]> git.sesse.net Git - ffmpeg/blobdiff - libavcodec/ac3dec.c
Apply 'cold' attribute to init/uninit functions in libavcodec
[ffmpeg] / libavcodec / ac3dec.c
index c1b3ec231651c5afb818d56b863b22ee09409ad0..75208de33d4669b22b9f1d7cb789fde266a33782 100644 (file)
  */
 static const uint8_t rematrix_band_tab[5] = { 13, 25, 37, 61, 253 };
 
-/**
- * table for exponent to scale_factor mapping
- * scale_factors[i] = 2 ^ -i
- */
-static float scale_factors[25];
-
 /** table for grouping exponents */
 static uint8_t exp_ungroup_tab[128][3];
 
 
 /** tables for ungrouping mantissas */
-static float b1_mantissas[32][3];
-static float b2_mantissas[128][3];
-static float b3_mantissas[8];
-static float b4_mantissas[128][2];
-static float b5_mantissas[16];
+static int b1_mantissas[32][3];
+static int b2_mantissas[128][3];
+static int b3_mantissas[8];
+static int b4_mantissas[128][2];
+static int b5_mantissas[16];
 
 /**
  * Quantization table: levels for symmetric. bits for asymmetric.
@@ -135,7 +129,6 @@ typedef struct {
     int phase_flags_in_use;                 ///< phase flags in use
     int phase_flags[18];                    ///< phase flags
     int cpl_band_struct[18];                ///< coupling band structure
-    int rematrixing_strategy;               ///< rematrixing strategy
     int num_rematrixing_bands;              ///< number of rematrixing bands
     int rematrixing_flags[4];               ///< rematrixing flags
     int exp_strategy[AC3_MAX_CHANNELS];     ///< exponent strategies
@@ -158,9 +151,12 @@ typedef struct {
     int output_mode;                        ///< output channel configuration
     int out_channels;                       ///< number of output channels
 
+    int center_mix_level;                   ///< Center mix level index
+    int surround_mix_level;                 ///< Surround mix level index
     float downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
+    float downmix_coeff_adjust[2];          ///< adjustment needed for each output channel when downmixing
     float dynamic_range[2];                 ///< dynamic range
-    float cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
+    int   cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates
     int   num_cpl_bands;                    ///< number of coupling bands
     int   num_cpl_subbands;                 ///< number of coupling sub bands
     int   start_freq[AC3_MAX_CHANNELS];     ///< start frequency bin
@@ -173,7 +169,9 @@ typedef struct {
     int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
     int16_t mask[AC3_MAX_CHANNELS][50];     ///< masking curve values
 
+    int fixed_coeffs[AC3_MAX_CHANNELS][256];    ///> fixed-point transform coefficients
     DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][256]);  ///< transform coefficients
+    int downmixed;                              ///< indicates if coeffs are currently downmixed
 
     /* For IMDCT. */
     MDCTContext imdct_512;                  ///< for 512 sample IMDCT
@@ -182,9 +180,9 @@ typedef struct {
     float       add_bias;                   ///< offset for float_to_int16 conversion
     float       mul_bias;                   ///< scaling for float_to_int16 conversion
 
-    DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS-1][256]);     ///< output after imdct transform and windowing
+    DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][256]);       ///< output after imdct transform and windowing
     DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][256]); ///< final 16-bit integer output
-    DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS-1][256]);      ///< delay - added to the next block
+    DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][256]);        ///< delay - added to the next block
     DECLARE_ALIGNED_16(float, tmp_imdct[256]);                      ///< temporary storage for imdct transform
     DECLARE_ALIGNED_16(float, tmp_output[512]);                     ///< temporary storage for output before windowing
     DECLARE_ALIGNED_16(float, window[256]);                         ///< window coefficients
@@ -195,45 +193,21 @@ typedef struct {
     AVCodecContext *avctx;                  ///< parent context
 } AC3DecodeContext;
 
