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More AAC support improvement.

Browse Source 
from Brad (maintainer)
This commit is contained in:
ajacoutot 2016-04-07 06:01:19 +00:00
parent 8910843fe9
commit f8df41ea27
9 changed files with 266 additions and 55 deletions
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4
graphics/ffmpeg/Makefile
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@ -1,11 +1,11 @@
# $OpenBSD: Makefile,v 1.147 2016/03/10 20:41:22 naddy Exp $
# $OpenBSD: Makefile,v 1.148 2016/04/07 06:01:19 ajacoutot Exp $
COMMENT= audio/video converter and streamer
V= 20160113
DISTNAME= ffmpeg-git-${V}
PKGNAME= ffmpeg-${V}
REVISION= 6
REVISION= 7
CATEGORIES= graphics multimedia
MASTER_SITES= http://comstyle.com/source/
EXTRACT_SUFX= .tar.xz

28
graphics/ffmpeg/patches/patch-libavcodec_aaccoder_twoloop_h
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@ -1,4 +1,4 @@
$OpenBSD: patch-libavcodec_aaccoder_twoloop_h,v 1.3 2015/12/06 08:51:46 ajacoutot Exp $
$OpenBSD: patch-libavcodec_aaccoder_twoloop_h,v 1.4 2016/04/07 06:01:19 ajacoutot Exp $
AAC encoder: refactor to resynchronize MIPS port
@ -12,9 +12,15 @@ avcodec/aac_tablegen: get rid of hardcoded tables entirely
AAC encoder: improve SF range utilization
--- libavcodec/aaccoder_twoloop.h.orig Sat Dec 5 15:13:02 2015
+++ libavcodec/aaccoder_twoloop.h Sat Dec 5 15:01:19 2015
@@ -0,0 +1,755 @@
AAC encoder: fix signed integer overflow
AAC encoder: fix undefined behavior
AAC encoder: fix initialization of minsf
--- libavcodec/aaccoder_twoloop.h.orig Sun Apr 3 18:33:06 2016
+++ libavcodec/aaccoder_twoloop.h Sun Apr 3 18:33:32 2016
@@ -0,0 +1,763 @@
+/*
+ * AAC encoder twoloop coder
+ * Copyright (C) 2008-2009 Konstantin Shishkov
@ -94,7 +100,7 @@ AAC encoder: improve SF range utilization
+ int toomanybits, toofewbits;
+ char nzs[128];
+ uint8_t nextband[128];
+ int maxsf[128];
+ int maxsf[128], minsf[128];
+ float dists[128] = { 0 }, qenergies[128] = { 0 }, uplims[128], euplims[128], energies[128];
+ float maxvals[128], spread_thr_r[128];
+ float min_spread_thr_r, max_spread_thr_r;
@ -311,11 +317,19 @@ AAC encoder: improve SF range utilization
+ abs_pow34_v(s->scoefs, sce->coeffs, 1024);
+ ff_quantize_band_cost_cache_init(s);
+
+ for (i = 0; i < sizeof(minsf) / sizeof(minsf[0]); ++i)
+ minsf[i] = 0;
+ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+ start = w*128;
+ for (g = 0; g < sce->ics.num_swb; g++) {
+ const float *scaled = s->scoefs + start;
+ int minsfidx;
+ maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled);
+ if (maxvals[w*16+g] > 0) {
+ minsfidx = coef2minsf(maxvals[w*16+g]);
+ for (w2 = 0; w2 < sce->ics.group_len[w]; w2++)
+ minsf[(w+w2)*16+g] = minsfidx;
+ }
+ start += sce->ics.swb_sizes[g];
+ }
+ }
@ -442,7 +456,7 @@ AAC encoder: improve SF range utilization
+ recomprd = 1;
+ for (i = 0; i < 128; i++) {
+ if (sce->sf_idx[i] > SCALE_ONE_POS) {
+ int new_sf = FFMAX(SCALE_ONE_POS, sce->sf_idx[i] - qstep);
+ int new_sf = FFMAX3(minsf[i], SCALE_ONE_POS, sce->sf_idx[i] - qstep);
+ if (new_sf != sce->sf_idx[i]) {
+ sce->sf_idx[i] = new_sf;
+ changed = 1;
@ -612,7 +626,7 @@ AAC encoder: improve SF range utilization
+ int cmb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
+ int mindeltasf = FFMAX(0, prev - SCALE_MAX_DIFF);
+ int maxdeltasf = FFMIN(SCALE_MAX_POS - SCALE_DIV_512, prev + SCALE_MAX_DIFF);
+ if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > mindeltasf) {
+ if ((!cmb || dists[w*16+g] > uplims[w*16+g]) && sce->sf_idx[w*16+g] > FFMAX(mindeltasf, minsf[w*16+g])) {
+ /* Try to make sure there is some energy in every nonzero band
+ * NOTE: This algorithm must be forcibly imbalanced, pushing harder
+ * on holes or more distorted bands at first, otherwise there's

