int filter_bits; ///< filter precision for the current channel data
int32_t filter[64];
- int bias[2]; ///< a constant value added to channel data after filtering
+ unsigned bias[2]; ///< a constant value added to channel data after filtering
int num_blocks; ///< number of blocks inside the frame
int sample_offset;
val -= range;
}
if (bits)
- val = (val << bits) | get_bits(gb, bits);
+ val = ((unsigned)val << bits) | get_bits(gb, bits);
return val;
}
int *dst = ctx->channel_data[ch];
ctx->filter_params = get_vlc2(gb, set->filter_params.table, 9, 2);
- ctx->filter_bits = (ctx->filter_params - 2) >> 6;
- ctx->filter_length = ctx->filter_params - (ctx->filter_bits << 6) - 1;
+ if (ctx->filter_params > 1) {
+ ctx->filter_bits = (ctx->filter_params - 2) >> 6;
+ ctx->filter_length = ctx->filter_params - (ctx->filter_bits << 6) - 1;
+ }
if (ctx->filter_params == FILTER_RAW) {
for (i = 0; i < length; i++)
t = get_vlc2(gb, vlc[cmode].table, vlc[cmode].bits, 2);
t = extend_code(gb, t, 21, add_bits);
if (!cmode)
- coeff -= 12 << add_bits;
- coeff = t - coeff;
+ coeff -= 12U << add_bits;
+ coeff = (unsigned)t - coeff;
ctx->filter[i] = coeff;
cmode = coeff >> add_bits;
t = get_vlc2(gb, code_vlc->table, code_vlc->bits, 2);
code1 = t / range2;
code2 = t % range2;
- dst[i] = extend_code(gb, code1, range, 0) << add_bits;
- dst[i + 1] = extend_code(gb, code2, range, 0) << add_bits;
+ dst[i] = extend_code(gb, code1, range, 0) * (1U << add_bits);
+ dst[i + 1] = extend_code(gb, code2, range, 0) * (1U << add_bits);
if (add_bits) {
dst[i] |= get_bits(gb, add_bits);
dst[i + 1] |= get_bits(gb, add_bits);
acc = (acc + bias - 1) >> ctx->filter_bits;
acc = FFMAX(acc, min_clip);
} else {
- acc = (acc + bias) >> ctx->filter_bits;
+ acc = ((unsigned)acc + bias) >> ctx->filter_bits;
acc = FFMIN(acc, max_clip);
}
audio[i] += acc;
int len, ch, ret;
int dmode, mode[2], bits[2];
int *ch0, *ch1;
- int i, t, t2;
+ int i;
+ unsigned int t, t2;
len = 12 - get_unary(gb, 0, 6);
case 4:
for (i = 0; i < len; i++) {
t = ch1[i] + ctx->bias[1];
- t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);
- dst0[i] = (t2 + t) / 2;
- dst1[i] = (t2 - t) / 2;
+ t2 = ((ch0[i] + ctx->bias[0]) * 2) | (t & 1);
+ dst0[i] = (int)(t2 + t) / 2;
+ dst1[i] = (int)(t2 - t) / 2;
}
break;
}
init_get_bits(&gb, src + 2, table_size);
ctx->num_blocks = 0;
while (get_bits_left(&gb) > 0) {
+ if (ctx->num_blocks >= FF_ARRAY_ELEMS(ctx->block_size))
+ return AVERROR_INVALIDDATA;
ctx->block_size[ctx->num_blocks] = get_bits(&gb, 13 + avctx->channels);
if (get_bits1(&gb)) {
ctx->block_pts[ctx->num_blocks] = get_bits(&gb, 9);
}
-AVCodec ff_ralf_decoder = {
+const AVCodec ff_ralf_decoder = {
.name = "ralf",
.long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),
.type = AVMEDIA_TYPE_AUDIO,
.close = decode_close,
.decode = decode_frame,
.flush = decode_flush,
- .capabilities = AV_CODEC_CAP_DR1,
+ .capabilities = AV_CODEC_CAP_CHANNEL_CONF |
+ AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};