Cook compatibe decoder, patch by Benjamin Larsson

Add cook demucing, change rm demuxer so that it reorders audio packets
before sending them to the decoder, and send minimum decodeable sized
packets; pass only real codec extradata fo the decoder
Fix 28_8 decoder for the new demuxer strategy

Originally committed as revision 4726 to svn://svn.ffmpeg.org/ffmpeg/trunk

libavcodec/Makefile

@@ -32,6 +32,9 @@ endif
 ifeq ($(CONFIG_CINEPAK_DECODER),yes)
     OBJS+= cinepak.o
 endif
+ifeq ($(CONFIG_COOK_DECODER),yes)
+    OBJS+= cook.o
+endif
 ifneq ($(CONFIG_CLJR_DECODER)$(CONFIG_CLJR_ENCODER),)
     OBJS+= cljr.o
 endif

libavcodec/allcodecs.c

@@ -491,6 +491,9 @@ void avcodec_register_all(void)
 #ifdef CONFIG_QDM2_DECODER
     register_avcodec(&qdm2_decoder);
 #endif //CONFIG_QDM2_DECODER
+#ifdef CONFIG_COOK_DECODER
+    register_avcodec(&cook_decoder);
+#endif //CONFIG_COOK_DECODER
 #ifdef CONFIG_RAWVIDEO_DECODER
     register_avcodec(&rawvideo_decoder);
 #endif //CONFIG_RAWVIDEO_DECODER

libavcodec/avcodec.h

@@ -21,8 +21,8 @@ extern "C" {
 #define AV_STRINGIFY(s)	AV_TOSTRING(s)
 #define AV_TOSTRING(s) #s
 
-#define LIBAVCODEC_VERSION_INT ((50<<16)+(1<<8)+0)
-#define LIBAVCODEC_VERSION     50.1.0
+#define LIBAVCODEC_VERSION_INT ((51<<16)+(0<<8)+0)
+#define LIBAVCODEC_VERSION     51.0.0
 #define LIBAVCODEC_BUILD       LIBAVCODEC_VERSION_INT
 
 #define LIBAVCODEC_IDENT       "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
@@ -187,6 +187,7 @@ enum CodecID {
     CODEC_ID_WESTWOOD_SND1,
     CODEC_ID_GSM,    
     CODEC_ID_QDM2,
+    CODEC_ID_COOK,
     
     CODEC_ID_OGGTHEORA= 0x16000, 
 
@@ -2004,6 +2005,7 @@ extern AVCodec mp3_decoder;
 extern AVCodec mp3adu_decoder;
 extern AVCodec mp3on4_decoder;
 extern AVCodec qdm2_decoder;
+extern AVCodec cook_decoder;
 extern AVCodec mace3_decoder;
 extern AVCodec mace6_decoder;
 extern AVCodec huffyuv_decoder;

libavcodec/ra288.c

@@ -228,41 +228,19 @@ static int ra288_decode_frame(AVCodecContext * avctx,
             void *data, int *data_size,
             uint8_t * buf, int buf_size)
 {
-  if(avctx->extradata_size>=6)
-  {
-//((short*)(avctx->extradata))[0]; /* subpacket size */
-//((short*)(avctx->extradata))[1]; /* subpacket height */
-//((short*)(avctx->extradata))[2]; /* subpacket flavour */
-//((short*)(avctx->extradata))[3]; /* coded frame size */
-//((short*)(avctx->extradata))[4]; /* codec's data length  */
-//((short*)(avctx->extradata))[5...] /* codec's data */
-    int bret;
     void *datao;
-    int w=avctx->block_align; /* 228 */
-    int h=((short*)(avctx->extradata))[1]; /* 12 */
-    int cfs=((short*)(avctx->extradata))[3]; /* coded frame size 38 */
-    int i,j;
-    if(buf_size<w*h)
+
+    if (buf_size < avctx->block_align)
     {
-	av_log(avctx, AV_LOG_ERROR, "ffra288: Error! Input buffer is too small [%d<%d]\n",buf_size,w*h);
+	av_log(avctx, AV_LOG_ERROR, "ffra288: Error! Input buffer is too small [%d<%d]\n",buf_size,avctx->block_align);
 	return 0;
     }
+
     datao = data;
-    bret = 0;
-    for (j = 0; j < h/2; j++)
-	for (i = 0; i < h; i++)
-    {
-	    data=decode_block(avctx,&buf[j*cfs+cfs*i*h/2],(signed short *)data,cfs);
-	    bret += cfs;
-    }
+    data = decode_block(avctx, buf, (signed short *)data, avctx->block_align);
+
     *data_size = (char *)data - (char *)datao;
-    return bret;
-  }
-  else
-  {
-    av_log(avctx, AV_LOG_ERROR, "ffra288: Error: need extra data!!!\n");
-    return 0;
-  }
+    return avctx->block_align;
 }
 