68
graphics/ffmpeg/patches/patch-libavcodec_aacenc_c
View File

@ -1,4 +1,4 @@
$OpenBSD: patch-libavcodec_aacenc_c,v 1.14 2016/03/09 17:29:36 ajacoutot Exp $
$OpenBSD: patch-libavcodec_aacenc_c,v 1.15 2016/04/07 06:01:19 ajacoutot Exp $
aacenc: copy PRNG from the decoder
@ -76,8 +76,10 @@ aacenc: temporarily disable Mid/Side coding with multichannel files
aacenc: use generational cache instead of resetting.
AAC encoder: fix valgrind errors
--- libavcodec/aacenc.c.orig Wed Jan 13 15:27:48 2016
+++ libavcodec/aacenc.c Tue Mar 8 19:23:12 2016
+++ libavcodec/aacenc.c Wed Apr 6 18:35:10 2016
@@ -29,6 +29,7 @@
* add sane pulse detection
***********************************/
@ -169,7 +171,7 @@ aacenc: use generational cache instead of resetting.
int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
int chan_el_counter[4];
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
@@ -517,10 +540,12 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -517,19 +540,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
for (ch = 0; ch < chans; ch++) {
@ -185,7 +187,18 @@ aacenc: use generational cache instead of resetting.
samples2 = overlap + 1024;
la = samples2 + (448+64);
if (!frame)
@@ -537,7 +562,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
la = NULL;
if (tag == TYPE_LFE) {
- wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
+ wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
wi[ch].window_shape = 0;
wi[ch].num_windows = 1;
wi[ch].grouping[0] = 1;
+ wi[ch].clipping[0] = 0;
/* Only the lowest 12 coefficients are used in a LFE channel.
* The expression below results in only the bottom 8 coefficients
@@ -537,7 +563,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
*/
ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
} else {
@ -194,7 +207,32 @@ aacenc: use generational cache instead of resetting.
ics->window_sequence[0]);
}
ics->window_sequence[1] = ics->window_sequence[0];
@@ -571,25 +596,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -554,10 +580,23 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
ff_tns_max_bands_128 [s->samplerate_index]:
ff_tns_max_bands_1024[s->samplerate_index];
- clip_avoidance_factor = 0.0f;
+
for (w = 0; w < ics->num_windows; w++)
ics->group_len[w] = wi[ch].grouping[w];
+
+ /* Calculate input sample maximums and evaluate clipping risk */
+ clip_avoidance_factor = 0.0f;
for (w = 0; w < ics->num_windows; w++) {
+ const float *wbuf = overlap + w * 128;
+ const int wlen = 2048 / ics->num_windows;
+ float max = 0;
+ int j;
+ /* mdct input is 2 * output */
+ for (j = 0; j < wlen; j++)
+ max = FFMAX(max, fabsf(wbuf[j]));
+ wi[ch].clipping[w] = max;
+ }
+ for (w = 0; w < ics->num_windows; w++) {
if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
ics->window_clipping[w] = 1;
clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
@@ -571,25 +610,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
ics->clip_avoidance_factor = 1.0f;
}
@ -236,7 +274,7 @@ aacenc: use generational cache instead of resetting.
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for (i = 0; i < s->chan_map[0]; i++) {
FFPsyWindowInfo* wi = windows + start_ch;
@@ -606,15 +640,28 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -606,15 +654,28 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
sce = &cpe->ch[ch];
coeffs[ch] = sce->coeffs;
sce->ics.predictor_present = 0;
@ -266,7 +304,7 @@ aacenc: use generational cache instead of resetting.
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
}
if (chans > 1
@@ -632,14 +679,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -632,14 +693,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
@ -283,7 +321,7 @@ aacenc: use generational cache instead of resetting.
}
s->cur_channel = start_ch;
if (s->options.intensity_stereo) { /* Intensity Stereo */
@@ -656,8 +703,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -656,8 +717,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
s->coder->search_for_pred(s, sce);
if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
}
@ -294,7 +332,7 @@ aacenc: use generational cache instead of resetting.
for (ch = 0; ch < chans; ch++) {
sce = &cpe->ch[ch];
s->cur_channel = start_ch + ch;
@@ -666,22 +713,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -666,22 +727,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
}
s->cur_channel = start_ch;
}
@ -333,7 +371,7 @@ aacenc: use generational cache instead of resetting.
encode_ms_info(&s->pb, cpe);
if (cpe->ms_mode) ms_mode = 1;
}
@@ -693,36 +752,78 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
@@ -693,36 +766,78 @@ static int aac_encode_frame(AVCodecContext *avctx, AVP
start_ch += chans;
}
@ -432,7 +470,7 @@ aacenc: use generational cache instead of resetting.
if (!frame)
s->last_frame++;
@@ -738,6 +839,8 @@ static av_cold int aac_encode_end(AVCodecContext *avct
@@ -738,6 +853,8 @@ static av_cold int aac_encode_end(AVCodecContext *avct
{
AACEncContext *s = avctx->priv_data;
@ -441,7 +479,7 @@ aacenc: use generational cache instead of resetting.
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
@@ -796,76 +899,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
@@ -796,76 +913,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
uint8_t grouping[AAC_MAX_CHANNELS];
int lengths[2];
@ -599,7 +637,7 @@ aacenc: use generational cache instead of resetting.
ff_af_queue_init(avctx, &s->afq);
return 0;
@@ -876,27 +1026,16 @@ fail:
@@ -876,27 +1040,16 @@ fail:
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
static const AVOption aacenc_options[] = {
@ -637,7 +675,7 @@ aacenc: use generational cache instead of resetting.
{NULL}
};
@@ -907,6 +1046,11 @@ static const AVClass aacenc_class = {
@@ -907,6 +1060,11 @@ static const AVClass aacenc_class = {
LIBAVUTIL_VERSION_INT,
};
@ -649,7 +687,7 @@ aacenc: use generational cache instead of resetting.
AVCodec ff_aac_encoder = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -916,9 +1060,9 @@ AVCodec ff_aac_encoder = {
@@ -916,9 +1074,9 @@ AVCodec ff_aac_encoder = {
.init = aac_encode_init,
.encode2 = aac_encode_frame,
.close = aac_encode_end,

