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>

libavcodec/aac.h

@@ -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];

libavcodec/aac_tablegen.h

@@ -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 */
 

libavcodec/aaccoder.c

@@ -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;

libavcodec/aacdec.c

@@ -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]);
             }
         }

libavcodec/aacdectab.h

@@ -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

libavcodec/ac3enc.c

@@ -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;

libavcodec/alpha/simple_idct_alpha.c

@@ -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
 

libavcodec/qpeg.c

@@ -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];

libavcodec/smacker.c

@@ -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;

libavcodec/ulti.c

@@ -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;
     }
 

libavcodec/utils.c

@@ -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, ...)

libavcodec/vorbisdec.c

@@ -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));