Replace remaining occurrences of CODEC_TYPE_* with AVMEDIA_TYPE*

[ffmpeg] / libavcodec / dca.c

diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index bc099e1376cbd78c4fa239ea9910ee2a7212bdc9..63ea32992e3c97175b29552aa5682a37969cef3b 100644 (file)
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -40,6 +40,7 @@
 #include "dca.h"
 #include "synth_filter.h"
 #include "dcadsp.h"
+#include "fmtconvert.h"
 
 //#define TRACE
 
@@ -311,7 +312,6 @@ typedef struct {
     DECLARE_ALIGNED(16, float, raXin)[32];
 
     int output;                 ///< type of output
-    float add_bias;             ///< output bias
     float scale_bias;           ///< output scale
 
     DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
@@ -348,6 +348,7 @@ typedef struct {
     FFTContext imdct;
     SynthFilterContext synth;
     DCADSPContext dcadsp;
+    FmtConvertContext fmt_conv;
 } DCAContext;
 
 static const uint16_t dca_vlc_offs[] = {
@@ -868,7 +869,7 @@ static int dca_subframe_header(DCAContext * s, int base_channel, int block_index
 
 static void qmf_32_subbands(DCAContext * s, int chans,
                             float samples_in[32][8], float *samples_out,
-                            float scale, float bias)
+                            float scale)
 {
     const float *prCoeff;
     int i;
@@ -897,7 +898,7 @@ static void qmf_32_subbands(DCAContext * s, int chans,
         s->synth.synth_filter_float(&s->imdct,
                               s->subband_fir_hist[chans], &s->hist_index[chans],
                               s->subband_fir_noidea[chans], prCoeff,
-                              samples_out, s->raXin, scale, bias);
+                              samples_out, s->raXin, scale);
         samples_out+= 32;
 
     }
@@ -905,8 +906,7 @@ static void qmf_32_subbands(DCAContext * s, int chans,
 
 static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
                                   int num_deci_sample, float *samples_in,
-                                  float *samples_out, float scale,
-                                  float bias)
+                                  float *samples_out, float scale)
 {
     /* samples_in: An array holding decimated samples.
      *   Samples in current subframe starts from samples_in[0],
@@ -931,7 +931,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
     /* Interpolation */
     for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
         s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor,
-                          scale, bias);
+                          scale);
         samples_in++;
         samples_out += 2 * decifactor;
     }
@@ -939,19 +939,19 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
 
 /* downmixing routines */
 #define MIX_REAR1(samples, si1, rs, coef) \
-     samples[i]     += (samples[si1] - add_bias) * coef[rs][0];  \
-     samples[i+256] += (samples[si1] - add_bias) * coef[rs][1];
+     samples[i]     += samples[si1] * coef[rs][0];  \
+     samples[i+256] += samples[si1] * coef[rs][1];
 
 #define MIX_REAR2(samples, si1, si2, rs, coef) \
-     samples[i]     += (samples[si1] - add_bias) * coef[rs][0] + (samples[si2] - add_bias) * coef[rs+1][0]; \
-     samples[i+256] += (samples[si1] - add_bias) * coef[rs][1] + (samples[si2] - add_bias) * coef[rs+1][1];
+     samples[i]     += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
+     samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
 
 #define MIX_FRONT3(samples, coef) \
-    t = samples[i+c] - add_bias; \
-    u = samples[i+l] - add_bias; \
-    v = samples[i+r] - add_bias; \
-    samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0] + add_bias; \
-    samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1] + add_bias;
+    t = samples[i+c]; \
+    u = samples[i+l]; \
+    v = samples[i+r]; \
+    samples[i]     = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
+    samples[i+256] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
 
 #define DOWNMIX_TO_STEREO(op1, op2) \
     for (i = 0; i < 256; i++){ \
@@ -961,7 +961,7 @@ static void lfe_interpolation_fir(DCAContext *s, int decimation_select,
 
 static void dca_downmix(float *samples, int srcfmt,
                         int downmix_coef[DCA_PRIM_CHANNELS_MAX][2],
-                        const int8_t *channel_mapping, float add_bias)
+                        const int8_t *channel_mapping)
 {
     int c,l,r,sl,sr,s;
     int i;
@@ -1117,7 +1117,7 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
                         block[m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
                 }
 
-                s->dsp.int32_to_float_fmul_scalar(subband_samples[k][l],
+                s->fmt_conv.int32_to_float_fmul_scalar(subband_samples[k][l],
                                                   block, rscale, 8);
             }
 
@@ -1193,13 +1193,12 @@ static int dca_filter_channels(DCAContext * s, int block_index)
 /*        static float pcm_to_double[8] =
             {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
          qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]],
-                            M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
-                            s->add_bias );
+                         M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ );
     }
 
