Merge commit 'dc70c19476e76f1118df73b5d97cc76f0e5f6f6c'
[ffmpeg.git] / libavcodec / ac3dec.c
index 97ce287..6df697e 100644 (file)
@@ -7,20 +7,20 @@
  * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
  * Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * 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.
  *
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -65,6 +65,7 @@ static const uint8_t quantization_tab[16] = {
 
 /** dynamic range table. converts codes to scale factors. */
 static float dynamic_range_tab[256];
+static float heavy_dynamic_range_tab[256];
 
 /** Adjustments in dB gain */
 static const float gain_levels[9] = {
@@ -111,7 +112,7 @@ static const uint8_t ac3_default_coeffs[8][5][2] = {
 static inline int
 symmetric_dequant(int code, int levels)
 {
-    return ((code - (levels >> 1)) << 24) / levels;
+    return ((code - (levels >> 1)) * (1 << 24)) / levels;
 }
 
 /*
@@ -164,6 +165,14 @@ static av_cold void ac3_tables_init(void)
         int v = (i >> 5) - ((i >> 7) << 3) - 5;
         dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0x1F) | 0x20);
     }
+
+    /* generate compr dynamic range table
+       reference: Section 7.7.2 Heavy Compression */
+    for (i = 0; i < 256; i++) {
+        int v = (i >> 4) - ((i >> 7) << 4) - 4;
+        heavy_dynamic_range_tab[i] = powf(2.0f, v) * ((i & 0xF) | 0x10);
+    }
+
 }
 
 /**
@@ -180,14 +189,23 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     ac3_tables_init();
     ff_mdct_init(&s->imdct_256, 8, 1, 1.0);
     ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
-    ff_kbd_window_init(s->window, 5.0, 256);
+    AC3_RENAME(ff_kbd_window_init)(s->window, 5.0, 256);
     ff_bswapdsp_init(&s->bdsp);
-    avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
-    ff_ac3dsp_init(&s->ac3dsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
+
+#if (USE_FIXED)
+    s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+#else
+    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
     ff_fmt_convert_init(&s->fmt_conv, avctx);
+#endif
+
+    ff_ac3dsp_init(&s->ac3dsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
     av_lfg_init(&s->dith_state, 0);
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+    if (USE_FIXED)
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+    else
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     /* allow downmixing to stereo or mono */
     if (avctx->channels > 1 &&
@@ -219,9 +237,19 @@ static int ac3_parse_header(AC3DecodeContext *s)
     /* read the rest of the bsi. read twice for dual mono mode. */
     i = !s->channel_mode;
     do {
-        skip_bits(gbc, 5); // skip dialog normalization
-        if (get_bits1(gbc))
-            skip_bits(gbc, 8); //skip compression
+        s->dialog_normalization[(!s->channel_mode)-i] = -get_bits(gbc, 5);
+        if (s->dialog_normalization[(!s->channel_mode)-i] == 0) {
+            s->dialog_normalization[(!s->channel_mode)-i] = -31;
+        }
+        if (s->target_level != 0) {
+            s->level_gain[(!s->channel_mode)-i] = powf(2.0f,
+                (float)(s->target_level -
+                s->dialog_normalization[(!s->channel_mode)-i])/6.0f);
+        }
+        if (s->compression_exists[(!s->channel_mode)-i] = get_bits1(gbc)) {
+            s->heavy_dynamic_range[(!s->channel_mode)-i] =
+                AC3_HEAVY_RANGE(get_bits(gbc, 8));
+        }
         if (get_bits1(gbc))
             skip_bits(gbc, 8); //skip language code
         if (get_bits1(gbc))
@@ -267,10 +295,10 @@ static int ac3_parse_header(AC3DecodeContext *s)
  */
 static int parse_frame_header(AC3DecodeContext *s)
 {
-    AC3HeaderInfo hdr;
+    AC3HeaderInfo hdr, *phdr=&hdr;
     int err;
 
-    err = avpriv_ac3_parse_header(&s->gbc, &hdr);
+    err = avpriv_ac3_parse_header2(&s->gbc, &phdr);
     if (err)
         return err;
 
@@ -338,40 +366,45 @@ static void set_downmix_coeffs(AC3DecodeContext *s)
     float cmix = gain_levels[s->  center_mix_level];
     float smix = gain_levels[s->surround_mix_level];
     float norm0, norm1;
+    float downmix_coeffs[AC3_MAX_CHANNELS][2];
 
