[FFmpeg-soc] [soc]AMR-WB decoder branch, master, updated.
Marcelo Póvoa
marspeoplester at gmail.com
Fri Jul 30 07:35:21 CEST 2010
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commit 121492a18969d4665210fe6dede43482776acce8
Author: Marcelo Povoa <marspeoplester at gmail.com>
Date: Fri Jul 30 02:34:33 2010 -0300
Write post-processing FIR filters for the high-band
diff --git a/libavcodec/amrwbdata.h b/libavcodec/amrwbdata.h
index 76a1a60..17f7d9e 100644
--- a/libavcodec/amrwbdata.h
+++ b/libavcodec/amrwbdata.h
@@ -28,6 +28,7 @@
#define LP_ORDER_16k 20 ///< lpc filter order at 16kHz
#define UPS_FIR_SIZE 12 ///< upsampling filter size
#define UPS_MEM_SIZE 2 * UPS_FIR_SIZE
+#define HB_FIR_SIZE 30 ///< amount of past data needed by HB filters
#define MIN_ISF_SPACING (128 / 32768.0) ///< minimum isf gap
#define PRED_FACTOR (1.0 / 3.0)
@@ -1840,6 +1841,36 @@ static const uint16_t qua_hb_gain[16] = {
11210, 12206, 13391, 14844, 16770, 19655, 24289, 32728
};
+/* High-band post-processing FIR filters coefficients in Q15 */
+// XXX: not sure if it is Q15 indeed (guessing)
+static const float bpf_6_7_coef[31] = { // band pass, 6kHz and 7kHz cutoffs
+ -9.765625e-04, 1.434326e-03, 9.765625e-04,
+ -8.239746e-04, -1.126099e-02, 3.424072e-02,
+ -4.336548e-02, 0.000000e+00, 1.159058e-01,
+ -2.709961e-01, 3.768616e-01, -3.352051e-01,
+ 1.082764e-01, 2.369995e-01, -5.493469e-01,
+ 6.749878e-01,
+ -5.493469e-01, 2.369995e-01, 1.082764e-01,
+ -3.352051e-01, 3.768616e-01, -2.709961e-01,
+ 1.159058e-01, 0.000000e+00, -4.336548e-02,
+ 3.424072e-02, -1.126099e-02, -8.239746e-04,
+ 9.765625e-04, 1.434326e-03, -9.765625e-04
+};
+
+static const float lpf_7_coef[31] = { // low pass, 7kHz cutoff
+ -6.408691e-04, 1.434326e-03, -2.716064e-03,
+ 4.455566e-03, -6.195068e-03, 6.988525e-03,
+ -5.401611e-03, 0.000000e+00, 1.022339e-02,
+ -2.560425e-02, 4.531860e-02, -6.747437e-02,
+ 8.944702e-02, -1.080933e-01, 1.206360e-01,
+ 8.753052e-01,
+ 1.206360e-01, -1.080933e-01, 8.944702e-02,
+ -6.747437e-02, 4.531860e-02, -2.560425e-02,
+ 1.022339e-02, 0.000000e+00, -5.401611e-03,
+ 6.988525e-03, -6.195068e-03, 4.455566e-03,
+ -2.716064e-03, 1.434326e-03, -6.408691e-04
+};
+
/* Core frame sizes in each mode */
static const uint16_t cf_sizes_wb[] = {
132, 177, 253, 285, 317, 365, 397, 461, 477,
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 2786816..de4b399 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -73,11 +73,13 @@ typedef struct {
float prev_tr_gain; ///< previous initial gain used by noise enhancer for thresold
float samples_az[LP_ORDER + AMRWB_SUBFRAME_SIZE]; ///< lower band samples from synthesis at 12.8kHz
- float samples_up[UPS_MEM_SIZE + AMRWB_SUBFRAME_SIZE]; ///< lower band samples processed for upsampling at 12.8kHz
+ float samples_up[UPS_MEM_SIZE + AMRWB_SUBFRAME_SIZE]; ///< lower band samples processed for upsampling
float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_OUT]; ///< higher band samples from synthesis at 16kHz
- float hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in the high-pass filters
+ float hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in the high pass filters
float demph_mem[1]; ///< previous value in the de-emphasis filter
+ float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
+ float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
AVLFG prng; ///< random number generator for white noise excitation
uint8_t first_frame; ///< flag active during decoding of the first frame
@@ -1176,6 +1178,39 @@ static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
}
/**
+ * Apply to high-band samples a 15th order filter
+ * The filter characteristic depends on the given coefficients
+ *
+ * @param out [out] buffer for filtered output
+ * @param fir_coef [in] filter coefficients
+ * @param mem [in/out] state from last filtering (updated)
+ * @param cp_gain [in] compensation gain (usually the filter gain)
+ * @param in [in] input speech data (high-band)
+ *
+ * @remark It is safe to pass the same array in in and out parameters.
