Libavcodec AC3/E-AC3/DTS decoders now output floating point data.

git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c

4 files changed, 75 insertions, 3 deletions

diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c
index 8a0f96a..b0e7647 100644
--- a/libavcodec/ac3dec.c
+++ b/libavcodec/ac3dec.c
@@ -188,8 +188,13 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
     ff_fmt_convert_init(&s->fmt_conv, avctx);
     av_lfg_init(&s->dith_state, 0);
 
+    /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+    s->mul_bias = 1.0f;
+#else
     /* set scale value for float to int16 conversion */
     s->mul_bias = 32767.0f;
+#endif
 
     /* allow downmixing to stereo or mono */
     if (avctx->channels > 0 && avctx->request_channels > 0 &&
@@ -204,7 +209,12 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
         if (!s->input_buffer)
             return AVERROR(ENOMEM);
 
+    /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+#else
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+#endif
     return 0;
 }
 
@@ -1299,7 +1309,12 @@ 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;
+    /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+    float *out_samples = (float *)data;
+#else
     int16_t *out_samples = (int16_t *)data;
+#endif
     int blk, ch, err;
     const uint8_t *channel_map;
     const float *output[AC3_MAX_CHANNELS];
@@ -1405,10 +1420,15 @@ 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;
         }
+        /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+        float_interleave_noscale(out_samples, output, 256, s->out_channels);
+#else
         s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
+#endif
         out_samples += 256 * s->out_channels;
     }
-    *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
+    *data_size = s->num_blocks * 256 * avctx->channels * sizeof (out_samples[0]); /* ffdshow custom code */
     return FFMIN(buf_size, s->frame_size);
 }
 
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index 14b0b2d..2053121 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -1626,7 +1626,12 @@ static int dca_decode_frame(AVCodecContext * avctx,
     int lfe_samples;
     int num_core_channels = 0;
     int i;
+    /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+    float *samples = data;
+#else
     int16_t *samples = data;
+#endif
     DCAContext *s = avctx->priv_data;
     int channels;
     int core_ss_end;
@@ -1812,9 +1817,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
         return -1;
     }
 
-    if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
+    /* ffdshow custom code */
+    if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(samples[0]) * channels)
         return -1;
-    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
+    *data_size = 256 / 8 * s->sample_blocks * sizeof(samples[0]) * channels;
 
     /* filter to get final output */
     for (i = 0; i < (s->sample_blocks / 8); i++) {
@@ -1833,7 +1839,13 @@ static int dca_decode_frame(AVCodecContext * avctx,
             }
         }
 
+        /* interleave samples */
+#if CONFIG_AUDIO_FLOAT
+        /* ffdshow custom code */
+        float_interleave(samples, s->samples_chanptr, 256, channels);
+#else
         s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
+#endif
         samples += 256 * channels;
     }
 
@@ -1870,7 +1882,12 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
 
     for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
         s->samples_chanptr[i] = s->samples + i * 256;
+    /* ffdshow custom code */
+#if CONFIG_AUDIO_FLOAT
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+#else
     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+#endif
 
     s->scale_bias = 1.0;
 
diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c
index e27c1f6..0e8aa5e 100644
--- a/libavcodec/fmtconvert.c
+++ b/libavcodec/fmtconvert.c
@@ -66,3 +66,34 @@ av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
     if (HAVE_ALTIVEC) ff_fmt_convert_init_altivec(c, avctx);
     if (HAVE_MMX) ff_fmt_convert_init_x86(c, avctx);
 }
+
+/* ffdshow custom code */
+void float_interleave(float *dst, const float **src, long len, int channels)
+{
+    int i,j,c;
+    if(channels==2){
+        for(i=0; i<len; i++){
+            dst[2*i]   = src[0][i] / 32768.0f;
+            dst[2*i+1] = src[1][i] / 32768.0f;
+        }
+    }else{
+        for(c=0; c<channels; c++)
+            for(i=0, j=c; i<len; i++, j+=channels)
+                dst[j] = src[c][i] / 32768.0f;
+    }
+}
+
+void float_interleave_noscale(float *dst, const float **src, long len, int channels)
+{
+    int i,j,c;
+    if(channels==2){
+        for(i=0; i<len; i++){
+            dst[2*i]   = src[0][i];
+            dst[2*i+1] = src[1][i];
+        }
+    }else{
+        for(c=0; c<channels; c++)
+            for(i=0, j=c; i<len; i++, j+=channels)
+                dst[j] = src[c][i];
+    }
+}
diff --git a/libavcodec/fmtconvert.h b/libavcodec/fmtconvert.h
index 7416184..82811d1 100644
--- a/libavcodec/fmtconvert.h
+++ b/libavcodec/fmtconvert.h
@@ -76,4 +76,8 @@ void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx);
 void ff_fmt_convert_init_altivec(FmtConvertContext *c, AVCodecContext *avctx);
 void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx);
 
+/* ffdshow custom code */
+void float_interleave(float *dst, const float **src, long len, int channels);
+void float_interleave_noscale(float *dst, const float **src, long len, int channels);
+
 #endif /* AVCODEC_FMTCONVERT_H */