rawdec: Set start_time to 0 for raw audio files.

[ffmpeg] / libavcodec / adxenc.c

diff --git a/libavcodec/adxenc.c b/libavcodec/adxenc.c
index 296ccbd58aa3b266d75d98d0fd8a2e50eb4dd838..ae4b9f7da20d6d2a64f96085c64946d302c837b2 100644 (file)
--- a/libavcodec/adxenc.c
+++ b/libavcodec/adxenc.c
@@ -22,6 +22,7 @@
 #include "libavutil/intreadwrite.h"
 #include "avcodec.h"
 #include "adx.h"
+#include "put_bits.h"
 
 /**
  * @file
@@ -34,8 +35,10 @@
 
 /* 18 bytes <-> 32 samples */
 
-static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
+static void adx_encode(ADXContext *c, unsigned char *adx, const short *wav,
+                       ADXChannelState *prev)
 {
+    PutBitContext pb;
     int scale;
     int i;
     int s0,s1,s2,d;
@@ -47,7 +50,7 @@ static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
     s2 = prev->s2;
     for(i=0;i<32;i++) {
         s0 = wav[i];
-        d = ((s0<<14) - SCALE1*s1 + SCALE2*s2)/BASEVOL;
+        d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
         data[i]=d;
         if (max<d) max=d;
         if (min>d) min=d;
@@ -71,9 +74,10 @@ static void adx_encode(unsigned char *adx,const short *wav,PREV *prev)
 
     AV_WB16(adx, scale);
 
-    for(i=0;i<16;i++) {
-        adx[i+2] = ((data[i*2]/scale)<<4) | ((data[i*2+1]/scale)&0xf);
-    }
+    init_put_bits(&pb, adx + 2, 16);
+    for (i = 0; i < 32; i++)
+        put_sbits(&pb, 4, av_clip(data[i]/scale, -8, 7));
+    flush_put_bits(&pb);
 }
 
 static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t bufsize)
@@ -101,19 +105,24 @@ static int adx_encode_header(AVCodecContext *avctx,unsigned char *buf,size_t buf
     } adxhdr; /* big endian */
     /* offset-6 "(c)CRI" */
 #endif
+    ADXContext *c = avctx->priv_data;
+
     AV_WB32(buf+0x00,0x80000000|0x20);
     AV_WB32(buf+0x04,0x03120400|avctx->channels);
     AV_WB32(buf+0x08,avctx->sample_rate);
     AV_WB32(buf+0x0c,0); /* FIXME: set after */
-    AV_WB32(buf+0x10,0x01040300);
-    AV_WB32(buf+0x14,0x00000000);
-    AV_WB32(buf+0x18,0x00000000);
-    memcpy(buf+0x1c,"\0\0(c)CRI",8);
+    AV_WB16(buf + 0x10, c->cutoff);
+    AV_WB32(buf + 0x12, 0x03000000);
+    AV_WB32(buf + 0x16, 0x00000000);
+    AV_WB32(buf + 0x1a, 0x00000000);
+    memcpy (buf + 0x1e, "(c)CRI", 6);
     return 0x20+4;
 }
 
 static av_cold int adx_encode_init(AVCodecContext *avctx)
 {
+    ADXContext *c = avctx->priv_data;
+
     if (avctx->channels > 2)
         return -1; /* only stereo or mono =) */
     avctx->frame_size = 32;
@@ -123,6 +132,10 @@ static av_cold int adx_encode_init(AVCodecContext *avctx)
 
 //    avctx->bit_rate = avctx->sample_rate*avctx->channels*18*8/32;
 
+    /* the cutoff can be adjusted, but this seems to work pretty well */
+    c->cutoff = 500;
+    ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
+
     av_log(avctx, AV_LOG_DEBUG, "adx encode init\n");
 
     return 0;
@@ -145,7 +158,7 @@ static int adx_encode_frame(AVCodecContext *avctx,
 
 /*
     input data size =
-    ffmpeg.c: do_audio_out()
+    avconv.c:do_audio_out()
     frame_bytes = enc->frame_size * 2 * enc->channels;
 */
 
@@ -158,7 +171,7 @@ static int adx_encode_frame(AVCodecContext *avctx,
 
     if (avctx->channels==1) {
         while(rest>=32) {
-            adx_encode(dst,samples,c->prev);
+            adx_encode(c, dst, samples, c->prev);
             dst+=18;
             samples+=32;
             rest-=32;
@@ -173,8 +186,8 @@ static int adx_encode_frame(AVCodecContext *avctx,
                 tmpbuf[i+32] = samples[i*2+1];
             }
 
-            adx_encode(dst,tmpbuf,c->prev);
-            adx_encode(dst+18,tmpbuf+32,c->prev+1);
+            adx_encode(c, dst,      tmpbuf,      c->prev);
+            adx_encode(c, dst + 18, tmpbuf + 32, c->prev + 1);
             dst+=18*2;
             samples+=32*2;
             rest-=32*2;