     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]];
+        downmix_coeffs[i][0] = gain_levels[ac3_default_coeffs[s->channel_mode][i][0]];
+        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] = cmix;
+        downmix_coeffs[1][0] = 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] = smix * LEVEL_MINUS_3DB;
+        downmix_coeffs[nf][0] = 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] = smix;
+        downmix_coeffs[nf][0] = downmix_coeffs[nf+1][1] = smix;
     }
 
     /* renormalize */
     norm0 = norm1 = 0.0;
     for (i = 0; i < s->fbw_channels; i++) {
-        norm0 += s->downmix_coeffs[i][0];
-        norm1 += s->downmix_coeffs[i][1];
+        norm0 += downmix_coeffs[i][0];
+        norm1 += downmix_coeffs[i][1];
     }
     norm0 = 1.0f / norm0;
     norm1 = 1.0f / norm1;
     for (i = 0; i < s->fbw_channels; i++) {
-        s->downmix_coeffs[i][0] *= norm0;
-        s->downmix_coeffs[i][1] *= norm1;
+        downmix_coeffs[i][0] *= norm0;
+        downmix_coeffs[i][1] *= norm1;
     }
 
     if (s->output_mode == AC3_CHMODE_MONO) {
         for (i = 0; i < s->fbw_channels; i++)
-            s->downmix_coeffs[i][0] = (s->downmix_coeffs[i][0] +
-                                       s->downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+            downmix_coeffs[i][0] = (downmix_coeffs[i][0] +
+                                    downmix_coeffs[i][1]) * LEVEL_MINUS_3DB;
+    }
+    for (i = 0; i < s->fbw_channels; i++) {
+        s->downmix_coeffs[i][0] = FIXR12(downmix_coeffs[i][0]);
+        s->downmix_coeffs[i][1] = FIXR12(downmix_coeffs[i][1]);
     }
 }
 
@@ -429,7 +462,7 @@ static void calc_transform_coeffs_cpl(AC3DecodeContext *s)
                 int cpl_coord = s->cpl_coords[ch][band] << 5;
                 for (bin = band_start; bin < band_end; bin++) {
                     s->fixed_coeffs[ch][bin] =
-                        MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord);
+                        MULH(s->fixed_coeffs[CPL_CH][bin] * (1 << 4), cpl_coord);
                 }
                 if (ch == 2 && s->phase_flags[band]) {
                     for (bin = band_start; bin < band_end; bin++)
@@ -475,7 +508,7 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
         case 0:
             /* random noise with approximate range of -0.707 to 0.707 */
             if (dither)
-                mantissa = (av_lfg_get(&s->dith_state) / 362) - 5932275;
+                mantissa = (((av_lfg_get(&s->dith_state)>>8)*181)>>8) - 5931008;
             else
                 mantissa = 0;
             break;
@@ -522,8 +555,11 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma
             break;
         default: /* 6 to 15 */
             /* Shift mantissa and sign-extend it. */
-            mantissa = get_sbits(gbc, quantization_tab[bap]);
-            mantissa <<= 24 - quantization_tab[bap];
+            if (bap > 15) {
+                av_log(s->avctx, AV_LOG_ERROR, "bap %d is invalid in plain AC-3\n", bap);
+                bap = 15;
+            }
+            mantissa = (unsigned)get_sbits(gbc, quantization_tab[bap]) << (24 - quantization_tab[bap]);
             break;
         }
         coeffs[freq] = mantissa >> exps[freq];
@@ -557,7 +593,7 @@ static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch,
         /* if AHT is used, mantissas for all blocks are encoded in the first
            block of the frame. */
         int bin;
-        if (!blk && CONFIG_EAC3_DECODER)
+        if (CONFIG_EAC3_DECODER && !blk)
             ff_eac3_decode_transform_coeffs_aht_ch(s, ch);
         for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) {
             s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin];
@@ -635,20 +671,30 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
     for (ch = 1; ch <= channels; ch++) {
         if (s->block_switch[ch]) {
             int i;
-            float *x = s->tmp_output + 128;
+            FFTSample *x = s->tmp_output + 128;
             for (i = 0; i < 128; i++)
                 x[i] = s->transform_coeffs[ch][2 * i];
             s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x);
-            s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+#if USE_FIXED
+            s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+                                       s->tmp_output, s->window, 128, 8);
+#else
+            s->fdsp->vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
                                        s->tmp_output, s->window, 128);
+#endif
             for (i = 0; i < 128; i++)
                 x[i] = s->transform_coeffs[ch][2 * i + 1];
             s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch - 1], x);
         } else {
             s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]);
-            s->fdsp.vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
+#if USE_FIXED
+            s->fdsp->vector_fmul_window_scaled(s->outptr[ch - 1], s->delay[ch - 1],
+                                       s->tmp_output, s->window, 128, 8);
+#else
+            s->fdsp->vector_fmul_window(s->outptr[ch - 1], s->delay[ch - 1],
                                        s->tmp_output, s->window, 128);
-            memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(float));
+#endif
+            memcpy(s->delay[ch - 1], s->tmp_output + 128, 128 * sizeof(FFTSample));
         }
     }
 }
@@ -783,13 +829,14 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         if (get_bits1(gbc)) {
             /* Allow asymmetric application of DRC when drc_scale > 1.
                Amplification of quiet sounds is enhanced */
-            float range = dynamic_range_tab[get_bits(gbc, 8)];
-            if (range > 1.0 || s->drc_scale <= 1.0)
-                s->dynamic_range[i] = powf(range, s->drc_scale);
+            int range_bits = get_bits(gbc, 8);
+            INTFLOAT range = AC3_RANGE(range_bits);
+            if (range_bits <= 127 || s->drc_scale <= 1.0)
+                s->dynamic_range[i] = AC3_DYNAMIC_RANGE(range);
             else
                 s->dynamic_range[i] = range;
         } else if (blk == 0) {
-            s->dynamic_range[i] = 1.0f;
+            s->dynamic_range[i] = AC3_DYNAMIC_RANGE1;
         }
     } while (i--);
 
