RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity

Merge remote branch 'qatar/master' 

* qatar/master:
  vorbisdec: Rename silly "class_" variable to plain "class".
  simple_idct_alpha: Drop some useless casts.
  Simplify av_log_missing_feature().
  ac3enc: remove check for mismatching channels and channel_layout
  If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout.
  If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother.
  cosmetics: indentation
  Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open().
  aacdec: remove sf_scale and sf_offset.
  aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec.
  Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere.
  Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway.
  qpeg: use reget_buffer() in decode_frame()
  ultimotion: use reget_buffer() in ulti_decode_frame()
  smacker: remove unnecessary call to avctx->release_buffer in decode_frame()
  avparser: don't av_malloc(0).

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2011-04-28 04:23:36 +02:00
parent e5d80c7b2d b239526873
commit 0665199e43
12 changed files with 96 additions and 78 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

3
libavcodec/aac.h
View File

@ -130,6 +130,7 @@ typedef struct {
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
/**
* Long Term Prediction
@ -292,8 +293,6 @@ typedef struct {
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];

3
libavcodec/aac_tablegen.h
View File

@ -29,13 +29,14 @@
#include "libavcodec/aac_tables.h"
#else
#include "libavutil/mathematics.h"
#include "libavcodec/aac.h"
float ff_aac_pow2sf_tab[428];
void ff_aac_tableinit(void)
{
int i;
for (i = 0; i < 428; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
}
#endif /* CONFIG_HARDCODED_TABLES */

8
libavcodec/aaccoder.c
View File

@ -109,8 +109,8 @@ static av_always_inline float quantize_and_encode_band_cost_template(
int *bits, int BT_ZERO, int BT_UNSIGNED,
int BT_PAIR, int BT_ESC)
{
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float IQ = ff_aac_pow2sf_tab[POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
const float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
const float CLIPPED_ESCAPE = 165140.0f*IQ;
int i, j;
float cost = 0;
@ -281,7 +281,7 @@ static float find_max_val(int group_len, int swb_size, const float *scaled) {
}
static int find_min_book(float maxval, int sf) {
float Q = ff_aac_pow2sf_tab[200 - sf + SCALE_ONE_POS - SCALE_DIV_512];
float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
float Q34 = sqrtf(Q * sqrtf(Q));
int qmaxval, cb;
qmaxval = maxval * Q34 + 0.4054f;
@ -956,7 +956,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
dist -= b;
}
dist *= 1.0f / 512.0f / lambda;
quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[200 - scf + SCALE_ONE_POS - SCALE_DIV_512]);
quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
if (quant_max >= 8191) { // too much, return to the previous quantizer
sce->sf_idx[w*16+g] = prev_scf;
break;

51
libavcodec/aacdec.c
View File

@ -579,12 +579,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ac->random_state = 0x1f2e3d4c;
// -1024 - Compensate wrong IMDCT method.
// 60 - Required to scale values to the correct range [-32768,32767]
// for float to int16 conversion. (1 << (60 / 4)) == 32768
ac->sf_scale = 1. / -1024.;
ac->sf_offset = 60;
ff_aac_tableinit();
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
@ -592,9 +586,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
352);
ff_mdct_init(&ac->mdct, 11, 1, 1.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, 1.0);
ff_mdct_init(&ac->mdct, 11, 1, 1.0/1024.0);
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0/128.0);
ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0);
// window initialization
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
@ -652,7 +646,7 @@ static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
int sfb;
ltp->lag = get_bits(gb, 11);
ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
ltp->coef = ltp_coef[get_bits(gb, 3)];
for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
ltp->used[sfb] = get_bits1(gb);
}
@ -790,9 +784,9 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
enum BandType band_type[120],
int band_type_run_end[120])
{
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
int g, i, idx = 0;
int offset[3] = { global_gain, global_gain - 90, 100 };
int offset[3] = { global_gain, global_gain - 90, 0 };
int clipped_offset;
int noise_flag = 1;
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
for (g = 0; g < ics->num_window_groups; g++) {
@ -804,12 +798,14 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
for (; i < run_end; i++, idx++) {
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[2] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[2], offset[2]);
return -1;
clipped_offset = av_clip(offset[2], -155, 100);
if (offset[2] != clipped_offset) {
av_log_ask_for_sample(ac->avctx, "Intensity stereo "
"position clipped (%d -> %d).\nIf you heard an "
"audible artifact, there may be a bug in the "
"decoder. ", offset[2], clipped_offset);
}
sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
}
} else if (band_type[idx] == NOISE_BT) {
for (; i < run_end; i++, idx++) {
@ -817,12 +813,14 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
offset[1] += get_bits(gb, 9) - 256;
else
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
if (offset[1] > 255U) {
av_log(ac->avctx, AV_LOG_ERROR,
"%s (%d) out of range.\n", sf_str[1], offset[1]);
return -1;
clipped_offset = av_clip(offset[1], -100, 155);
if (offset[2] != clipped_offset) {
av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
"(%d -> %d).\nIf you heard an audible "
"artifact, there may be a bug in the decoder. ",
offset[1], clipped_offset);
}
sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
}
} else {
for (; i < run_end; i++, idx++) {
@ -832,7 +830,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
"%s (%d) out of range.\n", sf_str[0], offset[0]);
return -1;
}
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
}
}
}
@ -1243,7 +1241,6 @@ static av_always_inline float flt16_trunc(float pf)
}
static av_always_inline void predict(PredictorState *ps, float *coef,
float sf_scale, float inv_sf_scale,
int output_enable)
{
const float a = 0.953125; // 61.0 / 64
@ -1260,9 +1257,9 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
pv = flt16_round(k1 * r0 + k2 * r1);
if (output_enable)
*coef += pv * sf_scale;
*coef += pv;
e0 = *coef * inv_sf_scale;
e0 = *coef;
e1 = e0 - k1 * r0;
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
@ -1280,7 +1277,6 @@ static av_always_inline void predict(PredictorState *ps, float *coef,
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
{
int sfb, k;
float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
if (!sce->ics.predictor_initialized) {
reset_all_predictors(sce->predictor_state);
@ -1291,7 +1287,6 @@ static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
predict(&sce->predictor_state[k], &sce->coeffs[k],
sf_scale, inv_sf_scale,
sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
}
}