25
graphics/ffmpeg/patches/patch-libavcodec_aacenc_utils_h
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@ -1,4 +1,4 @@
$OpenBSD: patch-libavcodec_aacenc_utils_h,v 1.9 2016/03/09 17:29:36 ajacoutot Exp $
$OpenBSD: patch-libavcodec_aacenc_utils_h,v 1.10 2016/04/07 06:01:19 ajacoutot Exp $
aacenc: copy PRNG from the decoder
@ -28,11 +28,15 @@ aacenc: avoid double in quantize_bands.
aacenc_utils: Use temporary variable.
lavc/aacenc_utils: replace powf(x,y) by expf(logf(x), y)
--- libavcodec/aacenc_utils.h.orig Wed Jan 13 15:27:48 2016
+++ libavcodec/aacenc_utils.h Tue Mar 8 19:24:00 2016
@@ -29,8 +29,8 @@
+++ libavcodec/aacenc_utils.h Fri Mar 18 20:59:09 2016
@@ -28,9 +28,10 @@
#ifndef AVCODEC_AACENC_UTILS_H
#define AVCODEC_AACENC_UTILS_H
+#include "libavutil/internal.h"
#include "aac.h"
-#include "aac_tablegen_decl.h"
#include "aacenctab.h"
@ -40,7 +44,7 @@ aacenc_utils: Use temporary variable.
#define ROUND_STANDARD 0.4054f
#define ROUND_TO_ZERO 0.1054f
@@ -45,6 +45,11 @@ static inline void abs_pow34_v(float *out, const float
@@ -45,6 +46,11 @@ static inline void abs_pow34_v(float *out, const float
}
}
@ -52,7 +56,7 @@ aacenc_utils: Use temporary variable.
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
@@ -61,13 +66,13 @@ static inline void quantize_bands(int *out, const floa
@@ -61,13 +67,13 @@ static inline void quantize_bands(int *out, const floa
const float rounding)
{
int i;
@ -70,7 +74,7 @@ aacenc_utils: Use temporary variable.
}
}
@@ -85,20 +90,65 @@ static inline float find_max_val(int group_len, int sw
@@ -85,20 +91,68 @@ static inline float find_max_val(int group_len, int sw
static inline int find_min_book(float maxval, int sf)
{
@ -115,7 +119,10 @@ aacenc_utils: Use temporary variable.
+ if (s >= ethresh) {
+ nzl += 1.0f;
+ } else {
+ nzl += powf(s / ethresh, nzslope);
+ if (nzslope == 2.f)
+ nzl += (s / ethresh) * (s / ethresh);
+ else
+ nzl += ff_fast_powf(s / ethresh, nzslope);
+ }
+ }
+ if (e2 > thresh) {
@ -145,7 +152,7 @@ aacenc_utils: Use temporary variable.
/** Return the minimum scalefactor where the quantized coef does not clip. */
static inline uint8_t coef2minsf(float coef)
{
@@ -128,6 +178,76 @@ static inline int quant_array_idx(const float val, con
@@ -128,6 +182,76 @@ static inline int quant_array_idx(const float val, con
return index;
}
@ -222,7 +229,7 @@ aacenc_utils: Use temporary variable.
#define ERROR_IF(cond, ...) \
if (cond) { \
av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
@@ -138,6 +258,5 @@ static inline int quant_array_idx(const float val, con
@@ -138,6 +262,5 @@ static inline int quant_array_idx(const float val, con
if (cond) { \
av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
}