@@ -815,6 +862,9 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             if (start_subband > 7)
                 start_subband += start_subband - 7;
             end_subband    = get_bits(gbc, 3) + 5;
+#if USE_FIXED
+            s->spx_dst_end_freq = end_freq_inv_tab[end_subband-5];
+#endif
             if (end_subband   > 7)
                 end_subband   += end_subband   - 7;
             dst_start_freq = dst_start_freq * 12 + 25;
@@ -835,7 +885,8 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
 
             s->spx_dst_start_freq = dst_start_freq;
             s->spx_src_start_freq = src_start_freq;
-            s->spx_dst_end_freq   = dst_end_freq;
+            if (!USE_FIXED)
+                s->spx_dst_end_freq   = dst_end_freq;
 
             decode_band_structure(gbc, blk, s->eac3, 0,
                                   start_subband, end_subband,
@@ -855,26 +906,47 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         for (ch = 1; ch <= fbw_channels; ch++) {
             if (s->channel_uses_spx[ch]) {
                 if (s->first_spx_coords[ch] || get_bits1(gbc)) {
-                    float spx_blend;
+                    INTFLOAT spx_blend;
                     int bin, master_spx_coord;
 
                     s->first_spx_coords[ch] = 0;
-                    spx_blend = get_bits(gbc, 5) * (1.0f/32);
+                    spx_blend = AC3_SPX_BLEND(get_bits(gbc, 5));
                     master_spx_coord = get_bits(gbc, 2) * 3;
 
                     bin = s->spx_src_start_freq;
                     for (bnd = 0; bnd < s->num_spx_bands; bnd++) {
-                        int bandsize;
+                        int bandsize = s->spx_band_sizes[bnd];
                         int spx_coord_exp, spx_coord_mant;
-                        float nratio, sblend, nblend, spx_coord;
+                        INTFLOAT nratio, sblend, nblend;
+#if USE_FIXED
+                        /* calculate blending factors */
+                        int64_t accu = ((bin << 23) + (bandsize << 22))
+                                     * (int64_t)s->spx_dst_end_freq;
+                        nratio = (int)(accu >> 32);
+                        nratio -= spx_blend << 18;
+
+                        if (nratio < 0) {
+                            nblend = 0;
+                            sblend = 0x800000;
+                        } else if (nratio > 0x7fffff) {
+                            nblend = 14529495; // sqrt(3) in FP.23
+                            sblend = 0;
+                        } else {
+                            nblend = fixed_sqrt(nratio, 23);
+                            accu = (int64_t)nblend * 1859775393;
+                            nblend = (int)((accu + (1<<29)) >> 30);
+                            sblend = fixed_sqrt(0x800000 - nratio, 23);
+                        }
+#else
+                        float spx_coord;
 
                         /* calculate blending factors */
-                        bandsize = s->spx_band_sizes[bnd];
                         nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend;
                         nratio = av_clipf(nratio, 0.0f, 1.0f);
                         nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3)
                                                        // to give unity variance
                         sblend = sqrtf(1.0f - nratio);
+#endif
                         bin += bandsize;
 
                         /* decode spx coordinates */
@@ -883,11 +955,18 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                         if (spx_coord_exp == 15) spx_coord_mant <<= 1;
                         else                     spx_coord_mant += 4;
                         spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord);
-                        spx_coord = spx_coord_mant * (1.0f / (1 << 23));
 