 AVCodec ra_288_decoder =

libavformat/avformat.h

@@ -5,8 +5,8 @@
 extern "C" {
 #endif
 
-#define LIBAVFORMAT_VERSION_INT ((49<<16)+(2<<8)+0)
-#define LIBAVFORMAT_VERSION     49.2.0
+#define LIBAVFORMAT_VERSION_INT ((50<<16)+(0<<8)+0)
+#define LIBAVFORMAT_VERSION     50.0.0
 #define LIBAVFORMAT_BUILD       LIBAVFORMAT_VERSION_INT
 
 #define LIBAVFORMAT_IDENT       "Lavf" AV_STRINGIFY(LIBAVFORMAT_VERSION)

libavformat/rm.c

@@ -42,6 +42,14 @@ typedef struct {
     int old_format;
     int current_stream;
     int remaining_len;
+    /// Audio descrambling matrix parameters
+    uint8_t *audiobuf; ///< place to store reordered audio data
+    int64_t audiotimestamp; ///< Audio packet timestamp
+    int sub_packet_cnt; // Subpacket counter, used while reading
+    int sub_packet_size, sub_packet_h, coded_framesize; ///< Descrambling parameters from container
+    int audio_stream_num; ///< Stream number for audio packets
+    int audio_pkt_cnt; ///< Output packet counter
+    int audio_framesize; /// Audio frame size from container
 } RMContext;
 