6
libavcodec/aacdectab.h
View File

@ -36,11 +36,11 @@
#include <stdint.h>
/* @name ltp_coef
* Table of the LTP coefficient (multiplied by 2)
* Table of the LTP coefficients
*/
static const float ltp_coef[8] = {
1.141658, 1.393232, 1.626008, 1.822608,
1.969800, 2.135788, 2.2389202, 2.739066,
0.570829, 0.696616, 0.813004, 0.911304,
0.984900, 1.067894, 1.194601, 1.369533,
};
/* @name tns_tmp2_map

2
libavcodec/ac3enc.c
View File

@ -1962,8 +1962,6 @@ static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
ch_layout = *channel_layout;
if (!ch_layout)
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
if (av_get_channel_layout_nb_channels(ch_layout) != channels)
return AVERROR(EINVAL);
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
s->channels = channels;

14
libavcodec/alpha/simple_idct_alpha.c
View File

@ -33,13 +33,13 @@
// cos(i * M_PI / 16) * sqrt(2) * (1 << 14)
// W4 is actually exactly 16384, but using 16383 works around
// accumulating rounding errors for some encoders
#define W1 ((int_fast32_t) 22725)
#define W2 ((int_fast32_t) 21407)
#define W3 ((int_fast32_t) 19266)
#define W4 ((int_fast32_t) 16383)
#define W5 ((int_fast32_t) 12873)
#define W6 ((int_fast32_t) 8867)
#define W7 ((int_fast32_t) 4520)
#define W1 22725
#define W2 21407
#define W3 19266
#define W4 16383
#define W5 12873
#define W6 8867
#define W7 4520
#define ROW_SHIFT 11
#define COL_SHIFT 20

9
libavcodec/qpeg.c
View File

@ -259,12 +259,9 @@ static int decode_frame(AVCodecContext *avctx,
int delta;
const uint8_t *pal = av_packet_get_side_data(avpkt, AV_PKT_DATA_PALETTE, NULL);
if(p->data[0])
avctx->release_buffer(avctx, p);
p->reference= 0;
if(avctx->get_buffer(avctx, p) < 0){
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
p->reference = 3;
if (avctx->reget_buffer(avctx, p) < 0) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}
outdata = a->pic.data[0];

2
libavcodec/smacker.c
View File

@ -360,8 +360,6 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPac
if(buf_size <= 769)
return 0;
if(smk->pic.data[0])
avctx->release_buffer(avctx, &smk->pic);
smk->pic.reference = 1;
smk->pic.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;

7
libavcodec/ulti.c
View File

@ -224,13 +224,10 @@ static int ulti_decode_frame(AVCodecContext *avctx,
int skip;
int tmp;
if(s->frame.data[0])
avctx->release_buffer(avctx, &s->frame);
s->frame.reference = 1;
s->frame.buffer_hints = FF_BUFFER_HINTS_VALID | FF_BUFFER_HINTS_PRESERVE | FF_BUFFER_HINTS_REUSABLE;
if(avctx->get_buffer(avctx, &s->frame) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
if (avctx->reget_buffer(avctx, &s->frame) < 0) {
av_log(avctx, AV_LOG_ERROR, "reget_buffer() failed\n");
return -1;
}

