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commit 4e9169a62e5fdb901d0f1a0a4a85613b575e7b2f
Author: Marcelo Povoa <marspeoplester(a)gmail.com>
Date: Mon Jul 26 22:46:33 2010 -0300
Implement output upsampling to 16kHz, this finishes
the lower band generation part, but not functional yet
diff --git a/libavcodec/amrwbdata.h b/libavcodec/amrwbdata.h
index d8a1068..46a97ff 100644
--- a/libavcodec/amrwbdata.h
+++ b/libavcodec/amrwbdata.h
@@ -26,10 +26,14 @@
#define LP_ORDER 16 ///< linear predictive coding filter order
#define LP_ORDER_16k 20 ///< lpc filter order at 16kHz
+#define UPS_FIR_SIZE 12 ///< upsampling filter size
+#define SAMPLE_MEM 24 ///< number of stored past samples given by
+ ///< max(LP_ORDER, UPS_FIR_SIZE * 2)
#define MIN_ISF_SPACING (128 / 32768.0) ///< minimum isf gap
#define PRED_FACTOR (1.0 / 3.0)
#define MIN_ENERGY -14.0 ///< initial innnovation energy (dB)
#define ENERGY_MEAN 30.0 ///< mean innovation energy (dB) in all modes
+#define PREEMPH_FAC 0.68 ///< factor used to de-emphasize synthesis
#define AMRWB_SUBFRAME_SIZE 64 ///< samples per subframe at 12.8 kHz
#define AMRWB_SFR_SIZE_OUT 80 ///< samples per subframe at 16 kHz
@@ -38,8 +42,6 @@
#define AMRWB_P_DELAY_MAX 231 ///< maximum pitch delay value
#define AMRWB_P_DELAY_MIN 34
-#define PREEMPH_FAC 0.68 ///< factor used to de-emphasize synthesis
-
/* Mode ordering is sensitive, do not change */
enum Mode {
MODE_6k60 = 0, ///< 6.60 kbit/s
@@ -1757,7 +1759,6 @@ static const float energy_pred_fac[4] = { 0.2, 0.3, 0.4, 0.5 };
/** impulse response filter tables converted to float from Q15
* used for anti-sparseness processing */
-// XXX: Not sure whether it is Q15 indeed
static const float ir_filter_str[64] = {
0.615906, 0.295807, 0.099792, -0.104889, 0.087402, -0.159912,
0.048492, -0.041412, 0.018311, 0.118805, -0.045685, -0.021301,
@@ -1802,6 +1803,35 @@ static const float hpf_400_coef[2][3] = { // 400 kHz cutoff filter
{ 1.787109375, -0.864257812, 0 }
};
+/* Interpolation coefficients for 5/4 signal upsampling
+ * Table from the reference source was reordered for efficiency */
+static const float upsample_fir[4][24] = {
+ { -6.103516e-05, 7.324219e-04, -2.014160e-03, 4.150391e-03,
+ -7.263184e-03, 1.165771e-02, -1.776123e-02, 2.624512e-02,
+ -3.869629e-02, 5.877686e-02, -9.863281e-02, 2.314453e-01,
+ 9.348755e-01, -1.523438e-01, 7.861328e-02, -4.937744e-02,
+ 3.308105e-02, -2.252197e-02, 1.507568e-02, -9.765625e-03,
+ 5.859375e-03, -3.173828e-03, 1.403809e-03, -3.662109e-04 },
+ { -2.441406e-04, 1.464844e-03, -3.784180e-03, 7.568359e-03,
+ -1.300049e-02, 2.062988e-02, -3.112793e-02, 4.589844e-02,
+ -6.781006e-02, 1.042480e-01, -1.815186e-01, 5.016479e-01,
+ 7.548828e-01, -2.094727e-01, 1.148071e-01, -7.348633e-02,
+ 4.956055e-02, -3.369141e-02, 2.246094e-02, -1.434326e-02,
+ 8.483887e-03, -4.455566e-03, 1.831055e-03, -4.272461e-04 },
+ { -4.272461e-04, 1.831055e-03, -4.455566e-03, 8.483887e-03,
+ -1.434326e-02, 2.246094e-02, -3.369141e-02, 4.956055e-02,
+ -7.348633e-02, 1.148071e-01, -2.094727e-01, 7.548828e-01,
+ 5.016479e-01, -1.815186e-01, 1.042480e-01, -6.781006e-02,
+ 4.589844e-02, -3.112793e-02, 2.062988e-02, -1.300049e-02,
+ 7.568359e-03, -3.784180e-03, 1.464844e-03, -2.441406e-04 },
+ { -3.662109e-04, 1.403809e-03, -3.173828e-03, 5.859375e-03,
+ -9.765625e-03, 1.507568e-02, -2.252197e-02, 3.308105e-02,
+ -4.937744e-02, 7.861328e-02, -1.523438e-01, 9.348755e-01,
+ 2.314453e-01, -9.863281e-02, 5.877686e-02, -3.869629e-02,
+ 2.624512e-02, -1.776123e-02, 1.165771e-02, -7.263184e-03,
+ 4.150391e-03, -2.014160e-03, 7.324219e-04, -6.103516e-05 }
+};
+
/* High band quantized gains for 23k85 in Q14 */
static const uint16_t qua_hb_gain[16] = {
3624, 4673, 5597, 6479, 7425, 8378, 9324, 10264,
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 2a62a17..d7deb39 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -72,7 +72,7 @@ typedef struct {
uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
float prev_tr_gain; ///< previous initial gain used by noise enhancer for thresold
- float samples_in[LP_ORDER + AMRWB_SUBFRAME_SIZE]; ///< floating point samples
+ float samples_in[SAMPLE_MEM + AMRWB_SUBFRAME_SIZE]; ///< lower band floating point samples at 12.8kHz
float demph_mem[1]; ///< previous value in the de-emphasis filter
float hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in the high-pass filters
