libavcodec: Implementation of AC3 fixedpoint decoder
authorNedeljko Babic <nbabic@mips.com>
Tue, 1 Apr 2014 14:31:08 +0000 (16:31 +0200)
committerMichael Niedermayer <michaelni@gmx.at>
Tue, 1 Apr 2014 17:01:57 +0000 (19:01 +0200)
Signed-off-by: Nedeljko Babic <nbabic@mips.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
15 files changed:
Changelog
configure
doc/general.texi
libavcodec/Makefile
libavcodec/ac3.h
libavcodec/ac3dec.c
libavcodec/ac3dec.h
libavcodec/ac3dec_fixed.c [new file with mode: 0644]
libavcodec/ac3dec_float.c [new file with mode: 0644]
libavcodec/ac3dsp.c
libavcodec/ac3dsp.h
libavcodec/allcodecs.c
libavcodec/kbdwin.c
libavcodec/kbdwin.h
libavcodec/version.h

index 0e38ec8..b8147c9 100644 (file)
--- a/Changelog
+++ b/Changelog
@@ -2,6 +2,7 @@ Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
 version <next>:
+- added AC3 fixed-point decoding
 - shuffleplanes filter
 - subfile protocol
 - Phantom Cine demuxer
index b82891c..848dc7e 100755 (executable)
--- a/configure
+++ b/configure
@@ -1951,6 +1951,7 @@ aac_decoder_select="mdct sinewin"
 aac_encoder_select="audio_frame_queue mdct sinewin"
 aac_latm_decoder_select="aac_decoder aac_latm_parser"
 ac3_decoder_select="mdct ac3dsp ac3_parser dsputil"
+ac3_fixed_decoder_select="mdct ac3dsp ac3_parser dsputil"
 ac3_encoder_select="mdct ac3dsp dsputil"
 ac3_fixed_encoder_select="mdct ac3dsp dsputil"
 aic_decoder_select="dsputil golomb"
index e9087df..9f17d48 100644 (file)
@@ -820,7 +820,7 @@ following image formats are supported:
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
-@item AC-3                   @tab IX  @tab  X
+@item AC-3                   @tab IX  @tab  IX
 @item ADPCM 4X Movie         @tab     @tab  X
 @item ADPCM CDROM XA         @tab     @tab  X
 @item ADPCM Creative Technology @tab     @tab  X
index 70a6e25..f68371a 100644 (file)
@@ -94,7 +94,8 @@ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                           psymodel.o iirfilter.o \
                                           mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
-OBJS-$(CONFIG_AC3_DECODER)             += ac3dec.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
+OBJS-$(CONFIG_AC3_FIXED_DECODER)       += ac3dec_fixed.o ac3dec_data.o ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_ENCODER)             += ac3enc_float.o ac3enc.o ac3tab.o \
                                           ac3.o kbdwin.o
 OBJS-$(CONFIG_AC3_FIXED_ENCODER)       += ac3enc_fixed.o ac3enc.o ac3tab.o ac3.o
index 0cc9e2c..ac3936d 100644 (file)
 #define EXP_D25   2
 #define EXP_D45   3
 