99
graphics/ffmpeg/patches/patch-libavcodec_aacpsy_c
View File

@ -1,4 +1,4 @@
$OpenBSD: patch-libavcodec_aacpsy_c,v 1.5 2015/12/06 08:51:46 ajacoutot Exp $
$OpenBSD: patch-libavcodec_aacpsy_c,v 1.6 2016/04/07 06:01:19 ajacoutot Exp $
AAC encoder: tweak rate-distortion logic
@ -10,9 +10,21 @@ AAC encoder: make pe.min a local minimum
AAC encoder: improve SF range utilization
--- libavcodec/aacpsy.c.orig Sat Dec 5 15:14:00 2015
+++ libavcodec/aacpsy.c Sat Dec 5 15:01:19 2015
@@ -80,6 +80,8 @@
AAC encoder: fix filling of wi.clipping array
AAC encoder: fix valgrind errors
--- libavcodec/aacpsy.c.orig Wed Jan 13 15:27:48 2016
+++ libavcodec/aacpsy.c Wed Apr 6 18:31:01 2016
@@ -25,6 +25,7 @@
*/
#include "libavutil/attributes.h"
+#include "libavutil/internal.h"
#include "libavutil/libm.h"
#include "avcodec.h"
@@ -80,6 +81,8 @@
#define PSY_3GPP_AH_THR_LONG 0.5f
#define PSY_3GPP_AH_THR_SHORT 0.63f
@ -21,7 +33,7 @@ AAC encoder: improve SF range utilization
enum {
PSY_3GPP_AH_NONE,
PSY_3GPP_AH_INACTIVE,
@@ -87,6 +89,7 @@ enum {
@@ -87,6 +90,7 @@ enum {
};
#define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
@ -29,7 +41,7 @@ AAC encoder: improve SF range utilization
/* LAME psy model constants */
#define PSY_LAME_FIR_LEN 21 ///< LAME psy model FIR order
@@ -157,6 +160,7 @@ typedef struct AacPsyContext{
@@ -157,6 +161,7 @@ typedef struct AacPsyContext{
} pe;
AacPsyCoeffs psy_coef[2][64];
AacPsyChannel *ch;
@ -37,7 +49,7 @@ AAC encoder: improve SF range utilization
}AacPsyContext;
/**
@@ -299,17 +303,24 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
@@ -299,17 +304,24 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
float bark;
int i, j, g, start;
float prev, minscale, minath, minsnr, pe_min;
@ -65,7 +77,22 @@ AAC encoder: improve SF range utilization
pctx->pe.min = 8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
pctx->pe.max = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
ctx->bitres.size = 6144 - pctx->frame_bits;
@@ -397,7 +408,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsy
@@ -338,10 +350,10 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
for (g = 0; g < ctx->num_bands[j] - 1; g++) {
AacPsyCoeffs *coeff = &coeffs[g];
float bark_width = coeffs[g+1].barks - coeffs->barks;
- coeff->spread_low[0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_LOW);
- coeff->spread_hi [0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_HI);
- coeff->spread_low[1] = pow(10.0, -bark_width * en_spread_low);
- coeff->spread_hi [1] = pow(10.0, -bark_width * en_spread_hi);
+ coeff->spread_low[0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_LOW);
+ coeff->spread_hi [0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_HI);
+ coeff->spread_low[1] = ff_exp10(-bark_width * en_spread_low);
+ coeff->spread_hi [1] = ff_exp10(-bark_width * en_spread_hi);
pe_min = bark_pe * bark_width;
minsnr = exp2(pe_min / band_sizes[g]) - 1.5f;
coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
@@ -397,7 +409,7 @@ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsy
int channel, int prev_type)
{
int i, j;
@ -74,7 +101,7 @@ AAC encoder: improve SF range utilization
int attack_ratio = br <= 16000 ? 18 : 10;
AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
AacPsyChannel *pch = &pctx->ch[channel];
@@ -486,7 +497,7 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
@@ -486,7 +498,7 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
const float bitspend_add = short_window ? PSY_3GPP_SPEND_ADD_S : PSY_3GPP_SPEND_ADD_L;
