DV audio decoder by Roman Shaposhnick

Originally committed as revision 1514 to svn://svn.ffmpeg.org/ffmpeg/trunk

3 files changed, 155 insertions, 4 deletions

diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index a5d2e41cfb..cc06645395 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -72,7 +72,7 @@ void avcodec_register_all(void)
     register_avcodec(&rv10_decoder);
     register_avcodec(&svq1_decoder);
     register_avcodec(&dvvideo_decoder);
-    //    register_avcodec(&dvaudio_decoder);
+    register_avcodec(&dvaudio_decoder);
     register_avcodec(&mjpeg_decoder);
     register_avcodec(&mjpegb_decoder);
     register_avcodec(&mp2_decoder);
diff --git a/libavcodec/dv.c b/libavcodec/dv.c
index f312c0264c..554b593e7a 100644
--- a/libavcodec/dv.c
+++ b/libavcodec/dv.c
@@ -634,7 +634,6 @@ AVCodec dvvideo_decoder = {
 typedef struct DVAudioDecodeContext {
     AVCodecContext *avctx;
     GetBitContext gb;
-
 } DVAudioDecodeContext;
 
 static int dvaudio_decode_init(AVCodecContext *avctx)
@@ -643,13 +642,126 @@ static int dvaudio_decode_init(AVCodecContext *avctx)
     return 0;
 }
 
+static UINT16 dv_audio_12to16(UINT16 sample)
+{
+    UINT16 shift, result;
+    
+    sample = (sample < 0x800) ? sample : sample | 0xf000;
+    shift = (sample & 0xf00) >> 8;
+
+    if (shift < 0x2 || shift > 0xd) {
+	result = sample;
+    } else if (shift < 0x8) {
+        shift--;
+	result = (sample - (256 * shift)) << shift;
+    } else {
+	shift = 0xe - shift;
+	result = ((sample + ((256 * shift) + 1)) << shift) - 1;
+    }
+
+    return result;
+}
+
 /* NOTE: exactly one frame must be given (120000 bytes for NTSC,
-   144000 bytes for PAL) */
+   144000 bytes for PAL) 
+
+   There's a couple of assumptions being made here:
+         1. We don't do any kind of audio error correction. It means,
+	    that erroneous samples 0x8000 are being passed upwards.
+            Do we need to silence erroneous samples ? Average them ?
+	 2. We don't do software emphasis.
+	 3. We are not checking for 'speed' argument being valid.
+	 4. Audio is always returned as 16bit linear samples: 12bit
+	    nonlinear samples are converted into 16bit linear ones.
+*/
 static int dvaudio_decode_frame(AVCodecContext *avctx, 
                                  void *data, int *data_size,
                                  UINT8 *buf, int buf_size)
 {
-    //    DVAudioDecodeContext *s = avctx->priv_data;
+    DVVideoDecodeContext *s = avctx->priv_data;
+    const UINT16 (*unshuffle)[9];
+    int smpls, freq, quant, sys, stride, difseg, ad, dp, nb_dif_segs, i;
+    UINT16 lc, rc;
+    UINT8 *buf_ptr;
+    
+    /* parse id */
+    init_get_bits(&s->gb, &buf[AAUX_OFFSET], 5*8);
+    i = get_bits(&s->gb, 8);
+    if (i != 0x50) { /* No audio ? */
+	*data_size = 0;
+	return buf_size;
+    }
+    
+    get_bits(&s->gb, 1); /* 0 - locked audio, 1 - unlocked audio */
+    skip_bits(&s->gb, 1);
+    smpls = get_bits(&s->gb, 6); /* samples in this frame - min. samples */
+
+    skip_bits(&s->gb, 8);
+
+    skip_bits(&s->gb, 2);
+    sys = get_bits(&s->gb, 1); /* 0 - 60 fields, 1 = 50 fields */
+    skip_bits(&s->gb, 5);
+
+    get_bits(&s->gb, 1); /* 0 - emphasis on, 1 - emphasis off */
+    get_bits(&s->gb, 1); /* 0 - reserved, 1 - emphasis time constant 50/15us */
+    freq = get_bits(&s->gb, 3); /* 0 - 48KHz, 1 - 44,1kHz, 2 - 32 kHz */
+    quant = get_bits(&s->gb, 3); /* 0 - 16bit linear, 1 - 12bit nonlinear */
+
+    if (quant > 1)
+	return -1; /* Unsupported quantization */
+
+    avctx->sample_rate = dv_audio_frequency[freq];
+    // What about:
+    // avctx->bit_rate = 
+    // avctx->frame_size =
+   
+    *data_size = (dv_audio_min_samples[sys][freq] + smpls) * 
+	         avctx->channels * 2;
+
+    if (sys) {
+	nb_dif_segs = 12;
+	stride = 108;
+	unshuffle = dv_place_audio50;
+    } else {
+	nb_dif_segs = 10;
+	stride = 90;
+	unshuffle = dv_place_audio60;
+    }
+    
+    /* for each DIF segment */
+    buf_ptr = buf;
+    for (difseg = 0; difseg < nb_dif_segs; difseg++) {
+         buf_ptr += 6 * 80; /* skip DIF segment header */
+         for (ad = 0; ad < 9; ad++) {
+              
+              for (dp = 8; dp < 80; dp+=2) {
+		   if (quant == 0) {  /* 16bit quantization */
+		       i = unshuffle[difseg][ad] + (dp - 8)/2 * stride;
+		       ((short *)data)[i] = (buf_ptr[dp] << 8) | buf_ptr[dp+1]; 
+		   } else {           /* 12bit quantization */
+		       if (difseg >= nb_dif_segs/2)
+			   goto out;  /* We're not doing 4ch at this time */
+		       
+		       lc = ((UINT16)buf_ptr[dp] << 4) | 
+			    ((UINT16)buf_ptr[dp+2] >> 4);
+		       rc = ((UINT16)buf_ptr[dp+1] << 4) |
+			    ((UINT16)buf_ptr[dp+2] & 0x0f);
+		       lc = dv_audio_12to16(lc);
+		       rc = dv_audio_12to16(rc);
+
+		       i = unshuffle[difseg][ad] + (dp - 8)/3 * stride;
+		       ((short *)data)[i] = lc;
+		       i = unshuffle[difseg+nb_dif_segs/2][ad] + (dp - 8)/3 * stride;
+		       ((short *)data)[i] = rc;
+		       ++dp;
+		   }
+	      }
+		
+	    buf_ptr += 16 * 80; /* 15 Video DIFs + 1 Audio DIF */
+        }
+    }
+
+out:
     return buf_size;
 }
 