+#ifndef CONFIG_AC3_FIXED
+#define CONFIG_AC3_FIXED 0
+#endif
+
+#if CONFIG_AC3_FIXED
+
+#define FFT_FLOAT 0
+
+#define FIXR(a)                 ((int)((a) * 0 + 0.5))
+#define FIXR12(a)               ((int)((a) * 4096 + 0.5))
+#define FIXR15(a)               ((int)((a) * 32768 + 0.5))
+#define ROUND15(x)              ((x) + 16384) >> 15
+
+#define AC3_RENAME(x)           x ## _fixed
+#define AC3_NORM(norm)          (1<<24)/(norm)
+#define AC3_MUL(a,b)            ((((int64_t) (a)) * (b))>>12)
+#define AC3_RANGE(x)            (x)
+#define AC3_DYNAMIC_RANGE(x)    (x)
+#define AC3_SPX_BLEND(x)        (x)
+#define AC3_DYNAMIC_RANGE1      0
+
+#define INTFLOAT                int
+#define SHORTFLOAT              int16_t
+
+#else /* CONFIG_AC3_FIXED */
+
+#define FIXR(x)                 ((float)(x))
+#define FIXR12(x)               ((float)(x))
+#define FIXR15(x)               ((float)(x))
+#define ROUND15(x)              (x)
+
+#define AC3_RENAME(x)           x
+#define AC3_NORM(norm)          (1.0f/(norm))
+#define AC3_MUL(a,b)            ((a) * (b))
+#define AC3_RANGE(x)            (dynamic_range_tab[(x)])
+#define AC3_DYNAMIC_RANGE(x)    (powf(x,  s->drc_scale))
+#define AC3_SPX_BLEND(x)        (x)* (1.0f/32)
+#define AC3_DYNAMIC_RANGE1      1.0f
+
+#define INTFLOAT                float
+#define SHORTFLOAT              float
+
+#endif /* CONFIG_AC3_FIXED */
+
+#define AC3_LEVEL(x)            ROUND15((x) * FIXR15(0.7071067811865476))
+
 /* pre-defined gain values */
 #define LEVEL_PLUS_3DB          1.4142135623730950
 #define LEVEL_PLUS_1POINT5DB    1.1892071150027209
index 3a6a7ad..1d97df9 100644 (file)
@@ -179,14 +179,23 @@ 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_dsputil_init(&s->dsp, avctx);
+
+#if (CONFIG_AC3_FIXED)
+    s->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & CODEC_FLAG_BITEXACT);
+#else
     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+#endif
+
     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);
 
-    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+    if (CONFIG_AC3_FIXED)
+        avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+    else
+        avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
 
     /* allow downmixing to stereo or mono */
 #if FF_API_REQUEST_CHANNELS
@@ -345,40 +354,45 @@ 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]);
     }
 }
 
@@ -646,20 +660,30 @@ 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);
+#if CONFIG_AC3_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]);
+#if CONFIG_AC3_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));
         }
     }
 }
@@ -794,13 +818,13 @@ 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)];
+            INTFLOAT range = AC3_RANGE(get_bits(gbc, 8));
             if (range > 1.0 || s->drc_scale <= 1.0)
-                s->dynamic_range[i] = powf(range, s->drc_scale);
+                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--);
 
@@ -826,6 +850,9 @@ 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 CONFIG_AC3_FIXED
+            s->spx_dst_end_freq = end_freq_inv_tab[end_subband];
+#endif
             if (end_subband   > 7)
                 end_subband   += end_subband   - 7;
             dst_start_freq = dst_start_freq * 12 + 25;
@@ -846,7 +873,9 @@ 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;
+#if !CONFIG_AC3_FIXED
             s->spx_dst_end_freq   = dst_end_freq;
+#endif
 
             decode_band_structure(gbc, blk, s->eac3, 0,
                                   start_subband, end_subband,
@@ -866,18 +895,40 @@ 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 spx_coord_exp, spx_coord_mant;
-                        float nratio, sblend, nblend, spx_coord;
+                        INTFLOAT nratio, sblend, nblend;
+#if CONFIG_AC3_FIXED
+                        int64_t accu;
+                        /* calculate blending factors */
+                        bandsize = s->spx_band_sizes[bnd];
+                        accu = (int64_t)((bin << 23) + (bandsize << 22)) * 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 = 0x800000;
+                            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];
@@ -886,6 +937,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
                         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 */
@@ -894,11 +946,18 @@ 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 CONFIG_AC3_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 {
@@ -1255,14 +1314,19 @@ 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];
+        INTFLOAT gain;
+        if(s->channel_mode == AC3_CHMODE_DUALMONO) {
+            gain = s->dynamic_range[2-ch];
         } else {
-            gain *= s->dynamic_range[0];
+            gain = s->dynamic_range[0];
         }
+#if CONFIG_AC3_FIXED
+        scale_coefs(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256);
+#else
+        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 */
@@ -1287,19 +1351,24 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
         do_imdct(s, s->channels);
 
         if (downmix_output) {
+#if CONFIG_AC3_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);
@@ -1320,7 +1389,7 @@ 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;
 