                         /* multiply noise and signal blending factors by spx coordinate */
+#if USE_FIXED
+                        accu = (int64_t)nblend * spx_coord_mant;
+                        s->spx_noise_blend[ch][bnd]  = (int)((accu + (1<<22)) >> 23);
+                        accu = (int64_t)sblend * spx_coord_mant;
+                        s->spx_signal_blend[ch][bnd] = (int)((accu + (1<<22)) >> 23);
+#else
+                        spx_coord = spx_coord_mant * (1.0f / (1 << 23));
                         s->spx_noise_blend [ch][bnd] = nblend * spx_coord;
                         s->spx_signal_blend[ch][bnd] = sblend * spx_coord;
+#endif
                     }
                 }
             } else {
@@ -1244,18 +1323,28 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
 
     /* apply scaling to coefficients (headroom, dynrng) */
     for (ch = 1; ch <= s->channels; ch++) {
-        float gain = 1.0 / 4194304.0f;
-        if (s->channel_mode == AC3_CHMODE_DUALMONO) {
-            gain *= s->dynamic_range[2 - ch];
-        } else {
-            gain *= s->dynamic_range[0];
-        }
+        int audio_channel = 0;
+        INTFLOAT gain;
+        if (s->channel_mode == AC3_CHMODE_DUALMONO)
+            audio_channel = 2-ch;
+        if (s->heavy_compression && s->compression_exists[audio_channel])
+            gain = s->heavy_dynamic_range[audio_channel];
+        else
+            gain = s->dynamic_range[audio_channel];
+
+#if USE_FIXED
+        scale_coefs(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+#else
+        if (s->target_level != 0)
+          gain = gain * s->level_gain[audio_channel];
+        gain *= 1.0 / 4194304.0f;
         s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch],
                                                s->fixed_coeffs[ch], gain, 256);
+#endif
     }
 
     /* apply spectral extension to high frequency bins */
-    if (s->spx_in_use && CONFIG_EAC3_DECODER) {
+    if (CONFIG_EAC3_DECODER && s->spx_in_use) {
         ff_eac3_apply_spectral_extension(s);
     }
 
@@ -1276,19 +1365,24 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         do_imdct(s, s->channels);
 
         if (downmix_output) {
+#if USE_FIXED
+            ac3_downmix_c_fixed16(s->outptr, s->downmix_coeffs,
+                              s->out_channels, s->fbw_channels, 256);
+#else
             s->ac3dsp.downmix(s->outptr, s->downmix_coeffs,
                               s->out_channels, s->fbw_channels, 256);
+#endif
         }
     } else {
         if (downmix_output) {
-            s->ac3dsp.downmix(s->xcfptr + 1, s->downmix_coeffs,
-                              s->out_channels, s->fbw_channels, 256);
+            s->ac3dsp.AC3_RENAME(downmix)(s->xcfptr + 1, s->downmix_coeffs,
+                                          s->out_channels, s->fbw_channels, 256);
         }
 
         if (downmix_output && !s->downmixed) {
             s->downmixed = 1;
-            s->ac3dsp.downmix(s->dlyptr, s->downmix_coeffs, s->out_channels,
-                              s->fbw_channels, 128);
+            s->ac3dsp.AC3_RENAME(downmix)(s->dlyptr, s->downmix_coeffs,
+                                          s->out_channels, s->fbw_channels, 128);
         }
 
         do_imdct(s, s->out_channels);
@@ -1309,7 +1403,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     AC3DecodeContext *s = avctx->priv_data;
     int blk, ch, err, ret;
     const uint8_t *channel_map;
-    const float *output[AC3_MAX_CHANNELS];
+    const SHORTFLOAT *output[AC3_MAX_CHANNELS];
     enum AVMatrixEncoding matrix_encoding;
     AVDownmixInfo *downmix_info;
 
@@ -1324,7 +1418,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE));
     buf = s->input_buffer;
     /* initialize the GetBitContext with the start of valid AC-3 Frame */
-    init_get_bits(&s->gbc, buf, buf_size * 8);
+    if ((ret = init_get_bits8(&s->gbc, buf, buf_size)) < 0)
+        return ret;
 
     /* parse the syncinfo */
     err = parse_frame_header(s);
@@ -1367,7 +1462,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         if (s->frame_size > buf_size) {
             av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
             err = AAC_AC3_PARSE_ERROR_FRAME_SIZE;
-        } else if (avctx->err_recognition & AV_EF_CRCCHECK) {
+        } else if (avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_CAREFUL)) {
             /* check for crc mismatch */
             if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2],
                        s->frame_size - 2)) {
@@ -1401,6 +1496,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
             s->output_mode  = AC3_CHMODE_STEREO;
         }
 
