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authorRonald S. Bultje <rsbultje@gmail.com>2009-03-20 01:22:51 +0000
committerRonald S. Bultje <rsbultje@gmail.com>2009-03-20 01:22:51 +0000
commitcf16c17a69be7f665c9d465e0ce29a1aa9d5783c (patch)
treee96da15cc0174a43af9836670ed3f6766395a97b /libavformat/rmdec.c
parente9fce261a65dabea2a8d67039c3ae093e0f288c8 (diff)
Add byte reordering for the SIPRO audio codec. See "rmdec.c: add SIPR codec
try #2" thread. Patch originally by Vladimir Voroshilov. Originally committed as revision 18062 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rmdec.c')
-rw-r--r--libavformat/rmdec.c69
1 files changed, 62 insertions, 7 deletions
diff --git a/libavformat/rmdec.c b/libavformat/rmdec.c
index a1c0906d46..5d8270159b 100644
--- a/libavformat/rmdec.c
+++ b/libavformat/rmdec.c
@@ -48,6 +48,21 @@ typedef struct {
int audio_pkt_cnt; ///< Output packet counter
} RMDemuxContext;
+static const unsigned char sipr_swaps[38][2] = {
+ { 0, 63 }, { 1, 22 }, { 2, 44 }, { 3, 90 },
+ { 5, 81 }, { 7, 31 }, { 8, 86 }, { 9, 58 },
+ { 10, 36 }, { 12, 68 }, { 13, 39 }, { 14, 73 },
+ { 15, 53 }, { 16, 69 }, { 17, 57 }, { 19, 88 },
+ { 20, 34 }, { 21, 71 }, { 24, 46 }, { 25, 94 },
+ { 26, 54 }, { 28, 75 }, { 29, 50 }, { 32, 70 },
+ { 33, 92 }, { 35, 74 }, { 38, 85 }, { 40, 56 },
+ { 42, 87 }, { 43, 65 }, { 45, 59 }, { 48, 79 },
+ { 49, 93 }, { 51, 89 }, { 55, 95 }, { 61, 76 },
+ { 67, 83 }, { 77, 80 }
+};
+
+static const unsigned char sipr_subpk_size[4] = { 29, 19, 37, 20 };
+
static inline void get_strl(ByteIOContext *pb, char *buf, int buf_size, int len)
{
int i;
@@ -171,19 +186,28 @@ static int rm_read_audio_stream_info(AVFormatContext *s, ByteIOContext *pb,
return -1;
}
- if(sub_packet_size <= 0){
- av_log(s, AV_LOG_ERROR, "sub_packet_size is invalid\n");
- return -1;
- }
-
if (!strcmp(buf, "cook")) st->codec->codec_id = CODEC_ID_COOK;
else if (!strcmp(buf, "sipr")) st->codec->codec_id = CODEC_ID_SIPR;
else st->codec->codec_id = CODEC_ID_ATRAC3;
+
+ ast->audio_framesize = st->codec->block_align;
+ if (st->codec->codec_id == CODEC_ID_SIPR) {
+ if (flavor > 3) {
+ av_log(s, AV_LOG_ERROR, "bad SIPR file flavor %d\n",
+ flavor);
+ return -1;
+ }
+ st->codec->block_align = sipr_subpk_size[flavor];
+ } else {
+ if(sub_packet_size <= 0){
+ av_log(s, AV_LOG_ERROR, "sub_packet_size is invalid\n");
+ return -1;
+ }
+ st->codec->block_align = ast->sub_packet_size;
+ }
st->codec->extradata_size= codecdata_length;
st->codec->extradata= av_mallocz(st->codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
get_buffer(pb, st->codec->extradata, st->codec->extradata_size);
- ast->audio_framesize = st->codec->block_align;
- st->codec->block_align = ast->sub_packet_size;
if(ast->audio_framesize >= UINT_MAX / sub_packet_h){
av_log(s, AV_LOG_ERROR, "rm->audio_framesize * sub_packet_h too large\n");
@@ -637,6 +661,31 @@ rm_ac3_swap_bytes (AVStream *st, AVPacket *pkt)
}
}
+/** perform 4-bit block reordering for SIPR data */
+static void
+rm_reorder_sipr_data (RMStream *ast)
+{
+ int n, bs = ast->sub_packet_h * ast->audio_framesize * 2 / 96; // nibbles per subpacket
+
+ for (n = 0; n < 38; n++) {
+ int j;
+ int i = bs * sipr_swaps[n][0];
+ int o = bs * sipr_swaps[n][1];
+ uint8_t *buf = ast->pkt.data;
+
+ /* swap 4bit-nibbles of block 'i' with 'o' */
+ for (j = 0; j < bs; j++, i++, o++) {
+ int x = (buf[i >> 1] >> (4 * (i & 1))) & 0xF,
+ y = (buf[o >> 1] >> (4 * (o & 1))) & 0xF;
+
+ buf[o >> 1] = (x << (4 * (o & 1))) |
+ (buf[o >> 1] & (0xF << (4 * !(o & 1))));
+ buf[i >> 1] = (y << (4 * (i & 1))) |
+ (buf[i >> 1] & (0xF << (4 * !(i & 1))));
+ }
+ }
+}
+
int
ff_rm_parse_packet (AVFormatContext *s, ByteIOContext *pb,
AVStream *st, RMStream *ast, int len, AVPacket *pkt,
@@ -675,10 +724,16 @@ ff_rm_parse_packet (AVFormatContext *s, ByteIOContext *pb,
for (x = 0; x < w/sps; x++)
get_buffer(pb, ast->pkt.data+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
break;
+ case CODEC_ID_SIPR:
+ get_buffer(pb, ast->pkt.data + y * w, w);
+ break;
}
if (++(ast->sub_packet_cnt) < h)
return -1;
+ if (st->codec->codec_id == CODEC_ID_SIPR)
+ rm_reorder_sipr_data(ast);
+
ast->sub_packet_cnt = 0;
rm->audio_stream_num = st->index;
rm->audio_pkt_cnt = h * w / st->codec->block_align;