@@ -1447,7 +1516,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
     }
     for (ch = 0; ch < s->channels; ch++) {
         if (ch < s->out_channels)
-            s->outptr[channel_map[ch]] = (float *)frame->data[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)) {
@@ -1456,7 +1525,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
         }
         if (err)
             for (ch = 0; ch < s->out_channels; ch++)
-                memcpy(((float*)frame->data[ch]) + AC3_BLOCK_SIZE*blk, 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++) {
@@ -1469,7 +1538,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data,
 
     /* 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
@@ -1540,66 +1609,12 @@ 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);
+#if (CONFIG_AC3_FIXED)
+    av_free(s->fdsp);
+#endif
 
     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 },
-
-{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
-{"ltrt_cmixlev",   "Lt/Rt Center Mix Level",   OFFSET(ltrt_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_cmixlev",   "Lo/Ro Center Mix Level",   OFFSET(loro_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
-
-    { 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   = 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   = CODEC_CAP_DR1,
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
-                                                      AV_SAMPLE_FMT_NONE },
-    .priv_class     = &eac3_decoder_class,
-};
-#endif
index 58d8ee6..255b9df 100644 (file)
@@ -51,6 +51,7 @@
 #define AVCODEC_AC3DEC_H
 
 #include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
 #include "libavutil/lfg.h"
 #include "ac3.h"
 #include "ac3dsp.h"
@@ -138,8 +139,8 @@ typedef struct AC3DecodeContext {
     int num_spx_bands;                          ///< number of spx bands                    (nspxbnds)
     uint8_t spx_band_sizes[SPX_MAX_BANDS];      ///< number of bins in each spx band
     uint8_t first_spx_coords[AC3_MAX_CHANNELS]; ///< first spx coordinates states           (firstspxcos)
-    float spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor  (nblendfact)
-    float spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
+    INTFLOAT spx_noise_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS]; ///< spx noise blending factor  (nblendfact)
+    INTFLOAT spx_signal_blend[AC3_MAX_CHANNELS][SPX_MAX_BANDS];///< spx signal blending factor (sblendfact)
 ///@}
 
 ///@name Adaptive hybrid transform
@@ -151,15 +152,15 @@ typedef struct AC3DecodeContext {
     int fbw_channels;                           ///< number of full-bandwidth channels
     int channels;                               ///< number of total channels
     int lfe_ch;                                 ///< index of LFE channel
-    float downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
+    SHORTFLOAT downmix_coeffs[AC3_MAX_CHANNELS][2];  ///< stereo downmix coefficients
     int downmixed;                              ///< indicates if coeffs are currently downmixed
     int output_mode;                            ///< output channel configuration
     int out_channels;                           ///< number of output channels
 ///@}
 
 ///@name Dynamic range
-    float dynamic_range[2];                 ///< dynamic range
-    float drc_scale;                        ///< percentage of dynamic range compression to be applied
+    INTFLOAT dynamic_range[2];                 ///< dynamic range
+    INTFLOAT drc_scale;                        ///< percentage of dynamic range compression to be applied
 ///@}
 
 ///@name Bandwidth
@@ -207,22 +208,26 @@ typedef struct AC3DecodeContext {
 
 ///@name Optimization
     DSPContext dsp;                         ///< for optimization
+#if CONFIG_AC3_FIXED
+    AVFixedDSPContext *fdsp;
+#else
     AVFloatDSPContext fdsp;
+#endif
     AC3DSPContext ac3dsp;
     FmtConvertContext fmt_conv;             ///< optimized conversion functions
 ///@}
 