const float clip_low = short_window ? PSY_3GPP_CLIP_LO_S : PSY_3GPP_CLIP_LO_L;
const float clip_high = short_window ? PSY_3GPP_CLIP_HI_S : PSY_3GPP_CLIP_HI_L;
@ -83,7 +110,7 @@ AAC encoder: improve SF range utilization
ctx->fill_level += ctx->frame_bits - bits;
ctx->fill_level = av_clip(ctx->fill_level, 0, size);
@@ -503,11 +514,21 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
@@ -503,11 +515,21 @@ static int calc_bit_demand(AacPsyContext *ctx, float p
* Hopefully below is correct.
*/
bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min);
@ -108,7 +135,7 @@ AAC encoder: improve SF range utilization
}
static float calc_pe_3gpp(AacPsyBand *band)
@@ -574,26 +595,30 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, f
@@ -574,26 +596,30 @@ static float calc_reduced_thr_3gpp(AacPsyBand *band, f
#ifndef calc_thr_3gpp
static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
@ -144,7 +171,7 @@ AAC encoder: improve SF range utilization
}
}
}
@@ -634,9 +659,11 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
@@ -634,9 +660,11 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8];
AacPsyCoeffs *coeffs = pctx->psy_coef[wi->num_windows == 8];
const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
@ -157,7 +184,16 @@ AAC encoder: improve SF range utilization
//modify thresholds and energies - spread, threshold in quiet, pre-echo control
for (w = 0; w < wi->num_windows*16; w += 16) {
@@ -677,16 +704,36 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
@@ -658,7 +686,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath);
//5.4.2.5 "Pre-echo control"
- if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))
+ if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (!w && wi->window_type[1] == LONG_START_SEQUENCE)))
band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
@@ -677,16 +705,36 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
/* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */
ctx->ch[channel].entropy = pe;
@ -203,7 +239,7 @@ AAC encoder: improve SF range utilization
if (desired_pe < pe) {
/* 5.6.1.3.4 "First Estimation of the reduction value" */
@@ -788,6 +835,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
@@ -788,6 +836,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx
psy_band->threshold = band->thr;
psy_band->energy = band->energy;
psy_band->spread = band->active_lines * 2.0f / band_sizes[g];
@ -211,3 +247,38 @@ AAC encoder: improve SF range utilization
}
}
@@ -927,21 +976,6 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *c
lame_apply_block_type(pch, &wi, uselongblock);
- /* Calculate input sample maximums and evaluate clipping risk */
- if (audio) {
- for (i = 0; i < AAC_NUM_BLOCKS_SHORT; i++) {
- const float *wbuf = audio + i * AAC_BLOCK_SIZE_SHORT;
- float max = 0;
- int j;
- for (j = 0; j < AAC_BLOCK_SIZE_SHORT; j++)
- max = FFMAX(max, fabsf(wbuf[j]));
- clippings[i] = max;
- }
- } else {
- for (i = 0; i < 8; i++)
- clippings[i] = 0;
- }
-
wi.window_type[1] = prev_type;
if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
float clipping = 0.0f;
@@ -970,9 +1004,10 @@ static FFPsyWindowInfo psy_lame_window(FFPsyContext *c
for (i = 0; i < 8; i += wi.grouping[i]) {
int w;
float clipping = 0.0f;
- for (w = 0; w < wi.grouping[i] && !clipping; w++)
+ for (w = 0; w < wi.grouping[i]; w++)
clipping = FFMAX(clipping, clippings[i+w]);
- wi.clipping[i] = clipping;
+ for (w = 0; w < wi.grouping[i]; w++)
+ wi.clipping[i+w] = clipping;
}
}

