lavc: add a wrapper for AVCodecContext.get_buffer().

[ffmpeg] / libavcodec / flacdsp.c

diff --git a/libavcodec/flacdsp.c b/libavcodec/flacdsp.c
index b7e074644ee24db8cd78ad8e66e4bc0fb72b9f62..b9168693218aca5eecfdfe4d9813aa6777a32206 100644 (file)
--- a/libavcodec/flacdsp.c
+++ b/libavcodec/flacdsp.c
@@ -21,16 +21,81 @@
 #include "libavutil/attributes.h"
 #include "libavutil/samplefmt.h"
 #include "flacdsp.h"
+#include "config.h"
 
 #define SAMPLE_SIZE 16
+#define PLANAR 0
+#include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
+
+#undef  PLANAR
+#define PLANAR 1
 #include "flacdsp_template.c"
 
 #undef  SAMPLE_SIZE
+#undef  PLANAR
 #define SAMPLE_SIZE 32
+#define PLANAR 0
+#include "flacdsp_template.c"
+#include "flacdsp_lpc_template.c"
+
+#undef  PLANAR
+#define PLANAR 1
 #include "flacdsp_template.c"
 
-av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
+static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
+                          int pred_order, int qlevel, int len)
+{
+    int i, j;
+
+    for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
+        int c = coeffs[0];
+        int d = decoded[0];
+        int s0 = 0, s1 = 0;
+        for (j = 1; j < pred_order; j++) {
+            s0 += c*d;
+            d = decoded[j];
+            s1 += c*d;
+            c = coeffs[j];
+        }
+        s0 += c*d;
+        d = decoded[j] += s0 >> qlevel;
+        s1 += c*d;
+        decoded[j + 1] += s1 >> qlevel;
+    }
+    if (i < len) {
+        int sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += coeffs[j] * decoded[j];
+        decoded[j] += sum >> qlevel;
+    }
+}
+
+static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
+                          int pred_order, int qlevel, int len)
 {
+    int i, j;
+
+    for (i = pred_order; i < len; i++, decoded++) {
+        int64_t sum = 0;
+        for (j = 0; j < pred_order; j++)
+            sum += (int64_t)coeffs[j] * decoded[j];
+        decoded[j] += sum >> qlevel;
+    }
+
+}
+
+av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
+                             int bps)
+{
+    if (bps > 16) {
+        c->lpc            = flac_lpc_32_c;
+        c->lpc_encode     = flac_lpc_encode_c_32;
+    } else {
+        c->lpc            = flac_lpc_16_c;
+        c->lpc_encode     = flac_lpc_encode_c_16;
+    }
+
     switch (fmt) {
     case AV_SAMPLE_FMT_S32:
         c->decorrelate[0] = flac_decorrelate_indep_c_32;
@@ -39,11 +104,28 @@ av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt)
         c->decorrelate[3] = flac_decorrelate_ms_c_32;
         break;
 
+    case AV_SAMPLE_FMT_S32P:
+        c->decorrelate[0] = flac_decorrelate_indep_c_32p;
+        c->decorrelate[1] = flac_decorrelate_ls_c_32p;
+        c->decorrelate[2] = flac_decorrelate_rs_c_32p;
+        c->decorrelate[3] = flac_decorrelate_ms_c_32p;
+        break;
+
     case AV_SAMPLE_FMT_S16:
         c->decorrelate[0] = flac_decorrelate_indep_c_16;
         c->decorrelate[1] = flac_decorrelate_ls_c_16;
         c->decorrelate[2] = flac_decorrelate_rs_c_16;
         c->decorrelate[3] = flac_decorrelate_ms_c_16;
         break;
+
+    case AV_SAMPLE_FMT_S16P:
+        c->decorrelate[0] = flac_decorrelate_indep_c_16p;
+        c->decorrelate[1] = flac_decorrelate_ls_c_16p;
+        c->decorrelate[2] = flac_decorrelate_rs_c_16p;
+        c->decorrelate[3] = flac_decorrelate_ms_c_16p;
+        break;
     }
+
+    if (ARCH_ARM)
+        ff_flacdsp_init_arm(c, fmt, bps);
 }