     /* Down mixing */
     if (s->avctx->request_channels == 2 && s->prim_channels > 2) {
-        dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab, s->add_bias);
+        dca_downmix(s->samples, s->amode, s->downmix_coef, s->channel_order_tab);
     }
 
     /* Generate LFE samples for this subsubframe FIXME!!! */
@@ -1207,7 +1206,7 @@ static int dca_filter_channels(DCAContext * s, int block_index)
         lfe_interpolation_fir(s, s->lfe, 2 * s->lfe,
                               s->lfe_data + 2 * s->lfe * (block_index + 4),
                               &s->samples[256 * dca_lfe_index[s->amode]],
-                              (1.0/256.0)*s->scale_bias,  s->add_bias);
+                              (1.0/256.0)*s->scale_bias);
         /* Outputs 20bits pcm samples */
     }
 
@@ -1357,7 +1356,9 @@ static int dca_exss_mask2count(int mask)
  */
 static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch)
 {
-    for (int i = 0; i < channels; i++) {
+    int i;
+
+    for (i = 0; i < channels; i++) {
         int mix_map_mask = get_bits(gb, out_ch);
         int num_coeffs = av_popcount(mix_map_mask);
         skip_bits_long(gb, num_coeffs * 6);
@@ -1501,12 +1502,9 @@ static int dca_exss_parse_asset_header(DCAContext *s)
 
     if (extensions_mask & DCA_EXT_EXSS_XLL)
         s->profile = FF_PROFILE_DTS_HD_MA;
-    else if (extensions_mask & DCA_EXT_EXSS_XBR)
+    else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 |
+                                DCA_EXT_EXSS_XXCH))
         s->profile = FF_PROFILE_DTS_HD_HRA;
-    else if (extensions_mask & DCA_EXT_EXSS_X96)
-        s->profile = FF_PROFILE_DTS_96_24;
-    else if (extensions_mask & DCA_EXT_EXSS_XXCH)
-        s->profile = FFMAX(s->profile, FF_PROFILE_DTS_ES);
 
     if (!(extensions_mask & DCA_EXT_CORE))
         av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n");
@@ -1801,12 +1799,12 @@ static int dca_decode_frame(AVCodecContext * avctx,
             float* rt_chan   = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
             int j;
             for(j = 0; j < 256; ++j) {
-                lt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
-                rt_chan[j] -= (back_chan[j] - s->add_bias) * M_SQRT1_2;
+                lt_chan[j] -= back_chan[j] * M_SQRT1_2;
+                rt_chan[j] -= back_chan[j] * M_SQRT1_2;
             }
         }
 
-        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
+        s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
         samples += 256 * channels;
     }
 
@@ -1839,26 +1837,20 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
     ff_mdct_init(&s->imdct, 6, 1, 1.0);
     ff_synth_filter_init(&s->synth);
     ff_dcadsp_init(&s->dcadsp);
+    ff_fmt_convert_init(&s->fmt_conv, avctx);
 
     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
         s->samples_chanptr[i] = s->samples + i * 256;
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
 
-    if (s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
-        s->add_bias = 385.0f;
-        s->scale_bias = 1.0 / 32768.0;
-    } else {
-        s->add_bias = 0.0f;
-        s->scale_bias = 1.0;
+    s->scale_bias = 1.0;
 
-        /* allow downmixing to stereo */
-        if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
-                avctx->request_channels == 2) {
-            avctx->channels = avctx->request_channels;
-        }
+    /* allow downmixing to stereo */
+    if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
+        avctx->request_channels == 2) {
+        avctx->channels = avctx->request_channels;
     }
 
-
     return 0;
 }
 
@@ -1869,7 +1861,16 @@ static av_cold int dca_decode_end(AVCodecContext * avctx)
     return 0;
 }
 
-AVCodec dca_decoder = {
+static const AVProfile profiles[] = {
+    { FF_PROFILE_DTS,        "DTS"        },
+    { FF_PROFILE_DTS_ES,     "DTS-ES"     },
+    { FF_PROFILE_DTS_96_24,  "DTS 96/24"  },
+    { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" },
+    { FF_PROFILE_DTS_HD_MA,  "DTS-HD MA"  },
+    { FF_PROFILE_UNKNOWN },
+};
+
+AVCodec ff_dca_decoder = {
     .name = "dca",
     .type = AVMEDIA_TYPE_AUDIO,
     .id = CODEC_ID_DTS,
@@ -1879,4 +1880,5 @@ AVCodec dca_decoder = {
     .close = dca_decode_end,
     .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
     .capabilities = CODEC_CAP_CHANNEL_CONF,
+    .profiles = NULL_IF_CONFIG_SMALL(profiles),
 };