-/**
- * Generate a Kaiser-Bessel Derived Window.
- */
-static void ac3_window_init(float *window)
-{
-   int i, j;
-   double sum = 0.0, bessel, tmp;
-   double local_window[256];
-   double alpha2 = (5.0 * M_PI / 256.0) * (5.0 * M_PI / 256.0);
-
-   for (i = 0; i < 256; i++) {
-       tmp = i * (256 - i) * alpha2;
-       bessel = 1.0;
-       for (j = 100; j > 0; j--) /* default to 100 iterations */
-           bessel = bessel * tmp / (j * j) + 1;
-       sum += bessel;
-       local_window[i] = sum;
-   }
-
-   sum++;
-   for (i = 0; i < 256; i++)
-       window[i] = sqrt(local_window[i] / sum);
-}
-
 /**
  * Symmetrical Dequantization
  * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
  *            Tables 7.19 to 7.23
  */
-static inline float
+static inline int
 symmetric_dequant(int code, int levels)
 {
-    return (code - (levels >> 1)) * (2.0f / levels);
+    return ((code - (levels >> 1)) << 24) / levels;
 }
 
 /*
  * Initialize tables at runtime.
  */
-static void ac3_tables_init(void)
+static av_cold void ac3_tables_init(void)
 {
     int i;
 
@@ -273,11 +247,6 @@ static void ac3_tables_init(void)
         dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
     }
 
-    /* generate scale factors for exponents and asymmetrical dequantization
-       reference: Section 7.3.2 Expansion of Mantissas for Asymmetric Quantization */
-    for (i = 0; i < 25; i++)
-        scale_factors[i] = pow(2.0, -i);
-
     /* generate exponent tables
        reference: Section 7.1.3 Exponent Decoding */
     for(i=0; i<128; i++) {
@@ -291,7 +260,7 @@ static void ac3_tables_init(void)
 /**
  * AVCodec initialization
  */
-static int ac3_decode_init(AVCodecContext *avctx)
+static av_cold int ac3_decode_init(AVCodecContext *avctx)
 {
     AC3DecodeContext *s = avctx->priv_data;
     s->avctx = avctx;
@@ -300,7 +269,7 @@ static int ac3_decode_init(AVCodecContext *avctx)
     ac3_tables_init();
     ff_mdct_init(&s->imdct_256, 8, 1);
     ff_mdct_init(&s->imdct_512, 9, 1);
-    ac3_window_init(s->window);
+    ff_kbd_window_init(s->window, 5.0, 256);
     dsputil_init(&s->dsp, avctx);
     av_init_random(0, &s->dith_state);
 
@@ -319,6 +288,7 @@ static int ac3_decode_init(AVCodecContext *avctx)
             avctx->request_channels <= 2) {
         avctx->channels = avctx->request_channels;
     }
+    s->downmixed = 1;
 
     return 0;
 }
@@ -332,7 +302,6 @@ static int ac3_parse_header(AC3DecodeContext *s)
 {
     AC3HeaderInfo hdr;
     GetBitContext *gbc = &s->gbc;
-    float center_mix_level, surround_mix_level;
     int err, i;
 
     err = ff_ac3_parse_header(gbc->buffer, &hdr);
@@ -360,6 +329,10 @@ static int ac3_parse_header(AC3DecodeContext *s)
     if(s->lfe_on)
         s->output_mode |= AC3_OUTPUT_LFEON;
 
+    /* set default mix levels */
+    s->center_mix_level   = 3;  // -4.5dB
+    s->surround_mix_level = 4;  // -6.0dB
+
     /* skip over portion of header which has already been read */
     skip_bits(gbc, 16); // skip the sync_word
     skip_bits(gbc, 16); // skip crc1
@@ -369,9 +342,9 @@ static int ac3_parse_header(AC3DecodeContext *s)
         skip_bits(gbc, 2); // skip dsurmod
     } else {
         if((s->channel_mode & 1) && s->channel_mode != AC3_CHMODE_MONO)
-            center_mix_level = gain_levels[center_levels[get_bits(gbc, 2)]];
+            s->center_mix_level = center_levels[get_bits(gbc, 2)];
         if(s->channel_mode & 4)
-            surround_mix_level = gain_levels[surround_levels[get_bits(gbc, 2)]];
+            s->surround_mix_level = surround_levels[get_bits(gbc, 2)];
     }
     skip_bits1(gbc); // skip lfeon
 
@@ -404,25 +377,43 @@ static int ac3_parse_header(AC3DecodeContext *s)
         } while(i--);
     }
 