+        s->loro_center_mix_level   = gain_levels[s->  center_mix_level];
+        s->loro_surround_mix_level = gain_levels[s->surround_mix_level];
+        s->ltrt_center_mix_level   = LEVEL_MINUS_3DB;
+        s->ltrt_surround_mix_level = LEVEL_MINUS_3DB;
         /* set downmixing coefficients if needed */
         if (s->channels != s->out_channels && !((s->output_mode & AC3_OUTPUT_LFEON) &&
                 s->fbw_channels == s->out_channels)) {
@@ -1422,19 +1521,18 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
 
     /* get output buffer */
     frame->nb_samples = s->num_blocks * AC3_BLOCK_SIZE;
-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
-        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
-    }
 
     /* decode the audio blocks */
     channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on];
+    for (ch = 0; ch < AC3_MAX_CHANNELS; ch++) {
+        output[ch] = s->output[ch];
+        s->outptr[ch] = s->output[ch];
+    }
     for (ch = 0; ch < s->channels; ch++) {
         if (ch < s->out_channels)
-            s->outptr[channel_map[ch]] = (float *)frame->data[ch];
-        else
-            s->outptr[ch] = s->output[ch];
-        output[ch] = s->output[ch];
+            s->outptr[channel_map[ch]] = (SHORTFLOAT *)frame->data[ch];
     }
     for (blk = 0; blk < s->num_blocks; blk++) {
         if (!err && decode_audio_block(s, blk)) {
@@ -1443,16 +1541,20 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         }
         if (err)
             for (ch = 0; ch < s->out_channels; ch++)
-                memcpy(s->outptr[channel_map[ch]], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+                memcpy(((SHORTFLOAT*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
         for (ch = 0; ch < s->out_channels; ch++)
             output[ch] = s->outptr[channel_map[ch]];
-        for (ch = 0; ch < s->out_channels; ch++)
-            s->outptr[ch] += AC3_BLOCK_SIZE;
+        for (ch = 0; ch < s->out_channels; ch++) {
+            if (!ch || channel_map[ch])
+                s->outptr[channel_map[ch]] += AC3_BLOCK_SIZE;
+        }
     }
 
+    av_frame_set_decode_error_flags(frame, err ? FF_DECODE_ERROR_INVALID_BITSTREAM : 0);
+
     /* keep last block for error concealment in next frame */
     for (ch = 0; ch < s->out_channels; ch++)
-        memcpy(s->output[ch], output[ch], sizeof(**output) * AC3_BLOCK_SIZE);
+        memcpy(s->output[ch], output[ch], AC3_BLOCK_SIZE*sizeof(SHORTFLOAT));
 
     /*
      * AVMatrixEncoding
@@ -1523,59 +1625,10 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx)
     AC3DecodeContext *s = avctx->priv_data;
     ff_mdct_end(&s->imdct_512);
     ff_mdct_end(&s->imdct_256);
+    av_freep(&s->fdsp);
 
     return 0;
 }
 
 #define OFFSET(x) offsetof(AC3DecodeContext, x)
 #define PAR (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM)
-static const AVOption options[] = {
-    { "drc_scale", "percentage of dynamic range compression to apply", OFFSET(drc_scale), AV_OPT_TYPE_FLOAT, {.dbl = 1.0}, 0.0, 6.0, PAR },
-    { NULL},
-};
-
-static const AVClass ac3_decoder_class = {
-    .class_name = "AC3 decoder",
-    .item_name  = av_default_item_name,
-    .option     = options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-AVCodec ff_ac3_decoder = {
-    .name           = "ac3",
-    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_AC3,
-    .priv_data_size = sizeof (AC3DecodeContext),
-    .init           = ac3_decode_init,
-    .close          = ac3_decode_end,
-    .decode         = ac3_decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
-                                                      AV_SAMPLE_FMT_NONE },
-    .priv_class     = &ac3_decoder_class,
-};
-
-#if CONFIG_EAC3_DECODER
-static const AVClass eac3_decoder_class = {
-    .class_name = "E-AC3 decoder",
-    .item_name  = av_default_item_name,
-    .option     = options,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-AVCodec ff_eac3_decoder = {
-    .name           = "eac3",
-    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_EAC3,
-    .priv_data_size = sizeof (AC3DecodeContext),
-    .init           = ac3_decode_init,
-    .close          = ac3_decode_end,
-    .decode         = ac3_decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
-                                                      AV_SAMPLE_FMT_NONE },
-    .priv_class     = &eac3_decoder_class,
-};
-#endif