13
graphics/ffmpeg/patches/patch-libavcodec_lpc_c
View File

@ -1,10 +1,12 @@
$OpenBSD: patch-libavcodec_lpc_c,v 1.1 2015/10/13 05:44:18 ajacoutot Exp $
$OpenBSD: patch-libavcodec_lpc_c,v 1.2 2016/04/07 06:01:19 ajacoutot Exp $
lpc: correctly apply windowing to the samples in the float-only lpc
--- libavcodec/lpc.c.orig Mon Oct 12 23:03:10 2015
+++ libavcodec/lpc.c Mon Oct 12 23:03:21 2015
@@ -173,11 +173,13 @@ double ff_lpc_calc_ref_coefs_f(LPCContext *s, const fl
lavc/lpc: exploit even symmetry of window function
--- libavcodec/lpc.c.orig Wed Jan 13 15:27:48 2016
+++ libavcodec/lpc.c Thu Mar 24 20:27:56 2016
@@ -173,11 +173,14 @@ double ff_lpc_calc_ref_coefs_f(LPCContext *s, const fl
int i;
double signal = 0.0f, avg_err = 0.0f;
double autoc[MAX_LPC_ORDER+1] = {0}, error[MAX_LPC_ORDER+1] = {0};
@ -15,9 +17,10 @@ lpc: correctly apply windowing to the samples in the float-only lpc
- for (i = 0; i < len; i++)
- s->windowed_samples[i] = 1.0f - ((samples[i]-c)/c)*((samples[i]-c)/c);
+ /* Apply windowing */
+ for (i = 0; i < len; i++) {
+ for (i = 0; i <= len / 2; i++) {
+ double weight = a - b*cos((2*M_PI*i)/(len - 1));
+ s->windowed_samples[i] = weight*samples[i];
+ s->windowed_samples[len-1-i] = weight*samples[len-1-i];
+ }
s->lpc_compute_autocorr(s->windowed_samples, len, order, autoc);

