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- Log -----------------------------------------------------------------
commit 0df8ecfa7ac2ac271f0ceb67bc1f5d1ddafcf3af
Author: Marcelo Povoa <marspeoples...@gmail.com>
Date: Sat Aug 14 21:05:36 2010 -0300
Modify past pitch_lag decoding to be compliant with
the reference code rather than the spec
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 30fc262..d289b70 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -402,7 +402,7 @@ static void decode_pitch_lag_high(int *lag_int, int
*lag_frac, int pitch_index,
*lag_frac = 0;
}
/* minimum lag for next subframe */
- *base_lag_int = av_clip(*lag_int - 8, AMRWB_P_DELAY_MIN,
+ *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0 ? 1 : 0),
AMRWB_P_DELAY_MIN,
AMRWB_P_DELAY_MAX - 15);
/* XXX: the spec states clearly that *base_lag_int should be
* the nearest integer to *lag_int (minus 8), but the ref code
@@ -432,7 +432,8 @@ static void decode_pitch_lag_low(int *lag_int, int
*lag_frac, int pitch_index,
*lag_int = pitch_index - 24;
*lag_frac = 0;
}
- *base_lag_int = av_clip(*lag_int - 8, AMRWB_P_DELAY_MIN,
+ /* XXX: same problem as before */
+ *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0 ? 1 : 0),
AMRWB_P_DELAY_MIN,
AMRWB_P_DELAY_MAX - 15);
} else {
*lag_int = (pitch_index + 1) >> 1;
commit eaae7fce7962d7f77a6b40cc6e79654d8a751be2
Author: Marcelo Povoa <marspeoples...@gmail.com>
Date: Sat Aug 14 21:02:48 2010 -0300
Scaling and size bugfixes for extrapolate_isf (6K60 only)
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index a0e7457..30fc262 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -1129,7 +1129,7 @@ static void extrapolate_isf(float out[LP_ORDER_16k],
float isf[LP_ORDER])
float est, scale;
int i, i_max_corr;
- memcpy(out, isf, LP_ORDER - 1);
+ memcpy(out, isf, (LP_ORDER - 1) * sizeof(float));
out[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
/* Calculate the difference vector */
@@ -1154,8 +1154,8 @@ static void extrapolate_isf(float out[LP_ORDER_16k],
float isf[LP_ORDER])
- isf[i - 2 - i_max_corr];
/* Calculate an estimate for ISF(18) and scale ISF based on the error */
- est = 79.65 + (out[2] - out[3] - out[4]) / 6.0;
- scale = (FFMIN(est, 76.0) - out[LP_ORDER - 2]) /
+ est = 7965 + (out[2] - out[3] - out[4]) / 6.0;
+ scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) /
(out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]);
// XXX: should divide numerator by 2.0?
@@ -1172,12 +1172,11 @@ static void extrapolate_isf(float out[LP_ORDER_16k],
float isf[LP_ORDER])
}
for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
- out[i] = out[i - 1] + diff_hi[i];
+ out[i] = out[i - 1] + diff_hi[i] / (float) (1 << 15);
- /* XXX: Don't know why this 26214 coefficient, maybe it is not Q8 */
/* Scale the ISF vector for 16000 Hz */
for (i = 0; i < LP_ORDER_16k - 1; i++)
- out[i] *= 26214 / (float) (1 << 8);
+ out[i] *= 0.8;
}
/**
commit 276a401628b57729108a3d1b933615f14bae12b9
Author: Marcelo Povoa <marspeoples...@gmail.com>
Date: Sat Aug 14 21:00:03 2010 -0300
Create a slightly modified version of ff_lsp2polyf
to use with 16kHz ISPs at the higher band part
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 1cfa13e..a0e7457 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -313,6 +313,26 @@ static void interpolate_isp(double isp_q[4][LP_ORDER],
const double *isp4_past)
}
/**
+ * 16kHz version of ff_lsp2polyf
+ */
+static void lsp2polyf_16k(const double *lsp, double *f, int lp_half_order)
+{
+ int i, j;
+
+ f[0] = 0.25;
+ f[1] = -0.5 * lsp[0];
+ lsp -= 2;
+ for(i = 2; i <= lp_half_order; i++)
+ {
+ double val = -2 * lsp[2 * i];
+ f[i] = val * f[i - 1] + 2 * f[i - 2];
+ for(j = i - 1; j > 1; j--)
+ f[j] += f[j - 1] * val + f[j - 2];
+ f[1] += 0.25 * val;
+ }
+}
+
+/**
* Convert a ISP vector to LP coefficient domain {a_k}
* Equations from TS 26.190 section 5.2.4
*
@@ -321,20 +341,23 @@ static void interpolate_isp(double isp_q[4][LP_ORDER],
const double *isp4_past)
* @param[in] lp_half_order Half the number of LPs to construct
*/
static void isp2lp(const double *isp, float *lp, int lp_half_order) {
- double pa[MAX_LP_HALF_ORDER + 1], qa[MAX_LP_HALF_ORDER + 1];
+ double pa[10 + 1], qa[10 + 1];
float *lp2 = lp + (lp_half_order << 1);
double last_isp = isp[2 * lp_half_order - 1];
double qa_old = 0.0;