-    /* set stereo downmixing coefficients
-       reference: Section 7.8.2 Downmixing Into Two Channels */
+    return 0;
+}
+
+/**
+ * Set stereo downmixing coefficients based on frame header info.
+ * reference: Section 7.8.2 Downmixing Into Two Channels
+ */
+static void set_downmix_coeffs(AC3DecodeContext *s)
+{
+    int i;
+    float cmix = gain_levels[s->center_mix_level];
+    float smix = gain_levels[s->surround_mix_level];
+
     for(i=0; i<s->fbw_channels; i++) {
         s->downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
         s->downmix_coeffs[i][1] = gain_levels[ac3_default_coeffs[s->channel_mode][i][1]];
     }
     if(s->channel_mode > 1 && s->channel_mode & 1) {
-        s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = center_mix_level;
+        s->downmix_coeffs[1][0] = s->downmix_coeffs[1][1] = cmix;
     }
     if(s->channel_mode == AC3_CHMODE_2F1R || s->channel_mode == AC3_CHMODE_3F1R) {
         int nf = s->channel_mode - 2;
-        s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = surround_mix_level * LEVEL_MINUS_3DB;
+        s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf][1] = smix * LEVEL_MINUS_3DB;
     }
     if(s->channel_mode == AC3_CHMODE_2F2R || s->channel_mode == AC3_CHMODE_3F2R) {
         int nf = s->channel_mode - 4;
-        s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = surround_mix_level;
+        s->downmix_coeffs[nf][0] = s->downmix_coeffs[nf+1][1] = smix;
     }
 
-    return 0;
+    /* calculate adjustment needed for each channel to avoid clipping */
+    s->downmix_coeff_adjust[0] = s->downmix_coeff_adjust[1] = 0.0f;
+    for(i=0; i<s->fbw_channels; i++) {
+        s->downmix_coeff_adjust[0] += s->downmix_coeffs[i][0];
+        s->downmix_coeff_adjust[1] += s->downmix_coeffs[i][1];
+    }
+    s->downmix_coeff_adjust[0] = 1.0f / s->downmix_coeff_adjust[0];
+    s->downmix_coeff_adjust[1] = 1.0f / s->downmix_coeff_adjust[1];
 }
 
 /**
@@ -472,9 +463,9 @@ static void uncouple_channels(AC3DecodeContext *s)
             for(j=0; j<12; j++) {
                 for(ch=1; ch<=s->fbw_channels; ch++) {
                     if(s->channel_in_cpl[ch]) {
-                        s->transform_coeffs[ch][i] = s->transform_coeffs[CPL_CH][i] * s->cpl_coords[ch][bnd] * 8.0f;
+                        s->fixed_coeffs[ch][i] = ((int64_t)s->fixed_coeffs[CPL_CH][i] * (int64_t)s->cpl_coords[ch][bnd]) >> 23;
                         if (ch == 2 && s->phase_flags[bnd])
-                            s->transform_coeffs[ch][i] = -s->transform_coeffs[ch][i];
+                            s->fixed_coeffs[ch][i] = -s->fixed_coeffs[ch][i];
                     }
                 }
                 i++;
@@ -487,9 +478,9 @@ static void uncouple_channels(AC3DecodeContext *s)
  * Grouped mantissas for 3-level 5-level and 11-level quantization
  */
 typedef struct {
-    float b1_mant[3];
-    float b2_mant[3];
-    float b4_mant[2];
+    int b1_mant[3];
+    int b2_mant[3];
+    int b4_mant[2];
     int b1ptr;
     int b2ptr;
     int b4ptr;
@@ -505,11 +496,11 @@ static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_group
     int i, gcode, tbap, start, end;
     uint8_t *exps;
     uint8_t *bap;
-    float *coeffs;
+    int *coeffs;
 
     exps = s->dexps[ch_index];
     bap = s->bap[ch_index];
-    coeffs = s->transform_coeffs[ch_index];
+    coeffs = s->fixed_coeffs[ch_index];
     start = s->start_freq[ch_index];
     end = s->end_freq[ch_index];
 
