR */ + e->sbr.bs_intertes = get_bits1(gb); /* bs_interTes */ + e->sbr.bs_pvc = get_bits1(gb); /* bs_pvc */ + + e->sbr.dflt.start_freq = get_bits(gb, 4); /* dflt_start_freq */ + e->sbr.dflt.stop_freq = get_bits(gb, 4); /* dflt_stop_freq */ + + header_extra1 = get_bits1(gb); /* dflt_header_extra1 */ + header_extra2 = get_bits1(gb); /* dflt_header_extra2 */ + + e->sbr.dflt.freq_scale = 2; + e->sbr.dflt.alter_scale = 1; + e->sbr.dflt.noise_scale = 2; + if (header_extra1) { + e->sbr.dflt.freq_scale = get_bits(gb, 2); /* dflt_freq_scale */ + e->sbr.dflt.alter_scale = get_bits1(gb); /* dflt_alter_scale */ + e->sbr.dflt.noise_scale = get_bits(gb, 2); /* dflt_noise_scale */ + } + + e->sbr.dflt.limiter_bands = 2; + e->sbr.dflt.limiter_gains = 2; + e->sbr.dflt.interpol_freq = 1; + e->sbr.dflt.smoothing_mode = 1; + if (header_extra2) { + e->sbr.dflt.limiter_bands = get_bits(gb, 2); /* dflt_limiter_bands */ + e->sbr.dflt.limiter_gains = get_bits(gb, 2); /* dflt_limiter_gains */ + e->sbr.dflt.interpol_freq = get_bits1(gb); /* dflt_interpol_freq */ + e->sbr.dflt.smoothing_mode = get_bits1(gb); /* dflt_smoothing_mode */ + } +} + +static void decode_usac_element_core(AACUsacElemConfig *e, + GetBitContext *gb, + int sbr_ratio) +{ + e->tw_mdct = get_bits1(gb); /* tw_mdct */ + e->noise_fill = get_bits1(gb); + e->sbr.ratio = sbr_ratio; +} + +static void decode_usac_element_pair(AACUsacElemConfig *e, GetBitContext *gb) +{ + e->stereo_config_index = 0; + if (e->sbr.ratio) { + decode_usac_sbr_data(e, gb); + e->stereo_config_index = get_bits(gb, 2); + } + if (e->stereo_config_index) { + e->mps.freq_res = get_bits(gb, 3); /* bsFreqRes */ + e->mps.fixed_gain = get_bits(gb, 3); /* bsFixedGainDMX */ + e->mps.temp_shape_config = get_bits(gb, 2); /* bsTempShapeConfig */ + e->mps.decorr_config = get_bits(gb, 2); /* bsDecorrConfig */ + e->mps.high_rate_mode = get_bits1(gb); /* bsHighRateMode */ + e->mps.phase_coding = get_bits1(gb); /* bsPhaseCoding */ + + if (get_bits1(gb)) /* bsOttBandsPhasePresent */ + e->mps.otts_bands_phase = get_bits(gb, 5); /* bsOttBandsPhase */ + + e->mps.residual_coding = e->stereo_config_index >= 2; /* bsResidualCoding */ + if (e->mps.residual_coding) { + e->mps.residual_bands = get_bits(gb, 5); /* bsResidualBands */ + e->mps.pseudo_lr = get_bits1(gb); /* bsPseudoLr */ + } + if (e->mps.temp_shape_config == 2) + e->mps.env_quant_mode = get_bits1(gb); /* bsEnvQuantMode */ + } +} + +static int decode_usac_extension(AACDecContext *ac, AACUsacElemConfig *e, + GetBitContext *gb) +{ + int len = 0, ext_config_len; + + e->ext.type = get_escaped_value(gb, 4, 8, 16); /* usacExtElementType */ + ext_config_len = get_escaped_value(gb, 4, 8, 16); /* usacExtElementConfigLength */ + + if (get_bits1(gb)) /* usacExtElementDefaultLengthPresent */ + len = get_escaped_value(gb, 8, 16, 0) + 1; + + e->ext.default_len = len; + e->ext.payload_frag = get_bits1(gb); /* usacExtElementPayloadFrag */ + + av_log(ac->avctx, AV_LOG_INFO, "Extension present: type %i, len %i\n", + e->ext.type, ext_config_len); + + switch (e->ext.type) { +#if 0 /* Skip unsupported values */ + case ID_EXT_ELE_MPEGS: + break; + case ID_EXT_ELE_SAOC: + break; + case ID_EXT_ELE_UNI_DRC: + break; +#endif + case ID_EXT_ELE_FILL: + break; /* This is what the spec does */ + case ID_EXT_ELE_AUDIOPREROLL: + /* No configuration needed - fallthrough (len should be 0) */ + default: + skip_bits(gb, 8*ext_config_len); + break; + }; + + return 0; +} + +int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc) +{ + AACUSACConfig *usac = &oc->usac; + int elem_id[3 /* SCE, CPE, LFE */] = { 0, 0, 0 }; + + /* Initialize state */ + for (int i = 0; i < usac->nb_elems; i++) { + AACUsacElemConfig *e = &usac->elems[i]; + if (e->type == ID_USAC_SCE || e->type == ID_USAC_CPE) { + ChannelElement *che; + enum RawDataBlockType type; + int id; + if (e->type == ID_USAC_SCE) { + type = TYPE_SCE; + id = elem_id[0]++; + } else { + type = TYPE_CPE; + id = elem_id[1]++; + } + + che = ff_aac_get_che(ac, type, id); + if (che) { + che->ch[0].ue.noise.seed = 0x3039; + if (e->type == ID_USAC_CPE) + che->ch[1].ue.noise.seed = 0x10932; + } + } + } + + return 0; +} + +/* UsacConfig */ +int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx, + GetBitContext *gb, OutputConfiguration *oc, + int channel_config) +{ + int ret, idx; + uint8_t freq_idx; + uint8_t channel_config_idx; + int nb_elements; + int samplerate; + int sbr_ratio; + MPEG4AudioConfig *m4ac = &oc->m4ac; + AACUSACConfig *usac = &oc->usac; + int elem_id[3 /* SCE, CPE, LFE */]; + + uint8_t layout_map[MAX_ELEM_ID*4][3]; + + freq_idx = get_bits(gb, 5); /* usacSamplingFrequencyIndex */ + if (freq_idx == 0x1f) { + samplerate = get_bits(gb, 24); /* usacSamplingFrequency */ + + /* Try to match up an index for the custom sample rate. + * TODO: not sure if correct */ + for (idx = 0; idx < FF_ARRAY_ELEMS(aac_usac_samplerate); idx++) { + if (aac_usac_samplerate[idx] >= samplerate) + break; + } + idx = FFMIN(idx, FF_ARRAY_ELEMS(aac_usac_samplerate) - 1); + usac->rate_idx = idx; + } else { + samplerate = aac_usac_samplerate[freq_idx]; + usac->rate_idx = freq_idx; + } + + m4ac->sample_rate = avctx->sample_rate = samplerate; + + usac->core_sbr_frame_len_idx = get_bits(gb, 3); /* coreSbrFrameLengthIndex */ + m4ac->frame_length_short = usac->core_sbr_frame_len_idx == 0 || + usac->core_sbr_frame_len_idx == 2; + + usac->core_frame_len = (usac->core_sbr_frame_len_idx == 0 || + usac->core_sbr_frame_len_idx == 2) ? 768 : 1024; + + sbr_ratio = usac->core_sbr_frame_len_idx == 2 ? 2 : + usac->core_sbr_frame_len_idx == 3 ? 3 : + usac->core_sbr_frame_len_idx == 4 ? 1 : + 0; + + channel_config_idx = get_bits(gb, 5); /* channelConfigurationIndex */ + if (!channel_config_idx) { + /* UsacChannelConfig() */ + uint8_t channel_pos[64]; + uint8_t nb_channels = get_escaped_value(gb, 5, 8, 16); /* numOutChannels */ + if (nb_channels >= 64) + return AVERROR(EINVAL); + + av_channel_layout_uninit(&ac->oc[1].ch_layout); + for (int i = 0; i < nb_channels; i++) + channel_pos[i] = get_bits(gb, 5); /* bsOutputChannelPos */ + + ac->oc[1].ch_layout.order = AV_CHANNEL_ORDER_NATIVE; + ac->oc[1].ch_layout.nb_channels = nb_channels; + ac->oc[1].ch_layout.u.mask = 0; + + for (int i = 0; i < nb_channels; i++) + ac->oc[1].ch_layout.u.mask |= 1 << usac_ch_pos_to_av[channel_pos[i]]; + + av_channel_layout_copy(&avctx->ch_layout, &ac->oc[1].ch_layout); + } else { + if ((ret = ff_aac_set_default_channel_config(ac, avctx, layout_map, + &nb_elements, channel_config_idx))) + return ret; + } + + /* UsacDecoderConfig */ + elem_id[0] = elem_id[1] = elem_id[2] = 0; + usac->nb_elems = get_escaped_value(gb, 4, 8, 16) + 1; + + for (int i = 0; i < usac->nb_elems; i++) { + AACUsacElemConfig *e = &usac->elems[i]; + memset(e, 0, sizeof(*e)); + + e->type = get_bits(gb, 2); /* usacElementType */ + av_log(ac->avctx, AV_LOG_INFO, "Element present: idx %i, type %i\n", + i, e->type); + + switch (e->type) { + case ID_USAC_SCE: /* SCE */ + /* UsacCoreConfig */ + decode_usac_element_core(e, gb, sbr_ratio); + if (e->sbr.ratio > 0) + decode_usac_sbr_data(e, gb); + layout_map[i][0] = TYPE_SCE; + layout_map[i][1] = i; + layout_map[i][2] = AAC_CHANNEL_FRONT; + elem_id[0]++; + + break; + case ID_USAC_CPE: /* UsacChannelPairElementConf */ + /* UsacCoreConfig */ + decode_usac_element_core(e, gb, sbr_ratio); + decode_usac_element_pair(e, gb); + layout_map[i][0] = TYPE_CPE; + layout_map[i][1] = i; + layout_map[i][2] = AAC_CHANNEL_FRONT; + elem_id[1]++; + + break; + case ID_USAC_LFE: /* LFE */ + elem_id[2]++; + avpriv_report_missing_feature(ac->avctx, "AAC USAC LFE"); + return AVERROR_PATCHWELCOME; + case ID_USAC_EXT: /* EXT */ + ret = decode_usac_extension(ac, e, gb); + if (ret < 0) + return ret; + break; + }; + } + + ret = ff_aac_output_configure(ac, layout_map, elem_id[0] + elem_id[1] + elem_id[2], OC_GLOBAL_HDR, 0); + if (ret < 0) { + av_log(avctx, AV_LOG_ERROR, "Unable to parse channel config!\n"); + return ret; + } + + if (get_bits1(gb)) { /* usacConfigExtensionPresent */ + int invalid; + int nb_extensions = get_escaped_value(gb, 2, 4, 8) + 1; /* numConfigExtensions */ + for (int i = 0; i < nb_extensions; i++) { + int type = get_escaped_value(gb, 4, 8, 16); + int len = get_escaped_value(gb, 4, 8, 16); + switch (type) { + case ID_CONFIG_EXT_LOUDNESS_INFO: + ret = decode_loudness_set(ac, usac, gb); + if (ret < 0) + return ret; + break; + case ID_CONFIG_EXT_STREAM_ID: + usac->stream_identifier = get_bits(gb, 16); + break; + case ID_CONFIG_EXT_FILL: /* fallthrough */ + invalid = 0; + while (len--) { + if (get_bits(gb, 8) != 0xA5) + invalid++; + } + if (invalid) + av_log(avctx, AV_LOG_WARNING, "Invalid fill bytes: %i\n", + invalid); + break; + default: + while (len--) + skip_bits(gb, 8); + break; + } + } + } + + ret = ff_aac_usac_reset_state(ac, oc); + if (ret < 0) + return ret; + + return 0; +} + +int ec_debug = 0; + +/** + * Decode and dequantize arithmetically coded, uniformly quantized value + * + * @param coef array of dequantized, scaled spectral data + * @param sf array of scalefactors or intensity stereo positions + * + * @return Returns error status. 0 - OK, !0 - error + */ +static int decode_spectrum_and_dequant_ac(AACDecContext *s, float coef[1024], + GetBitContext *gb, const float sf[120], + enum BandType band_type[120], + AACArithState *state, int reset, + uint16_t len, uint16_t N) +{ + AACArith ac; + int i, a, b; + uint32_t c; + + int gb_count; + GetBitContext gb2; + + ff_aac_ac_init(&ac, gb); + c = ff_aac_ac_map_process(state, reset, N); + + /* Backup reader for rolling back by 14 bits at the end */ + gb2 = (GetBitContext)*gb; + gb_count = get_bits_count(&gb2); + + for (i = 0; i < len/2; i++) { + /* MSB */ + int lvl, esc_nb, m; + c = ff_aac_ac_get_context(state, c, i, N); + for (lvl=esc_nb=0;;) { + uint32_t pki = ff_aac_ac_get_pk(c + (esc_nb << 17)); + m = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_msb_cdfs[pki], + FF_ARRAY_ELEMS(ff_aac_ac_msb_cdfs[pki])); + if (m < FF_AAC_AC_ESCAPE) + break; + lvl++; + + /* Cargo-culted value. */ + if (lvl > 23) + return AVERROR(EINVAL); + + if ((esc_nb = lvl) > 7) + esc_nb = 7; + } + + b = m >> 2; + a = m - (b << 2); + + /* ARITH_STOP detection */ + if (!m) { + if (esc_nb) + break; + a = b = 0; + } + + /* LSB */ + for (int l = lvl; l > 0; l--) { + int lsbidx = !a ? 