int i;
- ff_lsp2polyf(isp, pa, lp_half_order);
- ff_lsp2polyf(isp + 1, qa, lp_half_order - 1);
-
if (lp_half_order > 8) { // high-band specific
+ lsp2polyf_16k(isp, pa, lp_half_order);
+ lsp2polyf_16k(isp + 1, qa, lp_half_order - 1);
+
for (i = 0; i <= lp_half_order; i++)
pa[i] *= 4.0;
for (i = 0; i < lp_half_order; i++)
qa[i] *= 4.0;
+ } else {
+ ff_lsp2polyf(isp, pa, lp_half_order);
+ ff_lsp2polyf(isp + 1, qa, lp_half_order - 1);
}
for (i = 1; i < lp_half_order; i++) {
commit cea139e25cd421a3de4370670ca1b6d8b9e5542f
Author: Marcelo Povoa <marspeoples...@gmail.com>
Date: Sat Aug 14 20:57:13 2010 -0300
Modify isf2isp to work with every ISP length
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 12bf531..1cfa13e 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -175,15 +175,16 @@ static enum Mode unpack_bitstream(AMRWBContext *ctx,
const uint8_t *buf,
*
* @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)
+static void isf2isp(const float *isf, double *isp, int size)
{
int i;
- for (i = 0; i < LP_ORDER - 1; i++)
+ for (i = 0; i < size - 1; i++)
isp[i] = cos(2.0 * M_PI * isf[i]);
- isp[LP_ORDER - 1] = cos(4.0 * M_PI * isf[LP_ORDER - 1]);
+ isp[size - 1] = cos(4.0 * M_PI * isf[size - 1]);
}
/**
@@ -1201,7 +1202,7 @@ static void hb_synthesis(AMRWBContext *ctx, int subframe,
float *samples,
1.0 - isfp_inter[subframe], LP_ORDER);
extrapolate_isf(e_isf, e_isf);
- isf2isp(e_isf, e_isp);
+ isf2isp(e_isf, e_isp, LP_ORDER_16k);
isp2lp(e_isp, hb_lpc, LP_ORDER_16k / 2);
lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k + 1);
@@ -1309,7 +1310,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void
*data, int *data_size,
stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
- isf2isp(ctx->isf_cur, ctx->isp[3]);
+ isf2isp(ctx->isf_cur, ctx->isp[3], LP_ORDER);
/* Generate a ISP vector for each subframe */
if (ctx->first_frame) {
ctx->first_frame = 0;
commit 74da10c018a038577644160b873b0a2720e46656
Author: Marcelo Povoa <marspeoples...@gmail.com>
Date: Sat Aug 14 20:53:37 2010 -0300
Fix a major bug at the oversampled data memory update
which was compromising output quality
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 228617d..12bf531 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -1373,8 +1373,6 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void
*data, int *data_size,
synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
- /* XXX: Tested against the ref code until here, it "succeeds" at least
- * for cases in which the "opencore bug" don't interfere */
/* Synthesis speech post-processing */
de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
@@ -1383,17 +1381,14 @@ static int amrwb_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
high_pass_filter(&ctx->samples_up[UPS_MEM_SIZE], hpf_31_coef,
ctx->hpf_31_mem, &ctx->samples_up[UPS_MEM_SIZE]);
- /*for (i = 0; i < AMRWB_SUBFRAME_SIZE; i++)
- ctx->samples_up[UPS_MEM_SIZE+i] =
rint(ctx->samples_up[UPS_MEM_SIZE+i]);*/
-
upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
AMRWB_SFR_SIZE_OUT);
- /* High frequency band generation */
- high_pass_filter(&ctx->samples_up[UPS_MEM_SIZE], hpf_400_coef,
- ctx->hpf_400_mem, &ctx->samples_up[UPS_MEM_SIZE]);
+ /* High frequency band generation part */
+ high_pass_filter(hb_samples, hpf_400_coef, ctx->hpf_400_mem,
+ &ctx->samples_up[UPS_MEM_SIZE]);
- hb_gain = find_hb_gain(ctx, &ctx->samples_up[UPS_MEM_SIZE],
+ hb_gain = find_hb_gain(ctx, hb_samples,
cur_subframe->hb_gain, cf->vad);
scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
@@ -1412,7 +1407,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void
*data, int *data_size,
/* Add low frequency and high frequency bands */
for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++) {
// XXX: the lower band should really be upscaled by 2.0?
- sub_buf[i] = (sub_buf[i] * 2.0 + hb_samples[i]) / 32768.0;
+ sub_buf[i] = (sub_buf[i] * 1.0 + hb_samples[i]) / 32768.0;
}
/* Update buffers and history */
-----------------------------------------------------------------------
Summary of changes:
libavcodec/amrwbdec.c | 73 +++++++++++++++++++++++++++++++------------------
1 files changed, 46 insertions(+), 27 deletions(-)
hooks/post-receive
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AMR-WB decoder
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