@@ -517,7 +508,7 @@ static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_group
         tbap = bap[i];
         switch (tbap) {
             case 0:
-                coeffs[i] = ((av_random(&s->dith_state) & 0xFFFF) / 65535.0f) - 0.5f;
+                coeffs[i] = (av_random(&s->dith_state) & 0x7FFFFF) - 4194304;
                 break;
 
             case 1:
@@ -560,12 +551,14 @@ static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_group
                 coeffs[i] = b5_mantissas[get_bits(gbc, 4)];
                 break;
 
-            default:
+            default: {
                 /* asymmetric dequantization */
-                coeffs[i] = get_sbits(gbc, quantization_tab[tbap]) * scale_factors[quantization_tab[tbap]-1];
+                int qlevel = quantization_tab[tbap];
+                coeffs[i] = get_sbits(gbc, qlevel) << (24 - qlevel);
                 break;
+            }
         }
-        coeffs[i] *= scale_factors[exps[i]];
+        coeffs[i] >>= exps[i];
     }
 
     return 0;
@@ -578,12 +571,12 @@ static int get_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, mant_group
 static void remove_dithering(AC3DecodeContext *s) {
     int ch, i;
     int end=0;
-    float *coeffs;
+    int *coeffs;
     uint8_t *bap;
 
     for(ch=1; ch<=s->fbw_channels; ch++) {
         if(!s->dither_flag[ch]) {
-            coeffs = s->transform_coeffs[ch];
+            coeffs = s->fixed_coeffs[ch];
             bap = s->bap[ch];
             if(s->channel_in_cpl[ch])
                 end = s->start_freq[CPL_CH];
@@ -591,13 +584,13 @@ static void remove_dithering(AC3DecodeContext *s) {
                 end = s->end_freq[ch];
             for(i=0; i<end; i++) {
                 if(!bap[i])
-                    coeffs[i] = 0.0f;
+                    coeffs[i] = 0;
             }
             if(s->channel_in_cpl[ch]) {
                 bap = s->bap[CPL_CH];
                 for(; i<s->end_freq[CPL_CH]; i++) {
                     if(!bap[i])
-                        coeffs[i] = 0.0f;
+                        coeffs[i] = 0;
                 }
             }
         }
@@ -654,7 +647,7 @@ static void do_rematrixing(AC3DecodeContext *s)
 {
     int bnd, i;
     int end, bndend;
-    float tmp0, tmp1;
+    int tmp0, tmp1;
 
     end = FFMIN(s->end_freq[1], s->end_freq[2]);
 
@@ -662,10 +655,10 @@ static void do_rematrixing(AC3DecodeContext *s)
         if(s->rematrixing_flags[bnd]) {
             bndend = FFMIN(end, rematrix_band_tab[bnd+1]);
             for(i=rematrix_band_tab[bnd]; i<bndend; i++) {
-                tmp0 = s->transform_coeffs[1][i];
-                tmp1 = s->transform_coeffs[2][i];
-                s->transform_coeffs[1][i] = tmp0 + tmp1;
-                s->transform_coeffs[2][i] = tmp0 - tmp1;
+                tmp0 = s->fixed_coeffs[1][i];
+                tmp1 = s->fixed_coeffs[2][i];
+                s->fixed_coeffs[1][i] = tmp0 + tmp1;
+                s->fixed_coeffs[2][i] = tmp0 - tmp1;
             }
         }
     }
@@ -717,15 +710,9 @@ static void do_imdct_256(AC3DecodeContext *s, int chindex)
  * Convert frequency domain coefficients to time-domain audio samples.
  * reference: Section 7.9.4 Transformation Equations
  */
-static inline void do_imdct(AC3DecodeContext *s)
+static inline void do_imdct(AC3DecodeContext *s, int channels)
 {
     int ch;
-    int channels;
-
-    /* Don't perform the IMDCT on the LFE channel unless it's used in the output */
-    channels = s->fbw_channels;
-    if(s->output_mode & AC3_OUTPUT_LFEON)
-        channels++;
 
     for (ch=1; ch<=channels; ch++) {
         if (s->block_switch[ch]) {
@@ -748,30 +735,57 @@ static inline void do_imdct(AC3DecodeContext *s)
 /**
  * Downmix the output to mono or stereo.
  */
-static void ac3_downmix(AC3DecodeContext *s)
+static void ac3_downmix(AC3DecodeContext *s,
+                        float samples[AC3_MAX_CHANNELS][256], int ch_offset)
 {
     int i, j;
-    float v0, v1, s0, s1;
+    float v0, v1;
 