-    float *outptr[AC3_MAX_CHANNELS];
-    float *xcfptr[AC3_MAX_CHANNELS];
-    float *dlyptr[AC3_MAX_CHANNELS];
+    SHORTFLOAT *outptr[AC3_MAX_CHANNELS];
+    INTFLOAT *xcfptr[AC3_MAX_CHANNELS];
+    INTFLOAT *dlyptr[AC3_MAX_CHANNELS];
 
 ///@name Aligned arrays
-    DECLARE_ALIGNED(16, int32_t, fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];     ///< fixed-point transform coefficients
-    DECLARE_ALIGNED(32, float, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
-    DECLARE_ALIGNED(32, float, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
-    DECLARE_ALIGNED(32, float, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
-    DECLARE_ALIGNED(32, float, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
-    DECLARE_ALIGNED(32, float, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
+    DECLARE_ALIGNED(16, int,   fixed_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];       ///< fixed-point transform coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, transform_coeffs)[AC3_MAX_CHANNELS][AC3_MAX_COEFS];   ///< transform coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, delay)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];             ///< delay - added to the next block
+    DECLARE_ALIGNED(32, INTFLOAT, window)[AC3_BLOCK_SIZE];                              ///< window coefficients
+    DECLARE_ALIGNED(32, INTFLOAT, tmp_output)[AC3_BLOCK_SIZE];                          ///< temporary storage for output before windowing
+    DECLARE_ALIGNED(32, SHORTFLOAT, output)[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE];            ///< output after imdct transform and windowing
     DECLARE_ALIGNED(32, uint8_t, input_buffer)[AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; ///< temp buffer to prevent overread
 ///@}
 } AC3DecodeContext;
diff --git a/libavcodec/ac3dec_fixed.c b/libavcodec/ac3dec_fixed.c
new file mode 100644 (file)
index 0000000..1eb9a0b
--- /dev/null
@@ -0,0 +1,176 @@
+/*
+ * Copyright (c) 2012
+ *      MIPS Technologies, Inc., California.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
+ *    contributors may be used to endorse or promote products derived from
+ *    this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * Author:  Stanislav Ocovaj (socovaj@mips.com)
+ *
+ * AC3 fixed-point decoder for MIPS platforms
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#define FFT_FLOAT 0
+#define CONFIG_AC3_FIXED 1
+#define FFT_FIXED_32 1
+#include "ac3dec.h"
+
+
+/**
+ * Table for center mix levels
+ * reference: Section 5.4.2.4 cmixlev
+ */
+static const uint8_t center_levels[4] = { 4, 5, 6, 5 };
+
+/**
+ * Table for surround mix levels
+ * reference: Section 5.4.2.5 surmixlev
+ */
+static const uint8_t surround_levels[4] = { 4, 6, 7, 6 };
+
+int end_freq_inv_tab[8] =
+{
+    50529027, 44278013, 39403370, 32292987, 27356480, 23729101, 20951060, 18755316
+};
+
+static void scale_coefs (
+    int32_t *dst,
+    const int32_t *src,
+    int dynrng,
+    int len)
+{
+    int i, shift, round;
+    int16_t mul;
+    int temp, temp1, temp2, temp3, temp4, temp5, temp6, temp7;
+
+    mul = (dynrng & 0x1f) + 0x20;
+    shift = 4 - ((dynrng << 24) >> 29);
+    round = 1 << (shift-1);
+    for (i=0; i<len; i+=8) {
+
+        temp = src[i] * mul;
+        temp1 = src[i+1] * mul;
+        temp = temp + round;
+        temp2 = src[i+2] * mul;
+
+        temp1 = temp1 + round;
+        dst[i] = temp >> shift;
+        temp3 = src[i+3] * mul;
+        temp2 = temp2 + round;
+
+        dst[i+1] = temp1 >> shift;
+        temp4 = src[i + 4] * mul;
+        temp3 = temp3 + round;