+ */
+static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
+ float mem[HB_FIR_SIZE], float cp_gain, const float *in)
+{
+ int i, j;
+ float data[AMRWB_SFR_SIZE_OUT + HB_FIR_SIZE]; // past and current samples
+
+ memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
+
+ for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++)
+ data[i + HB_FIR_SIZE] = in[i] / cp_gain;
+
+ for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++)
+ {
+ out[i] = 0.0;
+ for (j = 0; j <= HB_FIR_SIZE; j++)
+ out[i] += data[i + j] * fir_coef[j];
+ }
+
+ memcpy(mem, data + AMRWB_SFR_SIZE_OUT, HB_FIR_SIZE * sizeof(float));
+}
+
+/**
* Update context state before the next subframe
*/
static void update_sub_state(AMRWBContext *ctx)
@@ -1212,6 +1247,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
float voice_fac, stab_fac; // parameters used for gain smoothing
float synth_exc[AMRWB_SUBFRAME_SIZE]; // post-processed excitation for synthesis
float hb_exc[AMRWB_SFR_SIZE_OUT]; // excitation for the high frequency band
+ float hb_samples[AMRWB_SFR_SIZE_OUT]; // filtered high-band samples from synthesis
float hb_gain;
int sub, i;
@@ -1324,12 +1360,17 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
hb_exc, ctx->isf_cur, ctx->isf_past_final);
/* High-band post-processing filters */
+ hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
+ 4.0, &ctx->samples_hb[LP_ORDER_16k]);
+
+ if (ctx->fr_cur_mode == MODE_23k85)
+ hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
+ 1.0, hb_samples);
/* 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 +
- ctx->samples_hb[i + LP_ORDER_16k]) / 32768.0;
+ sub_buf[i] = (sub_buf[i] * 2.0 + hb_samples[i]) / 32768.0;
}
/* Update buffers and history */
commit 8cbd44b44a1b8d4332fc8d22ced69a93d85a31fc
Author: Marcelo Povoa <marspeoplester at gmail.com>
Date: Fri Jul 30 02:32:08 2010 -0300
Fix high-band excitation length
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 7f6152f..2786816 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -931,11 +931,10 @@ static void de_emphasis(float *out, float *in, float m, float mem[1])
/**
* Apply to synthesis a 2nd order high-pass filter
- * with cutoff frequency at 31 Hz
*
* @param out [out] buffer for filtered output
* @param hpf_coef [in] filter coefficients as used below
- * @param mem [in] state from last filtering
+ * @param mem [in/out] state from last filtering (updated)
* @param in [in] input speech data
*
* @remark It is safe to pass the same array in in and out parameters.
@@ -1028,13 +1027,13 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SUBFRAME_SIZE);
/* Generate a white-noise excitation */
- for (i = 0; i < AMRWB_SUBFRAME_SIZE; i++)
+ for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++)
hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc, energy,
- AMRWB_SUBFRAME_SIZE);
+ AMRWB_SFR_SIZE_OUT);
- for (i = 0; i < AMRWB_SUBFRAME_SIZE; i++)
+ for (i = 0; i < AMRWB_SFR_SIZE_OUT; i++)
hb_exc[i] *= hb_gain;
}
@@ -1212,7 +1211,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
float synth_fixed_gain; // the fixed gain that synthesis should use
float voice_fac, stab_fac; // parameters used for gain smoothing
float synth_exc[AMRWB_SUBFRAME_SIZE]; // post-processed excitation for synthesis
- float hb_exc[AMRWB_SUBFRAME_SIZE]; // excitation for the high frequency band
+ float hb_exc[AMRWB_SFR_SIZE_OUT]; // excitation for the high frequency band
float hb_gain;
int sub, i;
-----------------------------------------------------------------------
Summary of changes:
libavcodec/amrwbdata.h | 31 ++++++++++++++++++++++++
libavcodec/amrwbdec.c | 60 ++++++++++++++++++++++++++++++++++++++++--------
2 files changed, 81 insertions(+), 10 deletions(-)
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