h264: correct implicit weight table computation for long ref pics

[ffmpeg] / libavcodec / ac3dec.c

diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index a1c86405998a6d8ae9da72c558ac01100484482d..663acc0ac978c3dd36be821c7b9c26e79de7b1dc 100644 (file)
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -30,6 +30,7 @@
 #include <string.h>
 
 #include "libavutil/crc.h"
+#include "libavutil/opt.h"
 #include "internal.h"
 #include "aac_ac3_parser.h"
 #include "ac3_parser.h"
@@ -67,16 +68,6 @@ static const uint8_t quantization_tab[16] = {
 static float dynamic_range_tab[256];
 
 /** Adjustments in dB gain */
-#define LEVEL_PLUS_3DB          1.4142135623730950
-#define LEVEL_PLUS_1POINT5DB    1.1892071150027209
-#define LEVEL_MINUS_1POINT5DB   0.8408964152537145
-#define LEVEL_MINUS_3DB         0.7071067811865476
-#define LEVEL_MINUS_4POINT5DB   0.5946035575013605
-#define LEVEL_MINUS_6DB         0.5000000000000000
-#define LEVEL_MINUS_9DB         0.3535533905932738
-#define LEVEL_ZERO              0.0000000000000000
-#define LEVEL_ONE               1.0000000000000000
-
 static const float gain_levels[9] = {
     LEVEL_PLUS_3DB,
     LEVEL_PLUS_1POINT5DB,
@@ -194,11 +185,18 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     ff_mdct_init(&s->imdct_512, 9, 1, 1.0);
     ff_kbd_window_init(s->window, 5.0, 256);
     dsputil_init(&s->dsp, avctx);
+    ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
     ff_fmt_convert_init(&s->fmt_conv, avctx);
     av_lfg_init(&s->dith_state, 0);
 
     /* set scale value for float to int16 conversion */
-    s->mul_bias = 32767.0f;
+    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+        s->mul_bias = 1.0f;
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+    } else {
+        s->mul_bias = 32767.0f;
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+    }
 
     /* allow downmixing to stereo or mono */
     if (avctx->channels > 0 && avctx->request_channels > 0 &&
@@ -213,7 +211,6 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
         if (!s->input_buffer)
             return AVERROR(ENOMEM);
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
     return 0;
 }
 
@@ -1182,8 +1179,8 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         /* channel delta offset, len and bit allocation */
         for (ch = !cpl_in_use; ch <= fbw_channels; ch++) {
             if (s->dba_mode[ch] == DBA_NEW) {
-                s->dba_nsegs[ch] = get_bits(gbc, 3);
-                for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) {
+                s->dba_nsegs[ch] = get_bits(gbc, 3) + 1;
+                for (seg = 0; seg < s->dba_nsegs[ch]; seg++) {
                     s->dba_offsets[ch][seg] = get_bits(gbc, 5);
                     s->dba_lengths[ch][seg] = get_bits(gbc, 4);
                     s->dba_values[ch][seg] = get_bits(gbc, 3);
@@ -1223,7 +1220,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
             /* Compute bit allocation */
             const uint8_t *bap_tab = s->channel_uses_aht[ch] ?
                                      ff_eac3_hebap_tab : ff_ac3_bap_tab;
-            ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
+            s->ac3dsp.bit_alloc_calc_bap(s->mask[ch], s->psd[ch],
                                       s->start_freq[ch], s->end_freq[ch],
                                       s->snr_offset[ch],
                                       s->bit_alloc_params.floor,
@@ -1308,7 +1305,8 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;
     AC3DecodeContext *s = avctx->priv_data;
-    int16_t *out_samples = (int16_t *)data;
+    float   *out_samples_flt = data;
+    int16_t *out_samples_s16 = data;
     int blk, ch, err;
     const uint8_t *channel_map;
     const float *output[AC3_MAX_CHANNELS];
@@ -1414,10 +1412,18 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
             av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
             err = 1;
         }
-        s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
-        out_samples += 256 * s->out_channels;
+        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+            s->fmt_conv.float_interleave(out_samples_flt, output, 256,
+                                         s->out_channels);
+            out_samples_flt += 256 * s->out_channels;
+        } else {
+            s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256,
+                                                  s->out_channels);
+            out_samples_s16 += 256 * s->out_channels;
+        }
     }
-    *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
+    *data_size = s->num_blocks * 256 * avctx->channels *
+                 av_get_bytes_per_sample(avctx->sample_fmt);
     return FFMIN(buf_size, s->frame_size);
 }
 
@@ -1435,6 +1441,20 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx)
     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), FF_OPT_TYPE_FLOAT, {1.0}, 0.0, 1.0, PAR },
+    { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+    .class_name = "(E-)AC3 decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
 AVCodec ff_ac3_decoder = {
     .name = "ac3",
     .type = AVMEDIA_TYPE_AUDIO,
@@ -1444,6 +1464,10 @@ AVCodec ff_ac3_decoder = {
     .close = ac3_decode_end,
     .decode = ac3_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
+    .priv_class = &ac3_decoder_class,
 };
 
 #if CONFIG_EAC3_DECODER
@@ -1456,5 +1480,9 @@ AVCodec ff_eac3_decoder = {
     .close = ac3_decode_end,
     .decode = ac3_decode_frame,
     .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+    .sample_fmts = (const enum AVSampleFormat[]) {
+        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+    },
+    .priv_class = &ac3_decoder_class,
 };
 #endif