+        dst[i+2] = temp2 >> shift;
+
+        temp5 = src[i+5] * mul;
+        temp4 = temp4 + round;
+        dst[i+3] = temp3 >> shift;
+        temp6 = src[i+6] * mul;
+
+        dst[i+4] = temp4 >> shift;
+        temp5 = temp5 + round;
+        temp7 = src[i+7] * mul;
+        temp6 = temp6 + round;
+
+        dst[i+5] = temp5 >> shift;
+        temp7 = temp7 + round;
+        dst[i+6] = temp6 >> shift;
+        dst[i+7] = temp7 >> shift;
+
+    }
+}
+
+/**
+ * Downmix samples from original signal to stereo or mono (this is for 16-bit samples
+ * and fixed point decoder - original (for 32-bit samples) is in ac3dsp.c).
+ */
+static void ac3_downmix_c_fixed16(int16_t **samples, int16_t (*matrix)[2],
+                                  int out_ch, int in_ch, int len)
+{
+    int i, j;
+    int v0, v1;
+    if (out_ch == 2) {
+        for (i = 0; i < len; i++) {
+            v0 = v1 = 0;
+            for (j = 0; j < in_ch; j++) {
+                v0 += samples[j][i] * matrix[j][0];
+                v1 += samples[j][i] * matrix[j][1];
+            }
+            samples[0][i] = (v0+2048)>>12;
+            samples[1][i] = (v1+2048)>>12;
+        }
+    } else if (out_ch == 1) {
+        for (i = 0; i < len; i++) {
+            v0 = 0;
+            for (j = 0; j < in_ch; j++)
+                v0 += samples[j][i] * matrix[j][0];
+            samples[0][i] = (v0+2048)>>12;
+        }
+    }
+}
+
+#include "ac3dec.c"
+
+static const AVOption options[] = {
+    { NULL},
+};
+
+static const AVClass ac3_decoder_class = {
+    .class_name = "Fixed-Point AC-3 Decoder",
+    .item_name  = av_default_item_name,
+    .option     = options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_ac3_fixed_decoder = {
+    .name           = "ac3_fixed",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_AC3,
+    .priv_data_size = sizeof (AC3DecodeContext),
+    .init           = ac3_decode_init,
+    .close          = ac3_decode_end,
+    .decode         = ac3_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &ac3_decoder_class,
+};
diff --git a/libavcodec/ac3dec_float.c b/libavcodec/ac3dec_float.c
new file mode 100644 (file)
index 0000000..7108921
--- /dev/null
@@ -0,0 +1,89 @@
+/*
+ * AC-3 Audio Decoder
+ * This code was developed as part of Google Summer of Code 2006.
+ * E-AC-3 support was added as part of Google Summer of Code 2007.
+ *
+ * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com)
+ * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com>
+ * Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * Upmix delay samples from stereo to original channel layout.
+ */
+#include "ac3dec.h"
+#include "ac3dec.c"
+
+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 },
+
+{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 2, 0, "dmix_mode"},
+{"ltrt_cmixlev",   "Lt/Rt Center Mix Level",   OFFSET(ltrt_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_cmixlev",   "Lo/Ro Center Mix Level",   OFFSET(loro_center_mix_level),    AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level),  AV_OPT_TYPE_FLOAT, {.dbl = -1.0 }, -1.0, 2.0, 0},
+
+    { 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",
+    .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   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
+    .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",
+    .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   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
+    .priv_class     = &eac3_decoder_class,
+};
+#endif
index 0d5ea39..b746817 100644 (file)
@@ -239,6 +239,31 @@ static void ac3_downmix_c(float **samples, float (*matrix)[2],
     }
 }
 