diff --git a/libavcodec/dvdata.h b/libavcodec/dvdata.h
index b5c1f56077..4e1fc39c7f 100644
--- a/libavcodec/dvdata.h
+++ b/libavcodec/dvdata.h
@@ -18,6 +18,7 @@
  */
 
 #define NB_DV_VLC 409
+#define AAUX_OFFSET (80*6 + 80*16*3 + 3)
 
 static const UINT16 dv_vlc_bits[409] = {
  0x0000, 0x0002, 0x0007, 0x0008, 0x0009, 0x0014, 0x0015, 0x0016,
@@ -905,3 +906,41 @@ static const UINT16 dv_place_411[1350] = {
  0x0834, 0x2320, 0x2f44, 0x3810, 0x1658,
 };
 
+static const UINT16 dv_place_audio60[10][9] = {
+  {  0, 30, 60, 20, 50, 80, 10, 40, 70 }, /* 1st channel */
+  {  6, 36, 66, 26, 56, 86, 16, 46, 76 },
+  { 12, 42, 72,  2, 32, 62, 22, 52, 82 },
+  { 18, 48, 78,  8, 38, 68, 28, 58, 88 },
+  { 24, 54, 84, 14, 44, 74,  4, 34, 64 },
+  
+  {  1, 31, 61, 21, 51, 81, 11, 41, 71 }, /* 2nd channel */
+  {  7, 37, 67, 27, 57, 87, 17, 47, 77 },
+  { 13, 43, 73,  3, 33, 63, 23, 53, 83 },
+  { 19, 49, 79,  9, 39, 69, 29, 59, 89 },
+  { 25, 55, 85, 15, 45, 75,  5, 35, 65 },
+};
+
+static const UINT16 dv_place_audio50[12][9] = {
+  {   0,  36,  72,  26,  62,  98,  16,  52,  88}, /* 1st channel */
+  {   6,  42,  78,  32,  68, 104,  22,  58,  94},
+  {  12,  48,  84,   2,  38,  74,  28,  64, 100},
+  {  18,  54,  90,   8,  44,  80,  34,  70, 106},
+  {  24,  60,  96,  14,  50,  86,   4,  40,  76},  
+  {  30,  66, 102,  20,  56,  92,  10,  46,  82},
+	
+  {   1,  37,  73,  27,  63,  99,  17,  53,  89}, /* 2nd channel */
+  {   7,  43,  79,  33,  69, 105,  23,  59,  95},
+  {  13,  49,  85,   3,  39,  75,  29,  65, 101},
+  {  19,  55,  91,   9,  45,  81,  35,  71, 107},
+  {  25,  61,  97,  15,  51,  87,   5,  41,  77},  
+  {  31,  67, 103,  21,  57,  93,  11,  47,  83},
+};
+
+static const int dv_audio_frequency[3] = {
+    48000, 44100, 32000, 
+};
+
+static const int dv_audio_min_samples[2][3] = {
+    { 1580, 1452, 1053 }, /* 60 fields */
+    { 1896, 1742, 1264 }, /* 50 fileds */
+};