51
graphics/ffmpeg/patches/patch-libavutil_internal_h Normal file
View File

@ -0,0 +1,51 @@
$OpenBSD: patch-libavutil_internal_h,v 1.1 2016/04/07 06:01:19 ajacoutot Exp $
lavu/internal: add ff_exp10
lavc/aacenc_utils: replace powf(x,y) by expf(logf(x), y)
--- libavutil/internal.h.orig Wed Jan 13 15:27:50 2016
+++ libavutil/internal.h Wed Apr 6 19:54:16 2016
@@ -257,6 +257,42 @@ void avpriv_request_sample(void *avc,
#endif
/**
+ * Compute 10^x for floating point values. Note: this function is by no means
+ * "correctly rounded", and is meant as a fast, reasonably accurate approximation.
+ * For instance, maximum relative error for the double precision variant is
+ * ~ 1e-13 for very small and very large values.
+ * This is ~2x faster than GNU libm's approach, which is still off by 2ulp on
+ * some inputs.
+ * @param x exponent
+ * @return 10^x
+ */
+static av_always_inline double ff_exp10(double x)
+{
+ return exp2(M_LOG2_10 * x);
+}
+
+static av_always_inline float ff_exp10f(float x)
+{
+ return exp2f(M_LOG2_10 * x);
+}
+
+/**
+ * Compute x^y for floating point x, y. Note: this function is faster than the
+ * libm variant due to mainly 2 reasons:
+ * 1. It does not handle any edge cases. In particular, this is only guaranteed
+ * to work correctly for x > 0.
+ * 2. It is not as accurate as a standard nearly "correctly rounded" libm variant.
+ * @param x base
+ * @param y exponent
+ * @return x^y
+ */
+static av_always_inline float ff_fast_powf(float x, float y)
+{
+ return expf(logf(x) * y);
+}
+
+
+/**
* A wrapper for open() setting O_CLOEXEC.
*/
int avpriv_open(const char *filename, int flags, ...);

14
graphics/ffmpeg/patches/patch-libavutil_libm_h Normal file
View File

@ -0,0 +1,14 @@
$OpenBSD: patch-libavutil_libm_h,v 1.1 2016/04/07 06:01:19 ajacoutot Exp $
lavu/libm: add exp10 support
--- libavutil/libm.h.orig Wed Apr 6 20:45:13 2016
+++ libavutil/libm.h Wed Apr 6 20:45:27 2016
@@ -28,6 +28,7 @@
#include "config.h"
#include "attributes.h"
#include "intfloat.h"
+#include "mathematics.h"
#if HAVE_MIPSFPU && HAVE_INLINE_ASM
#include "libavutil/mips/libm_mips.h"

19
graphics/ffmpeg/patches/patch-tests_fate_aac_mak
View File

@ -1,4 +1,4 @@
$OpenBSD: patch-tests_fate_aac_mak,v 1.9 2016/03/07 07:11:35 ajacoutot Exp $
$OpenBSD: patch-tests_fate_aac_mak,v 1.10 2016/04/07 06:01:19 ajacoutot Exp $
fate: adjust AAC encoder tests values
@ -70,9 +70,11 @@ tests/fate/aac: remove unneeded strict arguments from the encoder tests
fate/aac: Increase fuzz from of fate-aac-pns-encode from 72 to 74 for Loongson
AAC encoder: new regression test
--- tests/fate/aac.mak.orig Wed Jan 13 15:27:50 2016
+++ tests/fate/aac.mak Sun Mar 6 19:41:20 2016
@@ -142,56 +142,84 @@ FATE_AAC += $(FATE_AAC_CT:%=fate-aac-ct-%)
+++ tests/fate/aac.mak Sun Apr 3 18:34:00 2016
@@ -142,56 +142,95 @@ FATE_AAC += $(FATE_AAC_CT:%=fate-aac-ct-%)
FATE_AAC_ENCODE += fate-aac-aref-encode
fate-aac-aref-encode: ./tests/data/asynth-44100-2.wav
@ -160,6 +162,17 @@ fate/aac: Increase fuzz from of fate-aac-pns-encode from 72 to 74 for Loongson
+fate-aac-ltp-encode: CMP_TARGET = 1270
+fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
+fate-aac-ltp-encode: FUZZ = 17
+
+#Ticket1784
+FATE_AAC_ENCODE += fate-aac-yoraw-encode
+fate-aac-yoraw-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/yo.raw-short.wav -c:a aac -fflags +bitexact -flags +bitexact
+fate-aac-yoraw-encode: CMP = stddev
+fate-aac-yoraw-encode: REF = $(SAMPLES)/audio-reference/yo.raw-short.wav
+fate-aac-yoraw-encode: CMP_SHIFT = -12288
+fate-aac-yoraw-encode: CMP_TARGET = 259
+fate-aac-yoraw-encode: SIZE_TOLERANCE = 3560
+fate-aac-yoraw-encode: FUZZ = 17
+
+
FATE_AAC_ENCODE += fate-aac-pred-encode
-fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav -strict -2 -profile:a aac_main -c:a aac -aac_is 0 -aac_pns 0 -b:a 128k