+static void ac3_downmix_c_fixed(int32_t **samples, int16_t (*matrix)[2],
+                                int out_ch, int in_ch, int len)
+{
+    int i, j;
+    int64_t v0, v1;
+    if (out_ch == 2) {
+        for (i = 0; i < len; i++) {
+            v0 = v1 = 0;
+            for (j = 0; j < in_ch; j++) {
+                v0 += (int64_t)samples[j][i] * matrix[j][0];
+                v1 += (int64_t)samples[j][i] * matrix[j][1];
+            }
+            samples[0][i] = (v0+2048)>>12;
+            samples[1][i] = (v1+2048)>>12;
+        }
+    } else if (out_ch == 1) {
+        for (i = 0; i < len; i++) {
+            v0 = 0;
+            for (j = 0; j < in_ch; j++)
+                v0 += (int64_t)samples[j][i] * matrix[j][0];
+            samples[0][i] = (v0+2048)>>12;
+        }
+    }
+}
+
 static void apply_window_int16_c(int16_t *output, const int16_t *input,
                                  const int16_t *window, unsigned int len)
 {
@@ -266,6 +291,7 @@ av_cold void ff_ac3dsp_init(AC3DSPContext *c, int bit_exact)
     c->sum_square_butterfly_int32 = ac3_sum_square_butterfly_int32_c;
     c->sum_square_butterfly_float = ac3_sum_square_butterfly_float_c;
     c->downmix = ac3_downmix_c;
+    c->downmix_fixed = ac3_downmix_c_fixed;
     c->apply_window_int16 = apply_window_int16_c;
 
     if (ARCH_ARM)
index f18583c..ed98c8c 100644 (file)
@@ -135,6 +135,9 @@ typedef struct AC3DSPContext {
     void (*downmix)(float **samples, float (*matrix)[2], int out_ch,
                     int in_ch, int len);
 
+    void (*downmix_fixed)(int32_t **samples, int16_t (*matrix)[2], int out_ch,
+                          int in_ch, int len);
+
     /**
      * Apply symmetric window in 16-bit fixed-point.
      * @param output destination array
index 10edd39..c55a809 100644 (file)
@@ -323,7 +323,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC (AAC,               aac);
     REGISTER_DECODER(AAC_LATM,          aac_latm);
     REGISTER_ENCDEC (AC3,               ac3);
-    REGISTER_ENCODER(AC3_FIXED,         ac3_fixed);
+    REGISTER_ENCDEC (AC3_FIXED,         ac3_fixed);
     REGISTER_ENCDEC (ALAC,              alac);
     REGISTER_DECODER(ALS,               als);
     REGISTER_DECODER(AMRNB,             amrnb);
index 5a62e9d..bf32aeb 100644 (file)
@@ -45,3 +45,13 @@ av_cold void ff_kbd_window_init(float *window, float alpha, int n)
    for (i = 0; i < n; i++)
        window[i] = sqrt(local_window[i] / sum);
 }
+
+av_cold void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n)
+{
+    int i;
+    float local_window[FF_KBD_WINDOW_MAX];
+
+    ff_kbd_window_init(local_window, alpha, n);
+    for (i = 0; i < n; i++)
+        window[i] = (int)floor(2147483647.0 * local_window[i] + 0.5);
+}
index 4b93975..2e02e10 100644 (file)
@@ -31,5 +31,6 @@
  * @param   n       size of half window, max FF_KBD_WINDOW_MAX
  */
 void ff_kbd_window_init(float *window, float alpha, int n);
+void ff_kbd_window_init_fixed(int32_t *window, float alpha, int n);
 
 #endif /* AVCODEC_KBDWIN_H */
index 21f5dea..19b4934 100644 (file)
@@ -29,7 +29,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR 55
-#define LIBAVCODEC_VERSION_MINOR  55
+#define LIBAVCODEC_VERSION_MINOR  56
 #define LIBAVCODEC_VERSION_MICRO 107
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \