add pcm_s16le_planar support for electronicarts files

Originally committed as revision 11092 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/Makefile

@@ -245,6 +245,7 @@ OBJS-$(CONFIG_PCM_S24DAUD_DECODER)     += pcm.o
 OBJS-$(CONFIG_PCM_S24DAUD_ENCODER)     += pcm.o
 OBJS-$(CONFIG_PCM_S16LE_DECODER)       += pcm.o
 OBJS-$(CONFIG_PCM_S16LE_ENCODER)       += pcm.o
+OBJS-$(CONFIG_PCM_S16LE_PLANAR_DECODER)+= pcm.o
 OBJS-$(CONFIG_PCM_S16BE_DECODER)       += pcm.o
 OBJS-$(CONFIG_PCM_S16BE_ENCODER)       += pcm.o
 OBJS-$(CONFIG_PCM_U16LE_DECODER)       += pcm.o

libavcodec/allcodecs.c

@@ -211,6 +211,7 @@ void avcodec_register_all(void)
     REGISTER_ENCDEC  (PCM_S8, pcm_s8);
     REGISTER_ENCDEC  (PCM_S16BE, pcm_s16be);
     REGISTER_ENCDEC  (PCM_S16LE, pcm_s16le);
+    REGISTER_DECODER (PCM_S16LE_PLANAR, pcm_s16le_planar);
     REGISTER_ENCDEC  (PCM_S24BE, pcm_s24be);
     REGISTER_ENCDEC  (PCM_S24DAUD, pcm_s24daud);
     REGISTER_ENCDEC  (PCM_S24LE, pcm_s24le);

libavcodec/avcodec.h

@@ -189,6 +189,7 @@ enum CodecID {
     CODEC_ID_PCM_U24BE,
     CODEC_ID_PCM_S24DAUD,
     CODEC_ID_PCM_ZORK,
+    CODEC_ID_PCM_S16LE_PLANAR,
 
     /* various ADPCM codecs */
     CODEC_ID_ADPCM_IMA_QT= 0x11000,

libavcodec/pcm.c

@@ -28,6 +28,8 @@
 #include "bitstream.h" // for ff_reverse
 #include "bytestream.h"
 
+#define MAX_CHANNELS 64
+
 /* from g711.c by SUN microsystems (unrestricted use) */
 
 #define         SIGN_BIT        (0x80)      /* Sign bit for a A-law byte. */
@@ -374,15 +376,15 @@ static int pcm_decode_frame(AVCodecContext *avctx,
                             uint8_t *buf, int buf_size)
 {
     PCMDecode *s = avctx->priv_data;
-    int n;
+    int c, n;
     short *samples;
-    uint8_t *src;
+    uint8_t *src, *src2[MAX_CHANNELS];
 
     samples = data;
     src = buf;
 
     n= av_get_bits_per_sample(avctx->codec_id)/8;
-    if(n && buf_size % n){
+    if((n && buf_size % n) || avctx->channels > MAX_CHANNELS){
         av_log(avctx, AV_LOG_ERROR, "invalid PCM packet\n");
         return -1;
     }
@@ -390,6 +392,10 @@ static int pcm_decode_frame(AVCodecContext *avctx,
     buf_size= FFMIN(buf_size, *data_size/2);
     *data_size=0;
 
+    n = buf_size/avctx->channels;
+    for(c=0;c<avctx->channels;c++)
+        src2[c] = &src[c*n];
+
     switch(avctx->codec->id) {
     case CODEC_ID_PCM_S32LE:
         decode_to16(4, 1, 0, &src, &samples, buf_size);
@@ -430,6 +436,12 @@ static int pcm_decode_frame(AVCodecContext *avctx,
             *samples++ = bytestream_get_le16(&src);
         }
         break;
+    case CODEC_ID_PCM_S16LE_PLANAR:
+        for(n>>=1;n>0;n--)
+            for(c=0;c<avctx->channels;c++)
+                *samples++ = bytestream_get_le16(&src2[c]);
+        src = src2[avctx->channels-1];
+        break;
     case CODEC_ID_PCM_S16BE:
         n = buf_size >> 1;
         for(;n>0;n--) {
@@ -528,6 +540,7 @@ PCM_CODEC(CODEC_ID_PCM_U24LE, pcm_u24le);
 PCM_CODEC(CODEC_ID_PCM_U24BE, pcm_u24be);
 PCM_CODEC(CODEC_ID_PCM_S24DAUD, pcm_s24daud);
 PCM_CODEC(CODEC_ID_PCM_S16LE, pcm_s16le);
+PCM_DECODER(CODEC_ID_PCM_S16LE_PLANAR, pcm_s16le_planar);
 PCM_CODEC(CODEC_ID_PCM_S16BE, pcm_s16be);
 PCM_CODEC(CODEC_ID_PCM_U16LE, pcm_u16le);
 PCM_CODEC(CODEC_ID_PCM_U16BE, pcm_u16be);

libavcodec/utils.c

@@ -1189,6 +1189,7 @@ void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode)
             break;
         case CODEC_ID_PCM_S16LE:
         case CODEC_ID_PCM_S16BE:
+        case CODEC_ID_PCM_S16LE_PLANAR:
         case CODEC_ID_PCM_U16LE:
         case CODEC_ID_PCM_U16BE:
             bitrate = enc->sample_rate * enc->channels * 16;
@@ -1314,6 +1315,7 @@ int av_get_bits_per_sample(enum CodecID codec_id){
         return 8;
     case CODEC_ID_PCM_S16BE:
     case CODEC_ID_PCM_S16LE:
+    case CODEC_ID_PCM_S16LE_PLANAR:
     case CODEC_ID_PCM_U16BE:
     case CODEC_ID_PCM_U16LE:
         return 16;

libavformat/electronicarts.c

@@ -91,7 +91,7 @@ static int process_audio_header_elements(AVFormatContext *s)
     int inHeader = 1;
     EaDemuxContext *ea = s->priv_data;
     ByteIOContext *pb = s->pb;
-    int compression_type = -1, revision = -1;
+    int compression_type = -1, revision = -1, revision2 = -1;
 
     ea->bytes = 2;
     ea->sample_rate = -1;
@@ -136,6 +136,10 @@ static int process_audio_header_elements(AVFormatContext *s)
                     av_log (s, AV_LOG_INFO, "exited audio subheader\n");
                     inSubheader = 0;
                     break;
+                case 0xA0:
+                    revision2 = read_arbitary(pb);
+                    av_log (s, AV_LOG_INFO, "revision2 (element 0xA0) set to 0x%08x\n", revision2);
+                    break;
                 case 0xFF:
                     av_log (s, AV_LOG_INFO, "end of header block reached (within audio subheader)\n");
                     inSubheader = 0;
@@ -165,10 +169,17 @@ static int process_audio_header_elements(AVFormatContext *s)
         case  1: ea->audio_codec = CODEC_ID_ADPCM_EA_R1; break;
         case  2: ea->audio_codec = CODEC_ID_ADPCM_EA_R2; break;
         case  3: ea->audio_codec = CODEC_ID_ADPCM_EA_R3; break;
+        case -1: break;
         default:
             av_log(s, AV_LOG_ERROR, "unsupported stream type; revision=%i\n", revision);
             return 0;
         }
+        switch (revision2) {
+        case  8: ea->audio_codec = CODEC_ID_PCM_S16LE_PLANAR; break;
+        default:
+            av_log(s, AV_LOG_ERROR, "unsupported stream type; revision2=%i\n", revision2);
+            return 0;
+        }
         break;
     default:
         av_log(s, AV_LOG_ERROR, "unsupported stream type; compression_type=%i\n", compression_type);
@@ -392,6 +403,9 @@ static int ea_read_packet(AVFormatContext *s,
             if (!ea->audio_codec) {
                 url_fskip(pb, chunk_size);
                 break;
+            } else if (ea->audio_codec == CODEC_ID_PCM_S16LE_PLANAR) {
+                url_fskip(pb, 12);  /* planar header */
+                chunk_size -= 12;
             }
             ret = av_get_packet(pb, pkt, chunk_size);
             if (ret != chunk_size)