1 : (!b ? 0 : 2); + uint8_t r = ff_aac_ac_decode(&ac, &gb2, ff_aac_ac_lsb_cdfs[lsbidx], + FF_ARRAY_ELEMS(ff_aac_ac_lsb_cdfs[lsbidx])); + a = (a << 1) | (r & 1); + b = (b << 1) | ((r >> 1) & 1); + } + + /* Dequantize coeffs here */ + coef[2*i + 0] = a * cbrt(a); + coef[2*i + 1] = b * cbrt(b); + ff_aac_ac_update_context(state, i, a, b); + } + + if (len > 1) { + /* "Rewind" bitstream back by 14 bits */ + int gb_count2 = get_bits_count(&gb2); + skip_bits(gb, gb_count2 - gb_count - 14); + } else { + *gb = gb2; + } + + ff_aac_ac_finish(state, i, N); + + for (; i < N/2; i++) { + coef[2*i + 0] = 0; + coef[2*i + 1] = 0; + } + + /* Signs */ + for (i = 0; i < len; i++) { + if (coef[i]) { + if (!get_bits1(gb)) /* s */ + coef[i] *= -1; + } + } + + return 0; +} + +static int decode_usac_stereo_cplx(AACDecContext *ac, AACUsacStereo *us, + ChannelElement *cpe, GetBitContext *gb, + int num_window_groups, int indep_flag) +{ + int complex_coef; + + if (!get_bits1(gb)) { /* cplx_pred_all */ + for (int g = 0; g < num_window_groups; g++) { + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) { + us->cplx.pred_used[g][sfb] = get_bits1(gb); + if ((sfb + 1) < cpe->max_sfb_ste) + us->cplx.pred_used[g][sfb + 1] = us->cplx.pred_used[g][sfb]; + } + } + } else { + for (int g = 0; g < num_window_groups; g++) { + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { + us->cplx.pred_used[g][sfb] = 1; + } + } + } + + us->cplx.pred_dir = get_bits1(gb); + complex_coef = get_bits1(gb); + + us->cplx.use_prev_frame = 0; + if (complex_coef && !indep_flag) + us->cplx.use_prev_frame = get_bits1(gb); + + us->cplx.delta_code_time = 0; + if (!indep_flag) + us->cplx.delta_code_time = get_bits1(gb); + + memset(us->cplx.pred, 0, sizeof(us->cplx.pred)); + for (int g = 0; g < num_window_groups; g++) { + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb += SFB_PER_PRED_BAND) { + if (!us->cplx.pred_used[g][sfb]) + continue; + + us->cplx.pred[g][sfb].re = get_vlc2(gb, ff_vlc_scalefactors, 7, 3); + if (complex_coef) + us->cplx.pred[g][sfb].im = get_vlc2(gb, ff_vlc_scalefactors, 7, 3); + } + } + + return 0; +} + +static int setup_sce(SingleChannelElement *sce, AACUSACConfig *usac) +{ + AACUsacElemData *ue = &sce->ue; + IndividualChannelStream *ics = &sce->ics; + + /* Setup window parameters */ + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + if (usac->core_frame_len == 768) { + ics->swb_offset = ff_swb_offset_96[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_96[usac->rate_idx]; + } else { + ics->swb_offset = ff_swb_offset_128[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_128[usac->rate_idx]; + } + ics->tns_max_bands = ff_tns_max_bands_128[usac->rate_idx]; + + /* Setup scalefactor grouping. 7 bit mask. */ + ics->num_window_groups = 0; + for (int j = 0; j < 7; j++) { + ics->group_len[j] = 1; + if (ue->scale_factor_grouping & (1 << (6 - j))) + ics->group_len[ics->num_window_groups] += 1; + else + ics->num_window_groups++; + } + + ics->group_len[7] = 1; + ics->num_window_groups++; + ics->num_windows = 8; + } else { + if (usac->core_frame_len == 768) { + ics->swb_offset = ff_swb_offset_768[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_768[usac->rate_idx]; + } else { + ics->swb_offset = ff_swb_offset_1024[usac->rate_idx]; + ics->num_swb = ff_aac_num_swb_1024[usac->rate_idx]; + } + ics->tns_max_bands = ff_tns_max_bands_1024[usac->rate_idx]; + + ics->group_len[0] = 1; + ics->num_window_groups = 1; + ics->num_windows = 1; + } + + /* Just some defaults for the band types */ + for (int i = 0; i < FF_ARRAY_ELEMS(sce->band_type); i++) + sce->band_type[i] = ESC_BT; + + return 0; +} + +static int decode_usac_stereo_info(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemConfig *ec, ChannelElement *cpe, + GetBitContext *gb, int indep_flag) +{ + int ret, tns_active; + + AACUsacStereo *us = &cpe->us; + SingleChannelElement *sce1 = &cpe->ch[0]; + SingleChannelElement *sce2 = &cpe->ch[1]; + IndividualChannelStream *ics1 = &sce1->ics; + IndividualChannelStream *ics2 = &sce2->ics; + AACUsacElemData *ue1 = &sce1->ue; + AACUsacElemData *ue2 = &sce2->ue; + + us->common_window = 0; + us->common_tw = 0; + + if (!(!ue1->core_mode && !ue2->core_mode)) + return 0; + + tns_active = get_bits1(gb); + us->common_window = get_bits1(gb); + + if (us->common_window) { + /* ics_info() */ + ics1->window_sequence[1] = ics1->window_sequence[0]; + ics2->window_sequence[1] = ics2->window_sequence[0]; + ics1->window_sequence[0] = ics2->window_sequence[0] = get_bits(gb, 2); + + ics1->use_kb_window[1] = ics1->use_kb_window[0]; + ics2->use_kb_window[1] = ics2->use_kb_window[0]; + ics1->use_kb_window[0] = ics2->use_kb_window[0] = get_bits1(gb); + + if (ics1->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ics1->max_sfb = ics2->max_sfb = get_bits(gb, 4); + ue1->scale_factor_grouping = ue2->scale_factor_grouping = get_bits(gb, 7); + } else { + ics1->max_sfb = ics2->max_sfb = get_bits(gb, 6); + } + + if (!get_bits1(gb)) { /* common_max_sfb */ + if (ics2->window_sequence[0] == EIGHT_SHORT_SEQUENCE) + ics2->max_sfb = get_bits(gb, 4); + else + ics2->max_sfb = get_bits(gb, 6); + } + + ret = setup_sce(sce1, usac); + if (ret < 0) + return ret; + + ret = setup_sce(sce2, usac); + if (ret < 0) + return ret; + + cpe->max_sfb_ste = FFMAX(ics1->max_sfb, ics2->max_sfb); + + us->ms_mask_mode = get_bits(gb, 2); /* ms_mask_present */ + memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask)); + if (us->ms_mask_mode == 1) { + for (int g = 0; g < ics1->num_window_groups; g++) + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) + cpe->ms_mask[g*cpe->max_sfb_ste + sfb] = get_bits1(gb); + } else if (us->ms_mask_mode == 2) { + memset(cpe->ms_mask, 0xFF, sizeof(cpe->ms_mask)); + } else if ((us->ms_mask_mode == 3) && !ec->stereo_config_index) { + ret = decode_usac_stereo_cplx(ac, us, cpe, gb, + ics1->num_window_groups, indep_flag); + if (ret < 0) + return ret; + } + } + + if (ec->tw_mdct) { + us->common_tw = get_bits1(gb); + avpriv_report_missing_feature(ac->avctx, + "AAC USAC timewarping"); + return AVERROR_PATCHWELCOME; + } + + sce1->tns.present = sce2->tns.present = 0; + if (tns_active) { + av_unused int tns_on_lr; + int common_tns = 0; + if (us->common_window) + common_tns = get_bits1(gb); + + tns_on_lr = get_bits1(gb); + if (common_tns) { + ret = ff_aac_decode_tns(ac, &sce1->tns, gb, ics1); + if (ret < 0) + return ret; + memcpy(&sce2->tns, &sce1->tns, sizeof(sce1->tns)); + sce2->tns.present = 0; + sce1->tns.present = 0; + } else { + if (get_bits1(gb)) { + sce2->tns.present = 1; + sce1->tns.present = 1; + } else { + sce2->tns.present = get_bits1(gb); + sce1->tns.present = !sce2->tns.present; + } + } + } + + return 0; +} + +static int decode_usac_scale_factors(AACDecContext *ac, + SingleChannelElement *sce, + GetBitContext *gb, uint8_t global_gain) +{ + IndividualChannelStream *ics = &sce->ics; + + /* Decode all scalefactors. */ + int offset_sf = global_gain; + for (int g = 0; g < ics->num_window_groups; g++) { + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + /* First coefficient is just the global gain */ + if (!g && !sfb) { + /* The cannonical representation of quantized scalefactors + * in the spec is with 100 subtracted. */ + sce->sfo[0] = offset_sf - 100; + continue; + } + + offset_sf += get_vlc2(gb, ff_vlc_scalefactors, 7, 3) - SCALE_DIFF_ZERO; + if (offset_sf > 255U) { + av_log(ac->avctx, AV_LOG_ERROR, + "Scalefactor (%d) out of range.\n", offset_sf); + return AVERROR_INVALIDDATA; + } + + sce->sfo[g*ics->max_sfb + sfb] = offset_sf - 100; + } + } + + return 0; +} + +/* 7.2.4 Generation of random signs for spectral noise filling + * This function is exactly defined, though we've helped the definition + * along with being slightly faster. */ +static inline float noise_random_sign(unsigned int *seed) +{ + unsigned int new_seed = ((*seed) * 69069) + 5; + if (((new_seed) & 0x10000) > 0) + return -1.f; + + *seed = new_seed; + return +1.f; +} + +static void apply_noise_fill(AACDecContext *ac, SingleChannelElement *sce, + AACUsacElemData *ue) +{ + float *coef; + IndividualChannelStream *ics = &sce->ics; + + float noise_val = pow(2, (ue->noise.level - 14)/3); + int noise_offset = ue->noise.offset - 16; + int band_off; + + band_off = ff_usac_noise_fill_start_offset[ac->oc[1].m4ac.frame_length_short] + [ics->num_windows == 8]; + + coef = sce->coeffs; + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + float *cb = coef + ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb]; + int band_quantized_to_zero = 1; + + if (ics->swb_offset[sfb] < band_off) + continue; + + for (int group = 0; group < (unsigned)g_len; group++, cb += 128) { + for (int z = 0; z < cb_len; z++) { + if (cb[z] == 0) + cb[z] = noise_random_sign(&sce->ue.noise.seed) * noise_val; + else + band_quantized_to_zero = 0; + } + } + + if (band_quantized_to_zero) + sce->sf[g*ics->max_sfb + sfb] += noise_offset; + } + coef += g_len << 7; + } +} + +static void spectrum_scale(AACDecContext *ac, SingleChannelElement *sce, + AACUsacElemData *ue) +{ + IndividualChannelStream *ics = &sce->ics; + float *coef; + + /* Synthesise noise */ + if (ue->noise.level) + apply_noise_fill(ac, sce, ue); + + /* Apply scalefactors */ + coef = sce->coeffs; + for (int g = 0; g < ics->num_window_groups; g++) { + unsigned g_len = ics->group_len[g]; + + for (int sfb = 0; sfb < ics->max_sfb; sfb++) { + float *cb = coef + ics->swb_offset[sfb]; + int cb_len = ics->swb_offset[sfb + 1] - ics->swb_offset[sfb]; + float sf = sce->sf[g*ics->max_sfb + sfb]; + + for (int group = 0; group < (unsigned)g_len; group++, cb += 128) + ac->fdsp->vector_fmul_scalar(cb, cb, sf, cb_len); + } + coef += g_len << 7; + } +} + +static void spectrum_decode(AACDecContext *ac, ChannelElement *cpe, int nb_channels) +{ + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &cpe->ch[ch]; + AACUsacElemData *ue = &sce->ue; + + spectrum_scale(ac, sce, ue); + } + + if (nb_channels > 1) + ac->dsp.apply_mid_side_stereo(ac, cpe); + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &cpe->ch[ch]; + + /* Apply TNS */ + if (sce->tns.present) + ac->dsp.apply_tns(sce->coeffs, &sce->tns, &sce->ics, 1); + + ac->oc[1].m4ac.frame_length_short ? ac->dsp.imdct_and_windowing_768(ac, sce) : + ac->dsp.imdct_and_windowing(ac, sce); + } + +} + +static int decode_usac_core_coder(AACDecContext *ac, AACUSACConfig *usac, + AACUsacElemConfig *ec, ChannelElement *che, + GetBitContext *gb, int indep_flag, int nb_channels) +{ + int ret; + int arith_reset_flag; + AACUsacStereo *us = &che->us; + + /* Local symbols */ + uint8_t global_gain; + + us->common_window = 0; + che->ch[0].tns.present = che->ch[1].tns.present = 0; + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &che->ch[ch]; + AACUsacElemData *ue = &sce->ue; + + ue->core_mode = get_bits1(gb); + } + + if (nb_channels == 2) { + ret = decode_usac_stereo_info(ac, usac, ec, che, gb, indep_flag); + if (ret) + return ret; + } + + for (int ch = 0; ch < nb_channels; ch++) { + SingleChannelElement *sce = &che->ch[ch]; + IndividualChannelStream *ics = &sce->ics; + AACUsacElemData *ue = &sce->ue; + + if (ue->core_mode) { /* lpd_channel_stream */ + ret = ff_aac_ldp_parse_channel_stream(ac, usac, ue, gb); + if (ret < 0) + return ret; + } + + if ((nb_channels == 1) || + (che->ch[0].ue.core_mode != che->ch[1].ue.core_mode)) + sce->tns.present = get_bits1(gb); + + /* fd_channel_stream */ + global_gain = get_bits(gb, 8); + + ue->noise.level = 0; + if (ec->noise_fill) { + ue->noise.level = get_bits(gb, 3); + ue->noise.offset = get_bits(gb, 5); + } + + if (!us->common_window) { + /* ics_info() */ + ics->window_sequence[1] = ics->window_sequence[0]; + ics->window_sequence[0] = get_bits(gb, 2); + ics->use_kb_window[1] = ics->use_kb_window[0]; + ics->use_kb_window[0] = get_bits1(gb); + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { + ics->max_sfb = get_bits(gb, 4); + ue->scale_factor_grouping = get_bits(gb, 7); + } else { + ics->max_sfb = get_bits(gb, 6); + } + + ret = setup_sce(sce, usac); + if (ret < 0) + return ret; + } + + if (ec->tw_mdct && !us->common_tw) { + /* tw_data() */ + if (get_bits1(gb)) { /* tw_data_present */ + /* Time warping is not supported in baseline profile streams. */ + avpriv_report_missing_feature(ac->avctx, + "AAC USAC timewarping"); + return AVERROR_PATCHWELCOME; + } + } + + ret = decode_usac_scale_factors(ac, sce, gb, global_gain); + if (ret < 0) + return ret; + + ac->dsp.dequant_scalefactors(sce); + + if (sce->tns.present) { + ret = ff_aac_decode_tns(ac, &sce->tns, gb, ics); + if (ret < 0) + return ret; + } + + /* ac_spectral_data */ + arith_reset_flag = indep_flag; + if (!arith_reset_flag) + arith_reset_flag = get_bits1(gb); + + /* Decode coeffs */ + memset(&sce->coeffs[0], 0, 1024*sizeof(float)); + for (int win = 0; win < ics->num_windows; win++) { + int lg = ics->swb_offset[ics->max_sfb]; + int N; + if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) + N = usac->core_frame_len / 8; + else + N = usac->core_frame_len; + + ret = decode_spectrum_and_dequant_ac(ac, sce->coeffs + win*128, gb, + sce->sf, NULL, &ue->ac, + arith_reset_flag && (win == 0), + lg, N); + if (ret < 0) + return ret; + } + + if (get_bits1(gb)) { /* fac_data_present */ + const uint16_t len_8 = usac->core_frame_len / 8; + const uint16_t len_16 = usac->core_frame_len / 16; + const uint16_t fac_len = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? len_8 : len_16; + ret = ff_aac_parse_fac_data(ue, gb, 1, fac_len); + if (ret < 0) + return ret; + } + } + + spectrum_decode(ac, che, nb_channels); + + return 0; +} + +static int parse_audio_preroll(AACDecContext *ac, uint8_t *data, uint32_t len) +{ + GetBitContext gb, gbc; + OutputConfiguration *oc = &ac->oc[1]; + MPEG4AudioConfig *m4ac = &oc->m4ac; + + int ret, config_len; + av_unused int crossfade; + int num_preroll_frames; + + ret = init_get_bits8(&gb, data, len); + if (ret < 0) + return ret; + + config_len = get_escaped_value(&gb, 4, 4, 8); + +#if 0 + gbc = gb; + ret = ff_aac_usac_config_decode(ac, ac->avctx, &gbc, oc, m4ac->chan_config); + if (ret < 0) + return ret; +#endif + + skip_bits_long(&gb, config_len*8); + + crossfade = get_bits1(&gb); /* applyCrossfade */ + get_bits1(&gb); /* reserved */ + num_preroll_frames = get_escaped_value(&gb, 2, 4, 0); /* numPreRollFrames */ + + for (int i = 0; i < num_preroll_frames; i++) { + int got_frame_ptr = 0; + int au_len = get_escaped_value(&gb, 16, 16, 0); + + /* Overwrite the input, but its fine. Byte alignment is not guaranteed. */ + for (int j = 0; j < au_len; j++) + data[j] = get_bits(&gb, 8); + + ret = init_get_bits8(&gbc, data, au_len); + if (ret < 0) + return ret; + + ret = ff_aac_usac_decode_frame(ac->avctx, ac, &gbc, &got_frame_ptr); + if (ret < 0) + return ret; + } + + return 0; +} + +static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e, + GetBitContext *gb) +{ + if (get_bits1(gb)) { /* usacExtElementPresent */ + uint32_t len; + if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */ + len = e->ext.default_len; + } else { + len = get_bits(gb, 8); /* usacExtElementPayloadLength */ + if (len == 255) + len += get_bits(gb, 16) - 2; + } + + if (len) { + uint8_t *tmp; + uint8_t pl_frag_start = 1; + uint8_t pl_frag_end = 1; + if (e->ext.payload_frag) { + pl_frag_start = get_bits1(gb); /* usacExtElementStart */ + pl_frag_end = get_bits1(gb); /* usacExtElementStop */ + } + + if (pl_frag_start) + e->ext.pl_data_offset = 0; + + tmp = av_realloc(e->ext.pl_data, e->ext.pl_data_offset + len); + if (!tmp) { + free(e->ext.pl_data); + return AVERROR(ENOMEM); + } + e->ext.pl_data = tmp; + + for (int i = 0; i < len; i++) + e->ext.pl_data[e->ext.pl_data_offset + i] = get_bits(gb, 8); + + if (pl_frag_end) { + int ret; + e->ext.pl_data_offset = 0; + switch (e->ext.type) { + case ID_EXT_ELE_FILL: + av_freep(&e->ext.pl_data); + break; + case ID_EXT_ELE_AUDIOPREROLL: + ret = parse_audio_preroll(ac, e->ext.pl_data, + e->ext.pl_data_offset); + if (ret < 0) { + av_freep(&e->ext.pl_data); + return ret; + } + break; + default: + av_freep(&e->ext.pl_data); + /* This should never happen */ + av_assert0(0); + } + } + } + } + + return 0; +} + +int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr) +{ + int ret, nb_ch_el, is_dmono = 0; + int indep_flag, samples = 0; + int audio_found = 0, sce_count = 0; + AVFrame *frame = ac->frame; + + ff_aac_output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags, + ac->oc[1].status, 0); + + indep_flag = get_bits1(gb); + + nb_ch_el = 0; + for (int i = 0; i < ac->oc[1].usac.nb_elems; i++) { + AACUsacElemConfig *e = &ac->oc[1].usac.elems[i]; + ChannelElement *che; + + switch (e->type) { + case ID_USAC_SCE: + che = ff_aac_get_che(ac, TYPE_SCE, nb_ch_el++); + if (!che) { + av_log(ac->avctx, AV_LOG_ERROR, + "channel element %d.%d is not allocated\n", + TYPE_SCE, nb_ch_el - 1); + return AVERROR_INVALIDDATA; + } + + ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb, + indep_flag, 1); + if (ret < 0) + return ret; + + sce_count++; + audio_found = 1; + che->present = 1; + samples = ac->oc[1].m4ac.frame_length_short ? 768 : 1024; + break; + case ID_USAC_CPE: + che = ff_aac_get_che(ac, TYPE_CPE, nb_ch_el++); + if (!che) { + av_log(ac->avctx, AV_LOG_ERROR, + "channel element %d.%d is not allocated\n", + TYPE_SCE, nb_ch_el - 1); + return AVERROR_INVALIDDATA; + } + + ret = decode_usac_core_coder(ac, &ac->oc[1].usac, e, che, gb, + indep_flag, 2); + if (ret < 0) + return ret; + + audio_found = 1; + che->present = 1; + samples = ac->oc[1].m4ac.frame_length_short ? 768 : 1024; + break; + case ID_USAC_LFE: + avpriv_report_missing_feature(ac->avctx, + "AAC USAC LFE"); + return AVERROR_PATCHWELCOME; + break; + case ID_USAC_EXT: + ret = parse_ext_ele(ac, e, gb); + if (ret < 0) + return ret; + break; + } + } + + if (ac->oc[1].status && audio_found) { + avctx->sample_rate = ac->oc[1].m4ac.sample_rate; + avctx->frame_size = samples; + ac->oc[1].status = OC_LOCKED; + } + + if (!frame->data[0] && samples) { + av_log(avctx, AV_LOG_ERROR, "no frame data found\n"); + return AVERROR_INVALIDDATA; + } + + if (samples) { + frame->nb_samples = samples; + frame->sample_rate = avctx->sample_rate; + frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0; + *got_frame_ptr = 1; + } else { + av_frame_unref(ac->frame); + frame->flags = indep_flag ? AV_FRAME_FLAG_KEY : 0x0; + *got_frame_ptr = 0; + } + + /* for dual-mono audio (SCE + SCE) */ + is_dmono = ac->dmono_mode && sce_count == 2 && + !av_channel_layout_compare(&ac->oc[1].ch_layout, + &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO); + if (is_dmono) { + if (ac->dmono_mode == 1) + frame->data[1] = frame->data[0]; + else if (ac->dmono_mode == 2) + frame->data[0] = frame->data[1]; + } + + return 0; +} diff --git a/libavcodec/aac/aacdec_usac.h b/libavcodec/aac/aacdec_usac.h new file mode 100644 index 0000000000..d3e0ca4a7b --- /dev/null +++ b/libavcodec/aac/aacdec_usac.h @@ -0,0 +1,47 @@ +/* + * Copyright (c) 2024 Lynne <dev@lynne.ee> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavcodec/opusdsp.h" + +#ifndef AVCODEC_AAC_AACDEC_USAC_H +#define AVCODEC_AAC_AACDEC_USAC_H + +#include "aacdec.h" + +#include "libavcodec/get_bits.h" + +static inline uint8_t ff_aac_get_vlclbf(GetBitContext *gb) +{ + uint8_t ret = 0; + while (get_bits1(gb) && ret <= 36) + ret++; + return ret; +} + +int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx, + GetBitContext *gb, OutputConfiguration *oc, + int channel_config); + +int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc); + +int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac, + GetBitContext *gb, int *got_frame_ptr); + +#endif /* AVCODEC_AAC_AACDEC_USAC_H */ diff --git a/libavcodec/aactab.c b/libavcodec/aactab.c index 18afa69bad..6720a3373b 100644 --- a/libavcodec/aactab.c +++ b/libavcodec/aactab.c @@ -1998,6 +1998,11 @@ const uint8_t ff_tns_max_bands_128[] = { }; // @} +const uint8_t ff_usac_noise_fill_start_offset[2][2] = { + { 160, 20 }, + { 120, 15 }, +}; + const DECLARE_ALIGNED(32, float, ff_aac_eld_window_512)[1920] = { 0.00338834, 0.00567745, 0.00847677, 0.01172641, 0.01532555, 0.01917664, 0.02318809, 0.02729259, diff --git a/libavcodec/aactab.h b/libavcodec/aactab.h index 481fc57d93..46947bcd61 100644 --- a/libavcodec/aactab.h +++ b/libavcodec/aactab.h @@ -115,4 +115,7 @@ extern const uint8_t ff_tns_max_bands_512 [13]; extern const uint8_t ff_tns_max_bands_480 [13]; extern const uint8_t ff_tns_max_bands_128 [13]; +/* [x][y], x == 1 -> frame len is 768 frames, y == 1 -> is eight_short */ +extern const uint8_t ff_usac_noise_fill_start_offset[2][2]; + #endif /* AVCODEC_AACTAB_H */ -- 2.43.0.381.gb435a96ce8 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* Re: [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) Lynne via ffmpeg-devel @ 2024-05-16 10:26 ` Andreas Rheinhardt 2024-05-16 15:00 ` Lynne via ffmpeg-devel 0 siblings, 1 reply; 13+ messages in thread From: Andreas Rheinhardt @ 2024-05-16 10:26 UTC (permalink / raw) To: ffmpeg-devel Lynne via ffmpeg-devel: > This commit adds a decoder for the frequency-domain part of USAC. > > What works: > - Mono > - Stereo (no prediction) > - Stereo (mid/side coding) > > What doesn't: > - Preroll decoding (every single decoder seems faulty or weird?) > - Complex stereo prediction > > Known issues: > - Spec incompliance (noise synthesis in particular) > - Lack of robustness > --- > diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h > index 20545a24d4..3e6592cf0e 100644 > --- a/libavcodec/aac/aacdec.h > +++ b/libavcodec/aac/aacdec.h > @@ -42,6 +42,8 @@ > #include "libavcodec/avcodec.h" > #include "libavcodec/mpeg4audio.h" > > +#include "aacdec_ac.h" > + > typedef struct AACDecContext AACDecContext; > > /** > @@ -69,6 +71,32 @@ enum CouplingPoint { > AFTER_IMDCT = 3, > }; > > +enum AACUsacElem { > + ID_USAC_SCE = 0, > + ID_USAC_CPE = 1, > + ID_USAC_LFE = 2, > + ID_USAC_EXT = 3, > +}; > + > +enum ExtensionHeaderType { > + ID_CONFIG_EXT_FILL = 0, > + ID_CONFIG_EXT_LOUDNESS_INFO = 2, > + ID_CONFIG_EXT_STREAM_ID = 7, > +}; > + > +enum AACUsacExtension { > + ID_EXT_ELE_FILL, > + ID_EXT_ELE_MPEGS, > + ID_EXT_ELE_SAOC, > + ID_EXT_ELE_AUDIOPREROLL, > + ID_EXT_ELE_UNI_DRC, > +}; > + > +enum AACUSACLoudnessExt { > + UNIDRCLOUDEXT_TERM = 0x0, > + UNIDRCLOUDEXT_EQ = 0x1, > +}; > + > // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. > #define RENAME_FIXED(name) name ## _fixed > > @@ -93,6 +121,40 @@ typedef struct LongTermPrediction { > int8_t used[MAX_LTP_LONG_SFB]; > } LongTermPrediction; > > +/* Per channel core mode */ > +typedef struct AACUsacElemData { > + uint8_t core_mode; > + uint8_t scale_factor_grouping; > + > + /* Timewarping ratio */ > +#define NUM_TW_NODES 16 > + uint8_t tw_ratio[NUM_TW_NODES]; > + > + struct { > + uint8_t acelp_core_mode : 3; > + uint8_t lpd_mode : 5; > + > + uint8_t bpf_control_info : 1; > + uint8_t core_mode_last : 1; > + uint8_t fac_data_present : 1; > + > + int last_lpd_mode; > + } ldp; > + > + struct { > + unsigned int seed; > + uint8_t level : 3; > + uint8_t offset : 5; > + } noise; > + > + struct { > + uint8_t gain; > + uint32_t kv[8 /* (1024 / 16) / 8 */][8]; > + } fac; > + > + AACArithState ac; > +} AACUsacElemData; > + > /** > * Individual Channel Stream > */ > @@ -145,6 +207,7 @@ typedef struct ChannelCoupling { > */ > typedef struct SingleChannelElement { > IndividualChannelStream ics; > + AACUsacElemData ue; ///< USAC element data > TemporalNoiseShaping tns; > enum BandType band_type[128]; ///< band types > int sfo[128]; ///< scalefactor offsets > @@ -163,25 +226,141 @@ typedef struct SingleChannelElement { > }; > } SingleChannelElement; > > +typedef struct AACUsacStereo { > + uint8_t common_window; > + uint8_t common_tw; > + > + uint8_t ms_mask_mode; > + uint8_t config_idx; > + > + struct { > + uint8_t use_prev_frame; > + uint8_t pred_dir; > + uint8_t delta_code_time; > + uint8_t pred_used[8][64]; > + > + AVComplexFloat pred[8][64]; > + } cplx; > +} AACUsacStereo; > + > /** > * channel element - generic struct for SCE/CPE/CCE/LFE > */ > typedef struct ChannelElement { > int present; > // CPE specific > + uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values > uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band > // shared > SingleChannelElement ch[2]; > // CCE specific > ChannelCoupling coup; > + // USAC stereo coupling data > + AACUsacStereo us; > } ChannelElement; > > +typedef struct AACUSACLoudnessInfo { > + uint8_t drc_set_id : 6; > + uint8_t downmix_id : 7; > + struct { > + uint16_t lvl : 12; > + uint8_t present : 1; > + } sample_peak; > + > + struct { > + uint16_t lvl : 12; > + uint8_t measurement : 4; > + uint8_t reliability : 2; > + uint8_t present : 1; > + } true_peak; > + > + uint8_t nb_measurements : 4; > + struct { > + uint8_t method_def : 4; > + uint8_t method_val; > + uint8_t measurement : 4; > + uint8_t reliability : 2; > + } measurements[16]; > +} AACUSACLoudnessInfo; > + > +typedef struct AACUsacElemConfig { > + enum AACUsacElem type; > + > + uint8_t tw_mdct : 1; > + uint8_t noise_fill : 1; > + > + uint8_t stereo_config_index; > + > + struct { > + int ratio; > + > + uint8_t harmonic_sbr : 1; /* harmonicSBR */ > + uint8_t bs_intertes : 1; /* bs_interTes */ > + uint8_t bs_pvc : 1; /* bs_pvc */ > + > + struct { > + uint8_t start_freq; /* dflt_start_freq */ > + uint8_t stop_freq; /* dflt_stop_freq */ > + > + uint8_t freq_scale; /* dflt_freq_scale */ > + uint8_t alter_scale : 1; /* dflt_alter_scale */ > + uint8_t noise_scale; /* dflt_noise_scale */ > + > + uint8_t limiter_bands; /* dflt_limiter_bands */ > + uint8_t limiter_gains; /* dflt_limiter_gains */ > + uint8_t interpol_freq : 1; /* dflt_interpol_freq */ > + uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ > + } dflt; > + } sbr; > + > + struct { > + uint8_t freq_res; /* bsFreqRes */ > + uint8_t fixed_gain; /* bsFixedGainDMX */ > + uint8_t temp_shape_config; /* bsTempShapeConfig */ > + uint8_t decorr_config; /* bsDecorrConfig */ > + uint8_t high_rate_mode : 1; /* bsHighRateMode */ > + uint8_t phase_coding : 1; /* bsPhaseCoding */ > + > + uint8_t otts_bands_phase; /* bsOttBandsPhase */ > + uint8_t residual_coding; /* bsResidualCoding */ > + uint8_t residual_bands; /* bsResidualBands */ > + uint8_t pseudo_lr : 1; /* bsPseudoLr */ > + uint8_t env_quant_mode : 1; /* bsEnvQuantMode */ Is using bitfields really worth it given that they force to use masking for accesses? > + } mps; > + > + struct { > + enum AACUsacExtension type; > + uint8_t payload_frag; > + uint32_t default_len; > + uint32_t pl_data_offset; > + uint8_t *pl_data; > + } ext; > +} AACUsacElemConfig; > + > +typedef struct AACUSACConfig { > + uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ > + uint8_t rate_idx; > + uint16_t core_frame_len; > + uint16_t stream_identifier; > + > + AACUsacElemConfig elems[64]; > + int nb_elems; > + > + struct { > + uint8_t nb_album; > + AACUSACLoudnessInfo album_info[64]; > + uint8_t nb_info; > + AACUSACLoudnessInfo info[64]; > + } loudness; > +} AACUSACConfig; > + > typedef struct OutputConfiguration { > MPEG4AudioConfig m4ac; > uint8_t layout_map[MAX_ELEM_ID*4][3]; > int layout_map_tags; > AVChannelLayout ch_layout; > enum OCStatus status; > + AACUSACConfig usac; > } OutputConfiguration; > > /** > diff --git a/libavcodec/aac/aacdec_ac.c b/libavcodec/aac/aacdec_ac.c > new file mode 100644 > index 0000000000..326d716bd3 > --- /dev/null > +++ b/libavcodec/aac/aacdec_ac.c > @@ -0,0 +1,224 @@ > +/* > + * AAC definitions and structures > + * Copyright (c) 2024 Lynne > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#include "libavcodec/aactab.h" > +#include "aacdec_ac.h" > + > +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N) > +{ > + float ratio; > + if (reset) { > + memset(state->last, 0, sizeof(state->last)); > + state->last_len = N; > + memset(state->cur, 0, sizeof(state->cur)); > + state->cur[3] = 0; > + state->cur[2] = 0; > + state->cur[1] = 0; > + state->cur[0] = 1; > + state->last[0] = 0 << 12; > + state->state_pre = 0; > + return 0; > + } else if (state->last_len != N) { > + int i; > + uint8_t last[512 /* 2048 / 4 */]; > + memcpy(last, state->last, sizeof(last)); > + > + ratio = state->last_len / (float)N; > + for (i = 0; i < N/2; i++) { > + int k = (int)(i * ratio); > + state->last[i] = last[k]; > + } > + > + for (; i < FF_ARRAY_ELEMS(state->last); i++) > + state->last[i] = 0; > + > + state->last_len = N; > + } > + > + memset(state->cur, 0, sizeof(state->cur)); > + state->cur[3] = 0; > + state->cur[2] = 0; > + state->cur[1] = 0; > + state->cur[0] = 1; > + > + state->state_pre = state->last[0] << 12; > + return state->last[0] << 12; > +} > + > +extern int ec_debug; > + > +int trig = 0; > + > +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N) > +{ > + c = state->state_pre >> 8; > + c = c + (state->last[i + 1] << 8); > + c = (c << 4); > + c += state->cur[1]; > + > + state->state_pre = c; > + > + if (i > 3 && > + ((state->cur[3] + state->cur[2] + state->cur[1]) < 5)) > + return c + 0x10000; > + > + return c; > +} > + > +uint32_t ff_aac_ac_get_pk(uint32_t c) > +{ > + int i_min = -1; > + int i, j; > + int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1; > + while ((i_max - i_min) > 1) { > + i = i_min + ((i_max - i_min) / 2); > + j = ff_aac_ac_hash_m[i]; > + if (c < (j >> 8)) > + i_max = i; > + else if (c > (j >> 8)) > + i_min = i; > + else > + return (j & 0xFF); > + } > + return ff_aac_ac_lookup_m[i_max]; > +} > + > +void ff_aac_ac_update_context(AACArithState *state, int idx, > + uint16_t a, uint16_t b) > +{ > + state->cur[0] = a + b + 1; > + if (state->cur[0] > 0xF) > + state->cur[0] = 0xF; > + > + state->cur[3] = state->cur[2]; > + state->cur[2] = state->cur[1]; > + state->cur[1] = state->cur[0]; > + > + state->last[idx] = state->cur[0]; > +} > + > +/* Initialize AC */ > +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb) > +{ > + ac->low = 0; > + ac->high = UINT16_MAX; > + ac->val = get_bits(gb, 16); > +} > + > +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, > + const uint16_t *cdf, uint16_t cdf_len) > +{ > + int val = ac->val; > + int low = ac->low; > + int high = ac->high; > + > + int rng = high - low + 1; > + int c = ((((int)(val - low + 1)) << 14) - ((int)1)); > + > + /* Note: this could be done faster via heuristics, the total number of > + * configurations is low */ > + const uint16_t *p = cdf - 1; > + const uint16_t *q; > + > + switch (cdf_len) { > + case 2: > + if ((p[1] * rng) > c) > + p += 1; > + break; > + case 4: > + if ((p[2] * rng) > c) > + p += 2; > + if ((p[1] * rng) > c) > + p += 1; > + break; > + case 17: > + /* First check if the current probability is even met at all */ > + if ((p[1] * rng) <= c) > + break; > + p += 1; > + for (int i = 8; i >= 1; i >>= 1) > + if ((p[i] * rng) > c) > + p += i; > + break; > + case 27: > + const uint16_t *p_24 = p + 24; > + > + if ((p[16] * rng) > c) > + p += 16; > + if ((p[8] * rng) > c) > + p += 8; > + if (p != p_24) > + if ((p[4] * rng) > c) > + p += 4; > + if ((p[2] * rng) > c) > + p += 2; > + > + if (p != &p_24[2]) > + if ((p[1] * rng) > c) > + p += 1; > + break; > + default: > + /* This should never happen */ > + av_assert2(0); > + } > + > + int sym = (int)((ptrdiff_t)(p - cdf)) + 1; > + if (sym) > + high = low + ((rng * cdf[sym - 1]) >> 14) - 1; > + low += (rng * cdf[sym]) >> 14; > + > + /* This loop could be done faster */ > + while (1) { > + if (high < 32768) { > + ; > + } else if (low >= 32768) { > + val -= 32768; > + low -= 32768; > + high -= 32768; > + } else if (low >= 16384 && high < 49152) { > + val -= 16384; > + low -= 16384; > + high -= 16384; > + } else { > + break; > + } > + low += low; > + high += high + 1; > + val = (val << 1) | get_bits1(gb); > + }; > + > + ac->low = low; > + ac->high = high; > + ac->val = val; > + > + return sym; > +} > + > +void ff_aac_ac_finish(AACArithState *state, int offset, int N) > +{ > + int i; > + > + for (i = offset; i < N/2; i++) > + state->last[i] = 1; > + > + for (; i < FF_ARRAY_ELEMS(state->last); i++) > + state->last[i] = 0; > +} > diff --git a/libavcodec/aac/aacdec_ac.h b/libavcodec/aac/aacdec_ac.h > new file mode 100644 > index 0000000000..ef96bed770 > --- /dev/null > +++ b/libavcodec/aac/aacdec_ac.h > @@ -0,0 +1,54 @@ > +/* > + * AAC definitions and structures > + * Copyright (c) 2024 Lynne > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#ifndef AVCODEC_AACDEC_AC_H > +#define AVCODEC_AACDEC_AC_H > + > +#include "libavcodec/get_bits.h" > + > +typedef struct AACArithState { > + uint8_t last[512 /* 2048 / 4 */]; > + int last_len; > + uint8_t cur[4]; > + uint16_t state_pre; > +} AACArithState; > + > +typedef struct AACArith { > + uint16_t low; > + uint16_t high; > + uint16_t val; > +} AACArith; > + > +#define FF_AAC_AC_ESCAPE 16 > + > +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len); > +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len); > +uint32_t ff_aac_ac_get_pk(uint32_t c); > + > +void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b); > +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb); > + > +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, > + const uint16_t *cdf, uint16_t cdf_len); > + > +void ff_aac_ac_finish(AACArithState *state, int offset, int nb); > + > +#endif /* AVCODEC_AACDEC_AC_H */ > diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c > index 59a69d88f3..8d31af22f8 100644 > --- a/libavcodec/aac/aacdec_dsp_template.c > +++ b/libavcodec/aac/aacdec_dsp_template.c > @@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement > INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); > const uint16_t *offsets = ics->swb_offset; > for (int g = 0; g < ics->num_window_groups; g++) { > - for (int sfb = 0; sfb < ics->max_sfb; sfb++) { > - const int idx = g*ics->max_sfb + sfb; > + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { > + const int idx = g*cpe->max_sfb_ste + sfb; > if (cpe->ms_mask[idx] && > cpe->ch[0].band_type[idx] < NOISE_BT && > cpe->ch[1].band_type[idx] < NOISE_BT) { > diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h > index e40a2fe1a7..047c11e0fb 100644 > --- a/libavcodec/aac/aacdec_latm.h > +++ b/libavcodec/aac/aacdec_latm.h > @@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, > { > AACDecContext *ac = &latmctx->aac_ctx; > AVCodecContext *avctx = ac->avctx; > - MPEG4AudioConfig m4ac = { 0 }; > + OutputConfiguration oc = { 0 }; > + MPEG4AudioConfig *m4ac = &oc.m4ac; > GetBitContext gbc; > int config_start_bit = get_bits_count(gb); > int sync_extension = 0; > @@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, > if (get_bits_left(gb) <= 0) > return AVERROR_INVALIDDATA; > > - bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, > + bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc, > &gbc, config_start_bit, > sync_extension); > > @@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, > asclen = bits_consumed; > > if (!latmctx->initialized || > - ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || > - ac->oc[1].m4ac.chan_config != m4ac.chan_config) { > + ac->oc[1].m4ac.sample_rate != m4ac->sample_rate || > + ac->oc[1].m4ac.chan_config != m4ac->chan_config) { > > if (latmctx->initialized) { > - av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); > + av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", > + m4ac->sample_rate, m4ac->chan_config); > } else { > av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); > } > @@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, > } else { > push_output_configuration(&latmctx->aac_ctx); > if ((err = decode_audio_specific_config( > - &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, > + &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1], > avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { > pop_output_configuration(&latmctx->aac_ctx); > return err; > diff --git a/libavcodec/aac/aacdec_lpd.c b/libavcodec/aac/aacdec_lpd.c > new file mode 100644 > index 0000000000..be39e2c175 > --- /dev/null > +++ b/libavcodec/aac/aacdec_lpd.c > @@ -0,0 +1,192 @@ > +/* > + * Copyright (c) 2024 Lynne <dev@lynne.ee> > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#include "aacdec_lpd.h" > +#include "aacdec_usac.h" > + > +const uint8_t ff_aac_lpd_mode_tab[32][4] = { > + { 0, 0, 0, 0 }, > + { 1, 0, 0, 0 }, > + { 0, 1, 0, 0 }, > + { 1, 1, 0, 0 }, > + { 0, 0, 1, 0 }, > + { 1, 0, 1, 0 }, > + { 0, 1, 1, 0 }, > + { 1, 1, 1, 0 }, > + { 0, 0, 0, 1 }, > + { 1, 0, 0, 1 }, > + { 0, 1, 0, 1 }, > + { 1, 1, 0, 1 }, > + { 0, 0, 1, 1 }, > + { 1, 0, 1, 1 }, > + { 0, 1, 1, 1 }, > + { 1, 1, 1, 1 }, > + { 2, 2, 0, 0 }, > + { 2, 2, 1, 0 }, > + { 2, 2, 0, 1 }, > + { 2, 2, 1, 1 }, > + { 0, 0, 2, 2 }, > + { 1, 0, 2, 2 }, > + { 0, 1, 2, 2 }, > + { 1, 1, 2, 2 }, > + { 2, 2, 2, 2 }, > + { 3, 3, 3, 3 }, > + /* Larger values are reserved, but permit them for resilience */ > + { 0, 0, 0, 0 }, > + { 0, 0, 0, 0 }, > + { 0, 0, 0, 0 }, > + { 0, 0, 0, 0 }, > + { 0, 0, 0, 0 }, > + { 0, 0, 0, 0 }, > +}; > + > +static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn) > +{ > + if (nk_mode == 1) { > + for (int k = 0; k < no_qn; k++) { > + qn[k] = ff_aac_get_vlclbf(gb); > + if (qn[k]) > + qn[k]++; > + } > + return; > + } > + > + for (int k = 0; k < no_qn; k++) > + qn[k] = get_bits(gb, 2) + 2; > + > + if (nk_mode == 2) { > + for (int k = 0; k < no_qn; k++) { > + if (qn[k] > 4) { > + qn[k] = ff_aac_get_vlclbf(gb); > + if (qn[k]) > + qn[k] += 4; > + } > + } > + return; > + } > + > + for (int k = 0; k < no_qn; k++) { > + if (qn[k] > 4) { > + int qn_ext = ff_aac_get_vlclbf(gb); > + switch (qn_ext) { > + case 0: qn[k] = 5; break; > + case 1: qn[k] = 6; break; > + case 2: qn[k] = 0; break; > + default: qn[k] = qn_ext + 4; break; > + } > + } > + } > +} > + > +static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv, > + int nk_mode, int no_qn) > +{ > + int n, nk; > + > + int qn[2]; > + parse_qn(gb, qn, nk_mode, no_qn); > + > + for (int k = 0; k < no_qn; k++) { > + if (qn[k] > 4) { > + nk = (qn[k] - 3) / 2; > + n = qn[k] - nk*2; > + } else { > + nk = 0; > + n = qn[k]; > + } > + } > + > + int idx = get_bits(gb, 4*n); > + > + if (nk > 0) > + for (int i = 0; i < 8; i++) > + kv[i] = get_bits(gb, nk); > + > + return 0; > +} > + > +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, > + int use_gain, int len) > +{ > + int ret; > + if (use_gain) > + ce->fac.gain = get_bits(gb, 7); > + > + for (int i = 0; i < len/8; i++) { > + ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1); > + if (ret < 0) > + return ret; > + } > + > + return 0; > +} > + > +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, > + AACUsacElemData *ce, GetBitContext *gb) > +{ > + ce->ldp.acelp_core_mode = get_bits(gb, 3); > + ce->ldp.lpd_mode = get_bits(gb, 5); > + > + ce->ldp.bpf_control_info = get_bits1(gb); > + ce->ldp.core_mode_last = get_bits1(gb); > + ce->ldp.fac_data_present = get_bits1(gb); > + > + const uint8_t *mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode]; > + > + int first_ldp_flag = !ce->ldp.core_mode_last; > + int first_tcx_flag = 1; > + if (first_ldp_flag) > + ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */ > + > + int k = 0; > + while (k < 0) { > + if (!k) { > + if (ce->ldp.core_mode_last && ce->ldp.fac_data_present) > + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); > + } else { > + if (!ce->ldp.last_lpd_mode && mod[k] > 0 || > + ce->ldp.last_lpd_mode && !mod[k]) > + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); > + } > + if (!mod[k]) { > +// parse_acelp_coding(); > + ce->ldp.last_lpd_mode = 0; > + k++; > + } else { > +// parse_tcx_coding(); > + ce->ldp.last_lpd_mode = mod[k]; > + k += (1 << (mod[k] - 1)); > + first_tcx_flag = 0; > + } > + } > + > +// parse_lpc_data(first_lpd_flag); > + > + if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) { > + uint16_t len_8 = usac->core_frame_len / 8; > + uint16_t len_16 = usac->core_frame_len / 16; > + uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16; > + int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len); > + if (ret < 0) > + return ret; > + } > + > + return 0; > +} > diff --git a/libavcodec/aac/aacdec_lpd.h b/libavcodec/aac/aacdec_lpd.h > new file mode 100644 > index 0000000000..924ff75e52 > --- /dev/null > +++ b/libavcodec/aac/aacdec_lpd.h > @@ -0,0 +1,33 @@ > +/* > + * Copyright (c) 2024 Lynne <dev@lynne.ee> > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#ifndef AVCODEC_AAC_AACDEC_LPD_H > +#define AVCODEC_AAC_AACDEC_LPD_H > + > +#include "aacdec.h" > +#include "libavcodec/get_bits.h" > + > +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, > + int use_gain, int len); > + > +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, > + AACUsacElemData *ce, GetBitContext *gb); > + > +#endif /* AVCODEC_AAC_AACDEC_LPD_H */ > diff --git a/libavcodec/aac/aacdec_usac.c b/libavcodec/aac/aacdec_usac.c > new file mode 100644 > index 0000000000..4b48c4d6ca > --- /dev/null > +++ b/libavcodec/aac/aacdec_usac.c > @@ -0,0 +1,1230 @@ > +/* > + * Copyright (c) 2024 Lynne <dev@lynne.ee> > + * > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA > + */ > + > +#include "aacdec_usac.h" > +#include "aacdec_tab.h" > +#include "aacdec_lpd.h" > +#include "aacdec_ac.h" > + > +#include "libavcodec/opusdsp.h" > +#include "libavcodec/aactab.h" > +#include "libavutil/mem.h" > +#include "libavcodec/mpeg4audio.h" > + > +/* Number of scalefactor bands per complex prediction band, equal to 2. */ > +#define SFB_PER_PRED_BAND 2 > + > +static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3) > +{ > + uint32_t val = get_bits(gb, nb1); > + if (val < ((1 << nb1) - 1)) > + return val; > + > + val += get_bits(gb, nb2); > + if (val == ((1 << nb2) - 1)) > + val += get_bits(gb, nb3); > + > + return val; > +} > + > +static int aac_usac_samplerate[] = { Missing const > + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, > + 16000, 12000, 11025, 8000, 7350, -1, -1, 57600, 51200, > + 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, -1, -1, -1, -1, > +}; > + > +static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e, > + GetBitContext *gb) > +{ > + if (get_bits1(gb)) { /* usacExtElementPresent */ > + uint32_t len; > + if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */ > + len = e->ext.default_len; > + } else { > + len = get_bits(gb, 8); /* usacExtElementPayloadLength */ > + if (len == 255) > + len += get_bits(gb, 16) - 2; > + } > + > + if (len) { > + uint8_t *tmp; > + uint8_t pl_frag_start = 1; > + uint8_t pl_frag_end = 1; > + if (e->ext.payload_frag) { > + pl_frag_start = get_bits1(gb); /* usacExtElementStart */ > + pl_frag_end = get_bits1(gb); /* usacExtElementStop */ > + } > + > + if (pl_frag_start) > + e->ext.pl_data_offset = 0; > + > + tmp = av_realloc(e->ext.pl_data, e->ext.pl_data_offset + len); > + if (!tmp) { > + free(e->ext.pl_data); Wrong deallocator. > + return AVERROR(ENOMEM); > + } > + e->ext.pl_data = tmp; > + > + for (int i = 0; i < len; i++) > + e->ext.pl_data[e->ext.pl_data_offset + i] = get_bits(gb, 8); > + > + if (pl_frag_end) { > + int ret; > + e->ext.pl_data_offset = 0; > + switch (e->ext.type) { > + case ID_EXT_ELE_FILL: > + av_freep(&e->ext.pl_data); > + break; > + case ID_EXT_ELE_AUDIOPREROLL: > + ret = parse_audio_preroll(ac, e->ext.pl_data, > + e->ext.pl_data_offset); > + if (ret < 0) { > + av_freep(&e->ext.pl_data); > + return ret; > + } > + break; > + default: > + av_freep(&e->ext.pl_data); Pointless if you abort in the next line > + /* This should never happen */ > + av_assert0(0); > + } > + } > + } > + } > + > + return 0; > +} > + > +#include "libavcodec/opusdsp.h" > + > +#ifndef AVCODEC_AAC_AACDEC_USAC_H > +#define AVCODEC_AAC_AACDEC_USAC_H > + > +#include "aacdec.h" > + > +#include "libavcodec/get_bits.h" > + > +static inline uint8_t ff_aac_get_vlclbf(GetBitContext *gb) > +{ > + uint8_t ret = 0; > + while (get_bits1(gb) && ret <= 36) > + ret++; > + return ret; > +} Look at unary.h > + > +int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx, > + GetBitContext *gb, OutputConfiguration *oc, > + int channel_config); > + > +int ff_aac_usac_reset_state(AACDecContext *ac, OutputConfiguration *oc); > + > +int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac, > + GetBitContext *gb, int *got_frame_ptr); > + > +#endif /* AVCODEC_AAC_AACDEC_USAC_H */ _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
* Re: [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) 2024-05-16 10:26 ` Andreas Rheinhardt @ 2024-05-16 15:00 ` Lynne via ffmpeg-devel 0 siblings, 0 replies; 13+ messages in thread From: Lynne via ffmpeg-devel @ 2024-05-16 15:00 UTC (permalink / raw) To: ffmpeg-devel; +Cc: Lynne [-- Attachment #1.1.1.1: Type: text/plain, Size: 33773 bytes --] On 16/05/2024 12:26, Andreas Rheinhardt wrote: > Lynne via ffmpeg-devel: >> This commit adds a decoder for the frequency-domain part of USAC. >> >> What works: >> - Mono >> - Stereo (no prediction) >> - Stereo (mid/side coding) >> >> What doesn't: >> - Preroll decoding (every single decoder seems faulty or weird?) >> - Complex stereo prediction >> >> Known issues: >> - Spec incompliance (noise synthesis in particular) >> - Lack of robustness >> --- > > >> diff --git a/libavcodec/aac/aacdec.h b/libavcodec/aac/aacdec.h >> index 20545a24d4..3e6592cf0e 100644 >> --- a/libavcodec/aac/aacdec.h >> +++ b/libavcodec/aac/aacdec.h >> @@ -42,6 +42,8 @@ >> #include "libavcodec/avcodec.h" >> #include "libavcodec/mpeg4audio.h" >> >> +#include "aacdec_ac.h" >> + >> typedef struct AACDecContext AACDecContext; >> >> /** >> @@ -69,6 +71,32 @@ enum CouplingPoint { >> AFTER_IMDCT = 3, >> }; >> >> +enum AACUsacElem { >> + ID_USAC_SCE = 0, >> + ID_USAC_CPE = 1, >> + ID_USAC_LFE = 2, >> + ID_USAC_EXT = 3, >> +}; >> + >> +enum ExtensionHeaderType { >> + ID_CONFIG_EXT_FILL = 0, >> + ID_CONFIG_EXT_LOUDNESS_INFO = 2, >> + ID_CONFIG_EXT_STREAM_ID = 7, >> +}; >> + >> +enum AACUsacExtension { >> + ID_EXT_ELE_FILL, >> + ID_EXT_ELE_MPEGS, >> + ID_EXT_ELE_SAOC, >> + ID_EXT_ELE_AUDIOPREROLL, >> + ID_EXT_ELE_UNI_DRC, >> +}; >> + >> +enum AACUSACLoudnessExt { >> + UNIDRCLOUDEXT_TERM = 0x0, >> + UNIDRCLOUDEXT_EQ = 0x1, >> +}; >> + >> // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. >> #define RENAME_FIXED(name) name ## _fixed >> >> @@ -93,6 +121,40 @@ typedef struct LongTermPrediction { >> int8_t used[MAX_LTP_LONG_SFB]; >> } LongTermPrediction; >> >> +/* Per channel core mode */ >> +typedef struct AACUsacElemData { >> + uint8_t core_mode; >> + uint8_t scale_factor_grouping; >> + >> + /* Timewarping ratio */ >> +#define NUM_TW_NODES 16 >> + uint8_t tw_ratio[NUM_TW_NODES]; >> + >> + struct { >> + uint8_t acelp_core_mode : 3; >> + uint8_t lpd_mode : 5; >> + >> + uint8_t bpf_control_info : 1; >> + uint8_t core_mode_last : 1; >> + uint8_t fac_data_present : 1; >> + >> + int last_lpd_mode; >> + } ldp; >> + >> + struct { >> + unsigned int seed; >> + uint8_t level : 3; >> + uint8_t offset : 5; >> + } noise; >> + >> + struct { >> + uint8_t gain; >> + uint32_t kv[8 /* (1024 / 16) / 8 */][8]; >> + } fac; >> + >> + AACArithState ac; >> +} AACUsacElemData; >> + >> /** >> * Individual Channel Stream >> */ >> @@ -145,6 +207,7 @@ typedef struct ChannelCoupling { >> */ >> typedef struct SingleChannelElement { >> IndividualChannelStream ics; >> + AACUsacElemData ue; ///< USAC element data >> TemporalNoiseShaping tns; >> enum BandType band_type[128]; ///< band types >> int sfo[128]; ///< scalefactor offsets >> @@ -163,25 +226,141 @@ typedef struct SingleChannelElement { >> }; >> } SingleChannelElement; >> >> +typedef struct AACUsacStereo { >> + uint8_t common_window; >> + uint8_t common_tw; >> + >> + uint8_t ms_mask_mode; >> + uint8_t config_idx; >> + >> + struct { >> + uint8_t use_prev_frame; >> + uint8_t pred_dir; >> + uint8_t delta_code_time; >> + uint8_t pred_used[8][64]; >> + >> + AVComplexFloat pred[8][64]; >> + } cplx; >> +} AACUsacStereo; >> + >> /** >> * channel element - generic struct for SCE/CPE/CCE/LFE >> */ >> typedef struct ChannelElement { >> int present; >> // CPE specific >> + uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values >> uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band >> // shared >> SingleChannelElement ch[2]; >> // CCE specific >> ChannelCoupling coup; >> + // USAC stereo coupling data >> + AACUsacStereo us; >> } ChannelElement; >> >> +typedef struct AACUSACLoudnessInfo { >> + uint8_t drc_set_id : 6; >> + uint8_t downmix_id : 7; >> + struct { >> + uint16_t lvl : 12; >> + uint8_t present : 1; >> + } sample_peak; >> + >> + struct { >> + uint16_t lvl : 12; >> + uint8_t measurement : 4; >> + uint8_t reliability : 2; >> + uint8_t present : 1; >> + } true_peak; >> + >> + uint8_t nb_measurements : 4; >> + struct { >> + uint8_t method_def : 4; >> + uint8_t method_val; >> + uint8_t measurement : 4; >> + uint8_t reliability : 2; >> + } measurements[16]; >> +} AACUSACLoudnessInfo; >> + >> +typedef struct AACUsacElemConfig { >> + enum AACUsacElem type; >> + >> + uint8_t tw_mdct : 1; >> + uint8_t noise_fill : 1; >> + >> + uint8_t stereo_config_index; >> + >> + struct { >> + int ratio; >> + >> + uint8_t harmonic_sbr : 1; /* harmonicSBR */ >> + uint8_t bs_intertes : 1; /* bs_interTes */ >> + uint8_t bs_pvc : 1; /* bs_pvc */ >> + >> + struct { >> + uint8_t start_freq; /* dflt_start_freq */ >> + uint8_t stop_freq; /* dflt_stop_freq */ >> + >> + uint8_t freq_scale; /* dflt_freq_scale */ >> + uint8_t alter_scale : 1; /* dflt_alter_scale */ >> + uint8_t noise_scale; /* dflt_noise_scale */ >> + >> + uint8_t limiter_bands; /* dflt_limiter_bands */ >> + uint8_t limiter_gains; /* dflt_limiter_gains */ >> + uint8_t interpol_freq : 1; /* dflt_interpol_freq */ >> + uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ >> + } dflt; >> + } sbr; >> + >> + struct { >> + uint8_t freq_res; /* bsFreqRes */ >> + uint8_t fixed_gain; /* bsFixedGainDMX */ >> + uint8_t temp_shape_config; /* bsTempShapeConfig */ >> + uint8_t decorr_config; /* bsDecorrConfig */ >> + uint8_t high_rate_mode : 1; /* bsHighRateMode */ >> + uint8_t phase_coding : 1; /* bsPhaseCoding */ >> + >> + uint8_t otts_bands_phase; /* bsOttBandsPhase */ >> + uint8_t residual_coding; /* bsResidualCoding */ >> + uint8_t residual_bands; /* bsResidualBands */ >> + uint8_t pseudo_lr : 1; /* bsPseudoLr */ >> + uint8_t env_quant_mode : 1; /* bsEnvQuantMode */ > > Is using bitfields really worth it given that they force to use masking > for accesses? > >> + } mps; >> + >> + struct { >> + enum AACUsacExtension type; >> + uint8_t payload_frag; >> + uint32_t default_len; >> + uint32_t pl_data_offset; >> + uint8_t *pl_data; >> + } ext; >> +} AACUsacElemConfig; >> + >> +typedef struct AACUSACConfig { >> + uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ >> + uint8_t rate_idx; >> + uint16_t core_frame_len; >> + uint16_t stream_identifier; >> + >> + AACUsacElemConfig elems[64]; >> + int nb_elems; >> + >> + struct { >> + uint8_t nb_album; >> + AACUSACLoudnessInfo album_info[64]; >> + uint8_t nb_info; >> + AACUSACLoudnessInfo info[64]; >> + } loudness; >> +} AACUSACConfig; >> + >> typedef struct OutputConfiguration { >> MPEG4AudioConfig m4ac; >> uint8_t layout_map[MAX_ELEM_ID*4][3]; >> int layout_map_tags; >> AVChannelLayout ch_layout; >> enum OCStatus status; >> + AACUSACConfig usac; >> } OutputConfiguration; >> >> /** >> diff --git a/libavcodec/aac/aacdec_ac.c b/libavcodec/aac/aacdec_ac.c >> new file mode 100644 >> index 0000000000..326d716bd3 >> --- /dev/null >> +++ b/libavcodec/aac/aacdec_ac.c >> @@ -0,0 +1,224 @@ >> +/* >> + * AAC definitions and structures >> + * Copyright (c) 2024 Lynne >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA >> + */ >> + >> +#include "libavcodec/aactab.h" >> +#include "aacdec_ac.h" >> + >> +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int N) >> +{ >> + float ratio; >> + if (reset) { >> + memset(state->last, 0, sizeof(state->last)); >> + state->last_len = N; >> + memset(state->cur, 0, sizeof(state->cur)); >> + state->cur[3] = 0; >> + state->cur[2] = 0; >> + state->cur[1] = 0; >> + state->cur[0] = 1; >> + state->last[0] = 0 << 12; >> + state->state_pre = 0; >> + return 0; >> + } else if (state->last_len != N) { >> + int i; >> + uint8_t last[512 /* 2048 / 4 */]; >> + memcpy(last, state->last, sizeof(last)); >> + >> + ratio = state->last_len / (float)N; >> + for (i = 0; i < N/2; i++) { >> + int k = (int)(i * ratio); >> + state->last[i] = last[k]; >> + } >> + >> + for (; i < FF_ARRAY_ELEMS(state->last); i++) >> + state->last[i] = 0; >> + >> + state->last_len = N; >> + } >> + >> + memset(state->cur, 0, sizeof(state->cur)); >> + state->cur[3] = 0; >> + state->cur[2] = 0; >> + state->cur[1] = 0; >> + state->cur[0] = 1; >> + >> + state->state_pre = state->last[0] << 12; >> + return state->last[0] << 12; >> +} >> + >> +extern int ec_debug; >> + >> +int trig = 0; >> + >> +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t c, int i, int N) >> +{ >> + c = state->state_pre >> 8; >> + c = c + (state->last[i + 1] << 8); >> + c = (c << 4); >> + c += state->cur[1]; >> + >> + state->state_pre = c; >> + >> + if (i > 3 && >> + ((state->cur[3] + state->cur[2] + state->cur[1]) < 5)) >> + return c + 0x10000; >> + >> + return c; >> +} >> + >> +uint32_t ff_aac_ac_get_pk(uint32_t c) >> +{ >> + int i_min = -1; >> + int i, j; >> + int i_max = FF_ARRAY_ELEMS(ff_aac_ac_lookup_m) - 1; >> + while ((i_max - i_min) > 1) { >> + i = i_min + ((i_max - i_min) / 2); >> + j = ff_aac_ac_hash_m[i]; >> + if (c < (j >> 8)) >> + i_max = i; >> + else if (c > (j >> 8)) >> + i_min = i; >> + else >> + return (j & 0xFF); >> + } >> + return ff_aac_ac_lookup_m[i_max]; >> +} >> + >> +void ff_aac_ac_update_context(AACArithState *state, int idx, >> + uint16_t a, uint16_t b) >> +{ >> + state->cur[0] = a + b + 1; >> + if (state->cur[0] > 0xF) >> + state->cur[0] = 0xF; >> + >> + state->cur[3] = state->cur[2]; >> + state->cur[2] = state->cur[1]; >> + state->cur[1] = state->cur[0]; >> + >> + state->last[idx] = state->cur[0]; >> +} >> + >> +/* Initialize AC */ >> +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb) >> +{ >> + ac->low = 0; >> + ac->high = UINT16_MAX; >> + ac->val = get_bits(gb, 16); >> +} >> + >> +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, >> + const uint16_t *cdf, uint16_t cdf_len) >> +{ >> + int val = ac->val; >> + int low = ac->low; >> + int high = ac->high; >> + >> + int rng = high - low + 1; >> + int c = ((((int)(val - low + 1)) << 14) - ((int)1)); >> + >> + /* Note: this could be done faster via heuristics, the total number of >> + * configurations is low */ >> + const uint16_t *p = cdf - 1; >> + const uint16_t *q; >> + >> + switch (cdf_len) { >> + case 2: >> + if ((p[1] * rng) > c) >> + p += 1; >> + break; >> + case 4: >> + if ((p[2] * rng) > c) >> + p += 2; >> + if ((p[1] * rng) > c) >> + p += 1; >> + break; >> + case 17: >> + /* First check if the current probability is even met at all */ >> + if ((p[1] * rng) <= c) >> + break; >> + p += 1; >> + for (int i = 8; i >= 1; i >>= 1) >> + if ((p[i] * rng) > c) >> + p += i; >> + break; >> + case 27: >> + const uint16_t *p_24 = p + 24; >> + >> + if ((p[16] * rng) > c) >> + p += 16; >> + if ((p[8] * rng) > c) >> + p += 8; >> + if (p != p_24) >> + if ((p[4] * rng) > c) >> + p += 4; >> + if ((p[2] * rng) > c) >> + p += 2; >> + >> + if (p != &p_24[2]) >> + if ((p[1] * rng) > c) >> + p += 1; >> + break; >> + default: >> + /* This should never happen */ >> + av_assert2(0); >> + } >> + >> + int sym = (int)((ptrdiff_t)(p - cdf)) + 1; >> + if (sym) >> + high = low + ((rng * cdf[sym - 1]) >> 14) - 1; >> + low += (rng * cdf[sym]) >> 14; >> + >> + /* This loop could be done faster */ >> + while (1) { >> + if (high < 32768) { >> + ; >> + } else if (low >= 32768) { >> + val -= 32768; >> + low -= 32768; >> + high -= 32768; >> + } else if (low >= 16384 && high < 49152) { >> + val -= 16384; >> + low -= 16384; >> + high -= 16384; >> + } else { >> + break; >> + } >> + low += low; >> + high += high + 1; >> + val = (val << 1) | get_bits1(gb); >> + }; >> + >> + ac->low = low; >> + ac->high = high; >> + ac->val = val; >> + >> + return sym; >> +} >> + >> +void ff_aac_ac_finish(AACArithState *state, int offset, int N) >> +{ >> + int i; >> + >> + for (i = offset; i < N/2; i++) >> + state->last[i] = 1; >> + >> + for (; i < FF_ARRAY_ELEMS(state->last); i++) >> + state->last[i] = 0; >> +} >> diff --git a/libavcodec/aac/aacdec_ac.h b/libavcodec/aac/aacdec_ac.h >> new file mode 100644 >> index 0000000000..ef96bed770 >> --- /dev/null >> +++ b/libavcodec/aac/aacdec_ac.h >> @@ -0,0 +1,54 @@ >> +/* >> + * AAC definitions and structures >> + * Copyright (c) 2024 Lynne >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA >> + */ >> + >> +#ifndef AVCODEC_AACDEC_AC_H >> +#define AVCODEC_AACDEC_AC_H >> + >> +#include "libavcodec/get_bits.h" >> + >> +typedef struct AACArithState { >> + uint8_t last[512 /* 2048 / 4 */]; >> + int last_len; >> + uint8_t cur[4]; >> + uint16_t state_pre; >> +} AACArithState; >> + >> +typedef struct AACArith { >> + uint16_t low; >> + uint16_t high; >> + uint16_t val; >> +} AACArith; >> + >> +#define FF_AAC_AC_ESCAPE 16 >> + >> +uint32_t ff_aac_ac_map_process(AACArithState *state, int reset, int len); >> +uint32_t ff_aac_ac_get_context(AACArithState *state, uint32_t old_c, int idx, int len); >> +uint32_t ff_aac_ac_get_pk(uint32_t c); >> + >> +void ff_aac_ac_update_context(AACArithState *state, int idx, uint16_t a, uint16_t b); >> +void ff_aac_ac_init(AACArith *ac, GetBitContext *gb); >> + >> +uint16_t ff_aac_ac_decode(AACArith *ac, GetBitContext *gb, >> + const uint16_t *cdf, uint16_t cdf_len); >> + >> +void ff_aac_ac_finish(AACArithState *state, int offset, int nb); >> + >> +#endif /* AVCODEC_AACDEC_AC_H */ >> diff --git a/libavcodec/aac/aacdec_dsp_template.c b/libavcodec/aac/aacdec_dsp_template.c >> index 59a69d88f3..8d31af22f8 100644 >> --- a/libavcodec/aac/aacdec_dsp_template.c >> +++ b/libavcodec/aac/aacdec_dsp_template.c >> @@ -88,8 +88,8 @@ static void AAC_RENAME(apply_mid_side_stereo)(AACDecContext *ac, ChannelElement >> INTFLOAT *ch1 = cpe->ch[1].AAC_RENAME(coeffs); >> const uint16_t *offsets = ics->swb_offset; >> for (int g = 0; g < ics->num_window_groups; g++) { >> - for (int sfb = 0; sfb < ics->max_sfb; sfb++) { >> - const int idx = g*ics->max_sfb + sfb; >> + for (int sfb = 0; sfb < cpe->max_sfb_ste; sfb++) { >> + const int idx = g*cpe->max_sfb_ste + sfb; >> if (cpe->ms_mask[idx] && >> cpe->ch[0].band_type[idx] < NOISE_BT && >> cpe->ch[1].band_type[idx] < NOISE_BT) { >> diff --git a/libavcodec/aac/aacdec_latm.h b/libavcodec/aac/aacdec_latm.h >> index e40a2fe1a7..047c11e0fb 100644 >> --- a/libavcodec/aac/aacdec_latm.h >> +++ b/libavcodec/aac/aacdec_latm.h >> @@ -56,7 +56,8 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, >> { >> AACDecContext *ac = &latmctx->aac_ctx; >> AVCodecContext *avctx = ac->avctx; >> - MPEG4AudioConfig m4ac = { 0 }; >> + OutputConfiguration oc = { 0 }; >> + MPEG4AudioConfig *m4ac = &oc.m4ac; >> GetBitContext gbc; >> int config_start_bit = get_bits_count(gb); >> int sync_extension = 0; >> @@ -76,7 +77,7 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, >> if (get_bits_left(gb) <= 0) >> return AVERROR_INVALIDDATA; >> >> - bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, >> + bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc, >> &gbc, config_start_bit, >> sync_extension); >> >> @@ -88,11 +89,12 @@ static int latm_decode_audio_specific_config(struct LATMContext *latmctx, >> asclen = bits_consumed; >> >> if (!latmctx->initialized || >> - ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || >> - ac->oc[1].m4ac.chan_config != m4ac.chan_config) { >> + ac->oc[1].m4ac.sample_rate != m4ac->sample_rate || >> + ac->oc[1].m4ac.chan_config != m4ac->chan_config) { >> >> if (latmctx->initialized) { >> - av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); >> + av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", >> + m4ac->sample_rate, m4ac->chan_config); >> } else { >> av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); >> } >> @@ -280,7 +282,7 @@ static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, >> } else { >> push_output_configuration(&latmctx->aac_ctx); >> if ((err = decode_audio_specific_config( >> - &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, >> + &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1], >> avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { >> pop_output_configuration(&latmctx->aac_ctx); >> return err; >> diff --git a/libavcodec/aac/aacdec_lpd.c b/libavcodec/aac/aacdec_lpd.c >> new file mode 100644 >> index 0000000000..be39e2c175 >> --- /dev/null >> +++ b/libavcodec/aac/aacdec_lpd.c >> @@ -0,0 +1,192 @@ >> +/* >> + * Copyright (c) 2024 Lynne <dev@lynne.ee> >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA >> + */ >> + >> +#include "aacdec_lpd.h" >> +#include "aacdec_usac.h" >> + >> +const uint8_t ff_aac_lpd_mode_tab[32][4] = { >> + { 0, 0, 0, 0 }, >> + { 1, 0, 0, 0 }, >> + { 0, 1, 0, 0 }, >> + { 1, 1, 0, 0 }, >> + { 0, 0, 1, 0 }, >> + { 1, 0, 1, 0 }, >> + { 0, 1, 1, 0 }, >> + { 1, 1, 1, 0 }, >> + { 0, 0, 0, 1 }, >> + { 1, 0, 0, 1 }, >> + { 0, 1, 0, 1 }, >> + { 1, 1, 0, 1 }, >> + { 0, 0, 1, 1 }, >> + { 1, 0, 1, 1 }, >> + { 0, 1, 1, 1 }, >> + { 1, 1, 1, 1 }, >> + { 2, 2, 0, 0 }, >> + { 2, 2, 1, 0 }, >> + { 2, 2, 0, 1 }, >> + { 2, 2, 1, 1 }, >> + { 0, 0, 2, 2 }, >> + { 1, 0, 2, 2 }, >> + { 0, 1, 2, 2 }, >> + { 1, 1, 2, 2 }, >> + { 2, 2, 2, 2 }, >> + { 3, 3, 3, 3 }, >> + /* Larger values are reserved, but permit them for resilience */ >> + { 0, 0, 0, 0 }, >> + { 0, 0, 0, 0 }, >> + { 0, 0, 0, 0 }, >> + { 0, 0, 0, 0 }, >> + { 0, 0, 0, 0 }, >> + { 0, 0, 0, 0 }, >> +}; >> + >> +static void parse_qn(GetBitContext *gb, int *qn, int nk_mode, int no_qn) >> +{ >> + if (nk_mode == 1) { >> + for (int k = 0; k < no_qn; k++) { >> + qn[k] = ff_aac_get_vlclbf(gb); >> + if (qn[k]) >> + qn[k]++; >> + } >> + return; >> + } >> + >> + for (int k = 0; k < no_qn; k++) >> + qn[k] = get_bits(gb, 2) + 2; >> + >> + if (nk_mode == 2) { >> + for (int k = 0; k < no_qn; k++) { >> + if (qn[k] > 4) { >> + qn[k] = ff_aac_get_vlclbf(gb); >> + if (qn[k]) >> + qn[k] += 4; >> + } >> + } >> + return; >> + } >> + >> + for (int k = 0; k < no_qn; k++) { >> + if (qn[k] > 4) { >> + int qn_ext = ff_aac_get_vlclbf(gb); >> + switch (qn_ext) { >> + case 0: qn[k] = 5; break; >> + case 1: qn[k] = 6; break; >> + case 2: qn[k] = 0; break; >> + default: qn[k] = qn_ext + 4; break; >> + } >> + } >> + } >> +} >> + >> +static int parse_codebook_idx(GetBitContext *gb, uint32_t *kv, >> + int nk_mode, int no_qn) >> +{ >> + int n, nk; >> + >> + int qn[2]; >> + parse_qn(gb, qn, nk_mode, no_qn); >> + >> + for (int k = 0; k < no_qn; k++) { >> + if (qn[k] > 4) { >> + nk = (qn[k] - 3) / 2; >> + n = qn[k] - nk*2; >> + } else { >> + nk = 0; >> + n = qn[k]; >> + } >> + } >> + >> + int idx = get_bits(gb, 4*n); >> + >> + if (nk > 0) >> + for (int i = 0; i < 8; i++) >> + kv[i] = get_bits(gb, nk); >> + >> + return 0; >> +} >> + >> +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, >> + int use_gain, int len) >> +{ >> + int ret; >> + if (use_gain) >> + ce->fac.gain = get_bits(gb, 7); >> + >> + for (int i = 0; i < len/8; i++) { >> + ret = parse_codebook_idx(gb, ce->fac.kv[i], 1, 1); >> + if (ret < 0) >> + return ret; >> + } >> + >> + return 0; >> +} >> + >> +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, >> + AACUsacElemData *ce, GetBitContext *gb) >> +{ >> + ce->ldp.acelp_core_mode = get_bits(gb, 3); >> + ce->ldp.lpd_mode = get_bits(gb, 5); >> + >> + ce->ldp.bpf_control_info = get_bits1(gb); >> + ce->ldp.core_mode_last = get_bits1(gb); >> + ce->ldp.fac_data_present = get_bits1(gb); >> + >> + const uint8_t *mod = ff_aac_lpd_mode_tab[ce->ldp.lpd_mode]; >> + >> + int first_ldp_flag = !ce->ldp.core_mode_last; >> + int first_tcx_flag = 1; >> + if (first_ldp_flag) >> + ce->ldp.last_lpd_mode = -1; /* last_ldp_mode is a **STATEFUL** value */ >> + >> + int k = 0; >> + while (k < 0) { >> + if (!k) { >> + if (ce->ldp.core_mode_last && ce->ldp.fac_data_present) >> + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); >> + } else { >> + if (!ce->ldp.last_lpd_mode && mod[k] > 0 || >> + ce->ldp.last_lpd_mode && !mod[k]) >> + ff_aac_parse_fac_data(ce, gb, 0, usac->core_frame_len/8); >> + } >> + if (!mod[k]) { >> +// parse_acelp_coding(); >> + ce->ldp.last_lpd_mode = 0; >> + k++; >> + } else { >> +// parse_tcx_coding(); >> + ce->ldp.last_lpd_mode = mod[k]; >> + k += (1 << (mod[k] - 1)); >> + first_tcx_flag = 0; >> + } >> + } >> + >> +// parse_lpc_data(first_lpd_flag); >> + >> + if (!ce->ldp.core_mode_last && ce->ldp.fac_data_present) { >> + uint16_t len_8 = usac->core_frame_len / 8; >> + uint16_t len_16 = usac->core_frame_len / 16; >> + uint16_t fac_len = get_bits1(gb) /* short_fac_flag */ ? len_8 : len_16; >> + int ret = ff_aac_parse_fac_data(ce, gb, 1, fac_len); >> + if (ret < 0) >> + return ret; >> + } >> + >> + return 0; >> +} >> diff --git a/libavcodec/aac/aacdec_lpd.h b/libavcodec/aac/aacdec_lpd.h >> new file mode 100644 >> index 0000000000..924ff75e52 >> --- /dev/null >> +++ b/libavcodec/aac/aacdec_lpd.h >> @@ -0,0 +1,33 @@ >> +/* >> + * Copyright (c) 2024 Lynne <dev@lynne.ee> >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA >> + */ >> + >> +#ifndef AVCODEC_AAC_AACDEC_LPD_H >> +#define AVCODEC_AAC_AACDEC_LPD_H >> + >> +#include "aacdec.h" >> +#include "libavcodec/get_bits.h" >> + >> +int ff_aac_parse_fac_data(AACUsacElemData *ce, GetBitContext *gb, >> + int use_gain, int len); >> + >> +int ff_aac_ldp_parse_channel_stream(AACDecContext *ac, AACUSACConfig *usac, >> + AACUsacElemData *ce, GetBitContext *gb); >> + >> +#endif /* AVCODEC_AAC_AACDEC_LPD_H */ >> diff --git a/libavcodec/aac/aacdec_usac.c b/libavcodec/aac/aacdec_usac.c >> new file mode 100644 >> index 0000000000..4b48c4d6ca >> --- /dev/null >> +++ b/libavcodec/aac/aacdec_usac.c >> @@ -0,0 +1,1230 @@ >> +/* >> + * Copyright (c) 2024 Lynne <dev@lynne.ee> >> + * >> + * This file is part of FFmpeg. >> + * >> + * FFmpeg is free software; you can redistribute it and/or >> + * modify it under the terms of the GNU Lesser General Public >> + * License as published by the Free Software Foundation; either >> + * version 2.1 of the License, or (at your option) any later version. >> + * >> + * FFmpeg is distributed in the hope that it will be useful, >> + * but WITHOUT ANY WARRANTY; without even the implied warranty of >> + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU >> + * Lesser General Public License for more details. >> + * >> + * You should have received a copy of the GNU Lesser General Public >> + * License along with FFmpeg; if not, write to the Free Software >> + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA >> + */ >> + >> +#include "aacdec_usac.h" >> +#include "aacdec_tab.h" >> +#include "aacdec_lpd.h" >> +#include "aacdec_ac.h" >> + >> +#include "libavcodec/opusdsp.h" >> +#include "libavcodec/aactab.h" >> +#include "libavutil/mem.h" >> +#include "libavcodec/mpeg4audio.h" >> + >> +/* Number of scalefactor bands per complex prediction band, equal to 2. */ >> +#define SFB_PER_PRED_BAND 2 >> + >> +static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, int nb3) >> +{ >> + uint32_t val = get_bits(gb, nb1); >> + if (val < ((1 << nb1) - 1)) >> + return val; >> + >> + val += get_bits(gb, nb2); >> + if (val == ((1 << nb2) - 1)) >> + val += get_bits(gb, nb3); >> + >> + return val; >> +} >> + >> +static int aac_usac_samplerate[] = { > > Missing const > >> + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, >> + 16000, 12000, 11025, 8000, 7350, -1, -1, 57600, 51200, >> + 40000, 38400, 34150, 28800, 25600, 20000, 19200, 17075, 14400, 12800, 9600, -1, -1, -1, -1, >> +}; >> + > >> +static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e, >> + GetBitContext *gb) >> +{ >> + if (get_bits1(gb)) { /* usacExtElementPresent */ >> + uint32_t len; >> + if (get_bits1(gb)) { /* usacExtElementUseDefaultLength */ >> + len = e->ext.default_len; >> + } else { >> + len = get_bits(gb, 8); /* usacExtElementPayloadLength */ >> + if (len == 255) >> + len += get_bits(gb, 16) - 2; >> + } >> + >> + if (len) { >> + uint8_t *tmp; >> + uint8_t pl_frag_start = 1; >> + uint8_t pl_frag_end = 1; >> + if (e->ext.payload_frag) { >> + pl_frag_start = get_bits1(gb); /* usacExtElementStart */ >> + pl_frag_end = get_bits1(gb); /* usacExtElementStop */ >> + } >> + >> + if (pl_frag_start) >> + e->ext.pl_data_offset = 0; >> + >> + tmp = av_realloc(e->ext.pl_data, e->ext.pl_data_offset + len); >> + if (!tmp) { >> + free(e->ext.pl_data); > > Wrong deallocator. > >> + return AVERROR(ENOMEM); >> + } >> + e->ext.pl_data = tmp; >> + >> + for (int i = 0; i < len; i++) >> + e->ext.pl_data[e->ext.pl_data_offset + i] = get_bits(gb, 8); >> + >> + if (pl_frag_end) { >> + int ret; >> + e->ext.pl_data_offset = 0; >> + switch (e->ext.type) { >> + case ID_EXT_ELE_FILL: >> + av_freep(&e->ext.pl_data); >> + break; >> + case ID_EXT_ELE_AUDIOPREROLL: >> + ret = parse_audio_preroll(ac, e->ext.pl_data, >> + e->ext.pl_data_offset); >> + if (ret < 0) { >> + av_freep(&e->ext.pl_data); >> + return ret; >> + } >> + break; >> + default: >> + av_freep(&e->ext.pl_data); > > Pointless if you abort in the next line > >> + /* This should never happen */ >> + av_assert0(0); >> + } >> + } >> + } >> + } >> + >> + return 0; >> +} >> + > > >> +#include "libavcodec/opusdsp.h" >> + >> +#ifndef AVCODEC_AAC_AACDEC_USAC_H >> +#define AVCODEC_AAC_AACDEC_USAC_H >> + >> +#include "aacdec.h" >> + >> +#include "libavcodec/get_bits.h" >> + >> +static inline uint8_t ff_aac_get_vlclbf(GetBitContext *gb) >> +{ >> + uint8_t ret = 0; >> + while (get_bits1(gb) && ret <= 36) >> + ret++; >> + return ret; >> +} > > Look at unary.h That's convenient, thanks. I've synced my changes on my repo in https://github.com/cyanreg/FFmpeg/tree/xhe Though I'll likely upload the patchset on the ML again tomorrow with some fixes for preroll parsing and complex synth. [-- Attachment #1.1.1.2: OpenPGP public key --] [-- Type: application/pgp-keys, Size: 637 bytes --] [-- Attachment #1.2: OpenPGP digital signature --] [-- Type: application/pgp-signature, Size: 236 bytes --] [-- Attachment #2: Type: text/plain, Size: 251 bytes --] _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". ^ permalink raw reply [flat|nested] 13+ messages in thread
end of thread, other threads:[~2024-05-16 18:46 UTC | newest] Thread overview: 13+ messages (download: mbox.gz / follow: Atom feed) -- links below jump to the message on this page -- 2024-05-16 10:08 [FFmpeg-devel] [PATCH 0/8] aacdec: add a native xHE-AAC decoder Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 1/8] aacdec: move from scalefactor ranged arrays to flat arrays Lynne via ffmpeg-devel 2024-05-16 18:30 ` Michael Niedermayer 2024-05-16 18:46 ` Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 2/8] aacdec: expose channel layout related functions Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 3/8] aacdec: expose decode_tns Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 4/8] aacdec_dsp: implement 768-point transform and windowing Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 5/8] aactab: add deemphasis tables for USAC Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 6/8] aactab: add tables for the new USAC arithmetic coder Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 7/8] aactab: add new scalefactor offset tables for 96/768pt windows Lynne via ffmpeg-devel 2024-05-16 10:08 ` [FFmpeg-devel] [PATCH 8/8] aacdec: add a decoder for AAC USAC (xHE-AAC) Lynne via ffmpeg-devel 2024-05-16 10:26 ` Andreas Rheinhardt 2024-05-16 15:00 ` Lynne via ffmpeg-devel
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