     for(i=0; i<256; i++) {
-        v0 = v1 = s0 = s1 = 0.0f;
+        v0 = v1 = 0.0f;
         for(j=0; j<s->fbw_channels; j++) {
-            v0 += s->output[j][i] * s->downmix_coeffs[j][0];
-            v1 += s->output[j][i] * s->downmix_coeffs[j][1];
-            s0 += s->downmix_coeffs[j][0];
-            s1 += s->downmix_coeffs[j][1];
+            v0 += samples[j+ch_offset][i] * s->downmix_coeffs[j][0];
+            v1 += samples[j+ch_offset][i] * s->downmix_coeffs[j][1];
         }
-        v0 /= s0;
-        v1 /= s1;
+        v0 *= s->downmix_coeff_adjust[0];
+        v1 *= s->downmix_coeff_adjust[1];
         if(s->output_mode == AC3_CHMODE_MONO) {
-            s->output[0][i] = (v0 + v1) * LEVEL_MINUS_3DB;
+            samples[ch_offset][i] = (v0 + v1) * LEVEL_MINUS_3DB;
         } else if(s->output_mode == AC3_CHMODE_STEREO) {
-            s->output[0][i] = v0;
-            s->output[1][i] = v1;
+            samples[  ch_offset][i] = v0;
+            samples[1+ch_offset][i] = v1;
         }
     }
 }
 
+/**
+ * Upmix delay samples from stereo to original channel layout.
+ */
+static void ac3_upmix_delay(AC3DecodeContext *s)
+{
+    int channel_data_size = sizeof(s->delay[0]);
+    switch(s->channel_mode) {
+        case AC3_CHMODE_DUALMONO:
+        case AC3_CHMODE_STEREO:
+            /* upmix mono to stereo */
+            memcpy(s->delay[1], s->delay[0], channel_data_size);
+            break;
+        case AC3_CHMODE_2F2R:
+            memset(s->delay[3], 0, channel_data_size);
+        case AC3_CHMODE_2F1R:
+            memset(s->delay[2], 0, channel_data_size);
+            break;
+        case AC3_CHMODE_3F2R:
+            memset(s->delay[4], 0, channel_data_size);
+        case AC3_CHMODE_3F1R:
+            memset(s->delay[3], 0, channel_data_size);
+        case AC3_CHMODE_3F:
+            memcpy(s->delay[2], s->delay[1], channel_data_size);
+            memset(s->delay[1], 0, channel_data_size);
+            break;
+    }
+}
+
 /**
  * Parse an audio block from AC-3 bitstream.
  */
@@ -780,14 +794,20 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
     int fbw_channels = s->fbw_channels;
     int channel_mode = s->channel_mode;
     int i, bnd, seg, ch;
+    int different_transforms;
+    int downmix_output;
     GetBitContext *gbc = &s->gbc;
     uint8_t bit_alloc_stages[AC3_MAX_CHANNELS];
 
     memset(bit_alloc_stages, 0, AC3_MAX_CHANNELS);
 
     /* block switch flags */
-    for (ch = 1; ch <= fbw_channels; ch++)
+    different_transforms = 0;
+    for (ch = 1; ch <= fbw_channels; ch++) {
         s->block_switch[ch] = get_bits1(gbc);
+        if(ch > 1 && s->block_switch[ch] != s->block_switch[1])
+            different_transforms = 1;
+    }
 
     /* dithering flags */
     s->dither_all = 1;
@@ -862,10 +882,10 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
                         cpl_coord_exp = get_bits(gbc, 4);
                         cpl_coord_mant = get_bits(gbc, 4);
                         if (cpl_coord_exp == 15)
-                            s->cpl_coords[ch][bnd] = cpl_coord_mant / 16.0f;
+                            s->cpl_coords[ch][bnd] = cpl_coord_mant << 22;
                         else
-                            s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16.0f) / 32.0f;
-                        s->cpl_coords[ch][bnd] *= scale_factors[cpl_coord_exp + master_cpl_coord];
+                            s->cpl_coords[ch][bnd] = (cpl_coord_mant + 16) << 21;
+                        s->cpl_coords[ch][bnd] >>= (cpl_coord_exp + master_cpl_coord);
                     }
                 }
             }
@@ -880,8 +900,7 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
 
