On 08/30/2010 04:44 PM, Marcelo Galvão Póvoa wrote:
On 30 August 2010 09:31, Vitor Sessak<[email protected]> wrote:
On 08/27/2010 11:13 PM, Marcelo Galvão Póvoa wrote:
On 26 August 2010 18:55, Vitor Sessak<[email protected]> wrote:
Marcelo Galvão Póvoa wrote:
Fixed excitation array incorrect length.
Should now be applicable cleanly to the latest ffmpeg revision
(fd151a5f8bd152c456a).
First look:
[...]
+/**
+ * Convert an ISF vector into an ISP vector
+ *
+ * @param[in] isf Isf vector
+ * @param[out] isp Output isp vector
+ * @param[in] size Isf/isp size
+ */
+static void isf2isp(const float *isf, double *isp, int size)
+{
+ int i;
+
+ for (i = 0; i< size - 1; i++)
+ isp[i] = cos(2.0 * M_PI * isf[i]);
+
+ isp[size - 1] = cos(4.0 * M_PI * isf[size - 1]);
+}
Almost duplication of amrnbdec.c:lsf2lsp().
Yes, but this last element doubling diverges between AMR-NB and AMR-WB
reference sources.
Can't you just do "isp[size - 1] *= 2" or it is used elsewhere?
It's not the last element output that is doubled but the input. Since
it is a cosine, I would have to calculate it again for the last
element I guess.
I mean doubling the last element of the input before calling the
function, if it doesn't make the rest of the code more complex.
[...]
+ for(j = i - 1; j> 1; j--)
+ f[j] += f[j - 1] * val + f[j - 2];
+ f[1] += 0.25 * val;
+ }
+}
Hmm, can't this function be replaced by ff_lsp2polyf() with some
rescaling?
Actually I realize now that the difference between these functions is
just the entire output scaled down by 0.25. That was silly because I
was scaling by 4.0 after this function. It is weird why the reference
decoder does this process. Maybe it should worth someone checking it
out to be sure about this.
I suppose you checked that after this change the output of your decoder
didn't change, no?
Actually I cannot compare the output bitwise because the high band
involves a random excitation, I am afraid that it is not the same for
each run (or it is?).
Yes, you always send 1 as seed, so the prng output is always the same
(and this is a good thing), but since you use floats output might not be
bitexact. I suggest you do:
./tests/tiny_psnr output_ref.wav output_patched.wav 2
And see if stddev is ~ 0.1
[...]
+/**
+ * Parses a speech frame, storing data in the Context
+ *
+ * @param[in,out] ctx The context
+ * @param[in] buf Pointer to the input buffer
+ * @param[in] buf_size Size of the input buffer
+ *
+ * @return The frame mode
+ */
+static enum Mode unpack_bitstream(AMRWBContext *ctx, const uint8_t *buf,
+ int buf_size)
+{
+ GetBitContext gb;
+ uint16_t *data;
+
+ init_get_bits(&gb, buf, buf_size * 8);
+
+ /* decode frame header (1st octet) */
+ skip_bits(&gb, 1); // padding bit
+ ctx->fr_cur_mode = get_bits(&gb, 4);
+ ctx->fr_quality = get_bits1(&gb);
+ skip_bits(&gb, 2); // padding bits
+
+ // XXX: We are using only the "MIME/storage" format
+ // used by libopencore. This format is simpler and
+ // does not carry the auxiliary information of the frame
+
+ data = (uint16_t *)&ctx->frame;
+ memset(data, 0, sizeof(AMRWBFrame));
+ buf++;
+
+ if (ctx->fr_cur_mode< MODE_SID) { /* Normal speech frame */
+ const uint16_t *perm =
amr_bit_orderings_by_mode[ctx->fr_cur_mode];
+ int field_size;
+
+ while ((field_size = *perm++)) {
+ int field = 0;
+ int field_offset = *perm++;
+ while (field_size--) {
+ uint16_t bit_idx = *perm++;
+ field<<= 1;
+ /* The bit index inside the byte is reversed (MSB->LSB) */
+ field |= BIT_POS(buf[bit_idx>> 3], 7 - (bit_idx& 7));
+ }
+ data[field_offset] = field;
+ }
+ }
+
+ return ctx->fr_cur_mode;
+}
Can't you reorder the bit indexes in a way it is in the LSB->MSB so you can
use just the usual bitstream functions?
Which bitstream functions?
Those in libavcodec/get_bits.h (get_bits(), etc).
If the tables were in LSB->MSB order, it
would be the same algorithm as in AMR-NB, the only difference is that
I do (7 - index) to reverse the order.
Can't you change the tables in a way you can reuse the unpack function
of AMRNB?
-Vitor
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