55
libavcodec/utils.c
View File

@ -555,15 +555,50 @@ int attribute_align_arg avcodec_open(AVCodecContext *avctx, AVCodec *codec)
ret = AVERROR(EINVAL);
goto free_and_end;
}
if (avctx->codec->sample_fmts && avctx->codec->encode) {
if (avctx->codec->encode) {
int i;
for (i = 0; avctx->codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++)
if (avctx->sample_fmt == avctx->codec->sample_fmts[i])
break;
if (avctx->codec->sample_fmts[i] == AV_SAMPLE_FMT_NONE) {
av_log(avctx, AV_LOG_ERROR, "Specified sample_fmt is not supported.\n");
ret = AVERROR(EINVAL);
goto free_and_end;
if (avctx->codec->sample_fmts) {
for (i = 0; avctx->codec->sample_fmts[i] != AV_SAMPLE_FMT_NONE; i++)
if (avctx->sample_fmt == avctx->codec->sample_fmts[i])
break;
if (avctx->codec->sample_fmts[i] == AV_SAMPLE_FMT_NONE) {
av_log(avctx, AV_LOG_ERROR, "Specified sample_fmt is not supported.\n");
ret = AVERROR(EINVAL);
goto free_and_end;
}
}
if (avctx->codec->supported_samplerates) {
for (i = 0; avctx->codec->supported_samplerates[i] != 0; i++)
if (avctx->sample_rate == avctx->codec->supported_samplerates[i])
break;
if (avctx->codec->supported_samplerates[i] == 0) {
av_log(avctx, AV_LOG_ERROR, "Specified sample_rate is not supported\n");
ret = AVERROR(EINVAL);
goto free_and_end;
}
}
if (avctx->codec->channel_layouts) {
if (!avctx->channel_layout) {
av_log(avctx, AV_LOG_WARNING, "channel_layout not specified\n");
} else {
for (i = 0; avctx->codec->channel_layouts[i] != 0; i++)
if (avctx->channel_layout == avctx->codec->channel_layouts[i])
break;
if (avctx->codec->channel_layouts[i] == 0) {
av_log(avctx, AV_LOG_ERROR, "Specified channel_layout is not supported\n");
ret = AVERROR(EINVAL);
goto free_and_end;
}
}
}
if (avctx->channel_layout && avctx->channels) {
if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "channel layout does not match number of channels\n");
ret = AVERROR(EINVAL);
goto free_and_end;
}
} else if (avctx->channel_layout) {
avctx->channels = av_get_channel_layout_nb_channels(avctx->channel_layout);
}
}
@ -1194,11 +1229,9 @@ void av_log_missing_feature(void *avc, const char *feature, int want_sample)
av_log(avc, AV_LOG_WARNING, "%s not implemented. Update your FFmpeg "
"version to the newest one from Git. If the problem still "
"occurs, it means that your file has a feature which has not "
"been implemented.", feature);
"been implemented.\n", feature);
if(want_sample)
av_log_ask_for_sample(avc, NULL);
else
av_log(avc, AV_LOG_WARNING, "\n");
}
void av_log_ask_for_sample(void *avc, const char *msg, ...)

14
libavcodec/vorbisdec.c
View File

@ -1138,7 +1138,7 @@ static int vorbis_floor1_decode(vorbis_context *vc,
uint_fast16_t floor1_Y[258];
uint_fast16_t floor1_Y_final[258];
int floor1_flag[258];
uint_fast8_t class_;
uint_fast8_t class;
uint_fast8_t cdim;
uint_fast8_t cbits;
uint_fast8_t csub;
@ -1162,20 +1162,20 @@ static int vorbis_floor1_decode(vorbis_context *vc,
offset = 2;
for (i = 0; i < vf->partitions; ++i) {
class_ = vf->partition_class[i];
cdim = vf->class_dimensions[class_];
cbits = vf->class_subclasses[class_];
class = vf->partition_class[i];
cdim = vf->class_dimensions[class];
cbits = vf->class_subclasses[class];
csub = (1 << cbits) - 1;
cval = 0;
AV_DEBUG("Cbits %d \n", cbits);
if (cbits) // this reads all subclasses for this partition's class
cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[class_]].vlc.table,
vc->codebooks[vf->class_masterbook[class_]].nb_bits, 3);
cval = get_vlc2(gb, vc->codebooks[vf->class_masterbook[class]].vlc.table,
vc->codebooks[vf->class_masterbook[class]].nb_bits, 3);
for (j = 0; j < cdim; ++j) {
book = vf->subclass_books[class_][cval & csub];
book = vf->subclass_books[class][cval & csub];
AV_DEBUG("book %d Cbits %d cval %d bits:%d \n", book, cbits, cval, get_bits_count(gb));