     /* stereo rematrixing strategy and band structure */
     if (channel_mode == AC3_CHMODE_STEREO) {
-        s->rematrixing_strategy = get_bits1(gbc);
-        if (s->rematrixing_strategy) {
+        if (get_bits1(gbc)) {
             s->num_rematrixing_bands = 4;
             if(s->cpl_in_use && s->start_freq[CPL_CH] <= 61)
                 s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37);
@@ -1049,23 +1068,47 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
 
     /* apply scaling to coefficients (headroom, dynrng) */
     for(ch=1; ch<=s->channels; ch++) {
-        float gain = 2.0f * s->mul_bias;
+        float gain = s->mul_bias / 4194304.0f;
         if(s->channel_mode == AC3_CHMODE_DUALMONO) {
             gain *= s->dynamic_range[ch-1];
         } else {
             gain *= s->dynamic_range[0];
         }
-        for(i=0; i<s->end_freq[ch]; i++) {
-            s->transform_coeffs[ch][i] *= gain;
+        for(i=0; i<256; i++) {
+            s->transform_coeffs[ch][i] = s->fixed_coeffs[ch][i] * gain;
         }
     }
 
-    do_imdct(s);
+    /* downmix and MDCT. order depends on whether block switching is used for
+       any channel in this block. this is because coefficients for the long
+       and short transforms cannot be mixed. */
+    downmix_output = s->channels != s->out_channels &&
+                     !((s->output_mode & AC3_OUTPUT_LFEON) &&
+                     s->fbw_channels == s->out_channels);
+    if(different_transforms) {
+        /* the delay samples have already been downmixed, so we upmix the delay
+           samples in order to reconstruct all channels before downmixing. */
+        if(s->downmixed) {
+            s->downmixed = 0;
+            ac3_upmix_delay(s);
+        }
 
-    /* downmix output if needed */
-    if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
-            s->fbw_channels == s->out_channels)) {
-        ac3_downmix(s);
+        do_imdct(s, s->channels);
+
+        if(downmix_output) {
+            ac3_downmix(s, s->output, 0);
+        }
+    } else {
+        if(downmix_output) {
+            ac3_downmix(s, s->transform_coeffs, 1);
+        }
+
+        if(!s->downmixed) {
+            s->downmixed = 1;
+            ac3_downmix(s, s->delay, 0);
+        }
+
+        do_imdct(s, s->out_channels);
     }
 
     /* convert float to 16-bit integer */
@@ -1082,7 +1125,8 @@ static int ac3_parse_audio_block(AC3DecodeContext *s, int blk)
 /**
  * Decode a single AC-3 frame.
  */
-static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, uint8_t *buf, int buf_size)
+static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
+                            const uint8_t *buf, int buf_size)
 {
     AC3DecodeContext *s = avctx->priv_data;
     int16_t *out_samples = (int16_t *)data;
@@ -1121,7 +1165,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     }
 
     /* check for crc mismatch */
-    if(avctx->error_resilience > 0) {
+    if(avctx->error_resilience >= FF_ER_CAREFUL) {
         if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) {
             av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n");
             return -1;
@@ -1141,6 +1185,12 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     }
     avctx->channels = s->out_channels;
 
+    /* set downmixing coefficients if needed */
+    if(s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
+            s->fbw_channels == s->out_channels)) {
+        set_downmix_coeffs(s);
+    }
+
     /* parse the audio blocks */
     for (blk = 0; blk < NB_BLOCKS; blk++) {
         if (ac3_parse_audio_block(s, blk)) {
@@ -1159,7 +1209,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
 /**
  * Uninitialize the AC-3 decoder.
  */
-static int ac3_decode_end(AVCodecContext *avctx)
+static av_cold int ac3_decode_end(AVCodecContext *avctx)
 {
     AC3DecodeContext *s = avctx->priv_data;
     ff_mdct_end(&s->imdct_512);