 #ifdef CONFIG_MUXERS
@@ -478,6 +486,7 @@ static void get_str8(ByteIOContext *pb, char *buf, int buf_size)
 static void rm_read_audio_stream_info(AVFormatContext *s, AVStream *st, 
                                       int read_all)
 {
+    RMContext *rm = s->priv_data;
     ByteIOContext *pb = &s->pb;
     char buf[128];
     uint32_t version;
@@ -500,39 +509,60 @@ static void rm_read_audio_stream_info(AVFormatContext *s, AVStream *st,
         st->codec->codec_type = CODEC_TYPE_AUDIO;
         st->codec->codec_id = CODEC_ID_RA_144;
     } else {
-        int flavor, sub_packet_h, coded_framesize;
+        int flavor, sub_packet_h, coded_framesize, sub_packet_size;
         /* old version (4) */
         get_be32(pb); /* .ra4 */
         get_be32(pb); /* data size */
         get_be16(pb); /* version2 */
         get_be32(pb); /* header size */
         flavor= get_be16(pb); /* add codec info / flavor */
-        coded_framesize= get_be32(pb); /* coded frame size */
+        rm->coded_framesize = coded_framesize = get_be32(pb); /* coded frame size */
         get_be32(pb); /* ??? */
         get_be32(pb); /* ??? */
         get_be32(pb); /* ??? */
-        sub_packet_h= get_be16(pb); /* 1 */ 
+        rm->sub_packet_h = sub_packet_h = get_be16(pb); /* 1 */ 
         st->codec->block_align= get_be16(pb); /* frame size */
-        get_be16(pb); /* sub packet size */
+        rm->sub_packet_size = sub_packet_size = get_be16(pb); /* sub packet size */
         get_be16(pb); /* ??? */
+        if (((version >> 16) & 0xff) == 5) {
+            get_be16(pb); get_be16(pb); get_be16(pb); }
         st->codec->sample_rate = get_be16(pb);
         get_be32(pb);
         st->codec->channels = get_be16(pb);
+        if (((version >> 16) & 0xff) == 5) {
+            get_be32(pb);
+	    buf[0] = get_byte(pb);
+	    buf[1] = get_byte(pb);
+	    buf[2] = get_byte(pb);
+	    buf[3] = get_byte(pb);
+	    buf[4] = 0;
+	} else {
         get_str8(pb, buf, sizeof(buf)); /* desc */
         get_str8(pb, buf, sizeof(buf)); /* desc */
+	}
         st->codec->codec_type = CODEC_TYPE_AUDIO;
         if (!strcmp(buf, "dnet")) {
             st->codec->codec_id = CODEC_ID_AC3;
         } else if (!strcmp(buf, "28_8")) {
             st->codec->codec_id = CODEC_ID_RA_288;
-            st->codec->extradata_size= 10;
+            st->codec->extradata_size= 0;
+            rm->audio_framesize = st->codec->block_align;
+            st->codec->block_align = coded_framesize;
+            rm->audiobuf = av_malloc(rm->audio_framesize * sub_packet_h);
+        } else if (!strcmp(buf, "cook")) {
+            int codecdata_length, i;
+            get_be16(pb); get_byte(pb);
+            if (((version >> 16) & 0xff) == 5)
+                get_byte(pb);
+            codecdata_length = get_be32(pb);
+            st->codec->codec_id = CODEC_ID_COOK;
+            st->codec->extradata_size= codecdata_length;
             st->codec->extradata= av_mallocz(st->codec->extradata_size);
-            /* this is completly braindead and broken, the idiot who added this codec and endianness
-               specific reordering to mplayer and libavcodec/ra288.c should be drowned in a see of cola */
-            //FIXME pass the unpermutated extradata
-            ((uint16_t*)st->codec->extradata)[1]= sub_packet_h;
-            ((uint16_t*)st->codec->extradata)[2]= flavor;
-            ((uint16_t*)st->codec->extradata)[3]= coded_framesize;
+            for(i = 0; i < codecdata_length; i++)
+                ((uint8_t*)st->codec->extradata)[i] = get_byte(pb);
+            rm->audio_framesize = st->codec->block_align;
+            st->codec->block_align = rm->sub_packet_size;
+            rm->audiobuf = av_malloc(rm->audio_framesize * sub_packet_h);
         } else {
             st->codec->codec_id = CODEC_ID_NONE;
             pstrcpy(st->codec->codec_name, sizeof(st->codec->codec_name),
@@ -819,6 +849,16 @@ static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
         }
         pkt->size = len;
         st = s->streams[0];
+    } else if (rm->audio_pkt_cnt) {
+        // If there are queued audio packet return them first
+        st = s->streams[rm->audio_stream_num];
+        av_new_packet(pkt, st->codec->block_align);
+        memcpy(pkt->data, rm->audiobuf + st->codec->block_align *
+               (rm->sub_packet_h * rm->audio_framesize / st->codec->block_align - rm->audio_pkt_cnt),
+               st->codec->block_align);
+        rm->audio_pkt_cnt--;
+        pkt->flags = 0;
+        pkt->stream_index = rm->audio_stream_num;
     } else {
         int seq=1;
 resync:
@@ -850,15 +890,57 @@ resync:
             if(len2 && len2<len)
                 len=len2;
             rm->remaining_len-= len;
+            av_get_packet(pb, pkt, len);
+        }
+
+        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+            if ((st->codec->codec_id == CODEC_ID_RA_288) ||
+                (st->codec->codec_id == CODEC_ID_COOK)) {
+                int x;
+                int sps = rm->sub_packet_size;
+                int cfs = rm->coded_framesize;
+                int h = rm->sub_packet_h;
+                int y = rm->sub_packet_cnt;
+                int w = rm->audio_framesize;
+
+                if (flags & 2)
+                    y = rm->sub_packet_cnt = 0;
+                if (!y)
+                    rm->audiotimestamp = timestamp;
+
+                switch(st->codec->codec_id) {
+                    case CODEC_ID_RA_288:
+                        for (x = 0; x < h/2; x++)
+                            get_buffer(pb, rm->audiobuf+x*2*w+y*cfs, cfs);
+                        break;
+                    case CODEC_ID_COOK:
+                        for (x = 0; x < w/sps; x++)
+                            get_buffer(pb, rm->audiobuf+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
+                        break;
+                }
+
+                if (++(rm->sub_packet_cnt) < h)
+                    goto resync;
+                else {
+                    rm->sub_packet_cnt = 0;
+                    rm->audio_stream_num = i;
+                    rm->audio_pkt_cnt = h * w / st->codec->block_align - 1;
+                    // Release first audio packet
+                    av_new_packet(pkt, st->codec->block_align);
+                    memcpy(pkt->data, rm->audiobuf, st->codec->block_align);
+                    timestamp = rm->audiotimestamp;
+                    flags = 2; // Mark first packet as keyframe
+                }
+            } else
+                av_get_packet(pb, pkt, len);
         }
 
         if(  (st->discard >= AVDISCARD_NONKEY && !(flags&2))
            || st->discard >= AVDISCARD_ALL){
-            url_fskip(pb, len);
+            av_free_packet(pkt);
             goto resync;
         }
         
-        av_get_packet(pb, pkt, len);
         pkt->stream_index = i;
 
 #if 0
@@ -896,6 +978,9 @@ resync:
 
 static int rm_read_close(AVFormatContext *s)
 {
+    RMContext *rm = s->priv_data;
+
+    av_free(rm->audiobuf);
     return 0;
 }