@@ -961,6 +961,33 @@ static void high_pass_filter(float *out, const float hpf_coef[2][3],
}
/**
+ * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
+ * a FIR interpolation filter. Uses past data from before *in address
+ *
+ * @param out [out] buffer for interpolated signal
+ * @param in [in] current signal data (length 0.8*o_size)
+ * @param o_size [in] output signal length
+ */
+static void upsample_5_4(float *out, const float *in, int o_size)
+{
+ const float *in0 = in - UPS_FIR_SIZE + 1;
+ int i;
+
+ for (i = 0; i < o_size; i++) {
+ int int_part = (i << 2) / 5;
+ int frac_part = (i << 2) - 5 * int_part;
+
+ if (!frac_part) {
+ out[i] = in[i];
+ } else
+ out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
+ UPS_FIR_SIZE << 1);
+
+ out[i] *= 2.0; // upscale output
+ }
+}
+
+/**
* Calculate the high band gain based on encoded index (23k85 mode) or
* on the lower band speech signal and the Voice Activity Detection flag
*
@@ -1068,7 +1095,7 @@ static void update_sub_state(AMRWBContext *ctx)
memmove(&ctx->fixed_gain[0], &ctx->fixed_gain[1], 4 * sizeof(float));
memmove(&ctx->samples_in[0], &ctx->samples_in[AMRWB_SUBFRAME_SIZE],
- LP_ORDER * sizeof(float));
+ SAMPLE_MEM * sizeof(float));
}
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
@@ -1078,6 +1105,7 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AMRWBFrame *cf = &ctx->frame;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
+ float *buf_out = data;
AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing
float spare_vector[AMRWB_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing
float fixed_gain_factor; // fixed gain correction factor (gamma)
@@ -1172,22 +1200,22 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
pitch_enhancer(synth_fixed_vector, voice_fac);
synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
- synth_fixed_vector, &ctx->samples_in[LP_ORDER]);
+ synth_fixed_vector, &ctx->samples_in[SAMPLE_MEM]);
/* Synthesis speech post-processing */
- de_emphasis(&ctx->samples_in[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
-
- high_pass_filter(&ctx->samples_in[LP_ORDER], hpf_31_coef,
- ctx->hpf_31_mem, &ctx->samples_in[LP_ORDER]);
+ de_emphasis(&ctx->samples_in[SAMPLE_MEM], PREEMPH_FAC, ctx->demph_mem);
- // XXX: the 5/4 upsampling for the lower band goes in here
+ high_pass_filter(&ctx->samples_in[SAMPLE_MEM], hpf_31_coef,
+ ctx->hpf_31_mem, &ctx->samples_in[SAMPLE_MEM]);
+ upsample_5_4(buf_out + sub * AMRWB_SFR_SIZE_OUT,
+ &ctx->samples_in[UPS_FIR_SIZE], AMRWB_SFR_SIZE_OUT);
/* High frequency band generation */
- high_pass_filter(&ctx->samples_in[LP_ORDER], hpf_400_coef,
- ctx->hpf_400_mem, &ctx->samples_in[LP_ORDER]);
+ high_pass_filter(&ctx->samples_in[SAMPLE_MEM], hpf_400_coef,
+ ctx->hpf_400_mem, &ctx->samples_in[SAMPLE_MEM]);
- hb_gain = find_hb_gain(ctx, &ctx->samples_in[LP_ORDER],
+ hb_gain = find_hb_gain(ctx, &ctx->samples_in[SAMPLE_MEM],
cur_subframe->hb_gain, cf->vad);
scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
commit eff9bcbdb922197ef4720daa98bc6e01b8e4fb80
Author: Marcelo Povoa <marspeoplester(a)gmail.com>
Date: Mon Jul 26 18:03:34 2010 -0300
Remove overflow treatment from synthesis (it was NB specific)
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 8768863..2a62a17 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -872,7 +872,6 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac)
/**
* Conduct 16th order linear predictive coding synthesis from excitation
- * Return a overflow detection flag
*
* @param ctx [in] pointer to the AMRWBContext
* @param lpc [in] pointer to the LPC coefficients
@@ -880,25 +879,17 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac)
* @param fixed_gain [in] fixed codebook gain for synthesis
* @param fixed_vector [in] algebraic codebook vector
* @param samples [out] pointer to the output speech samples
- * @param overflow [in] 16-bit predicted overflow flag
*/
-static uint8_t synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
+static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
float fixed_gain, const float *fixed_vector,
- float *samples, uint8_t overflow)
+ float *samples)
{
- int i;
-
- // if an overflow has been detected, the pitch vector is scaled down by a
- // factor of 4
- if (overflow)
- for (i = 0; i < AMRWB_SUBFRAME_SIZE; i++)
- ctx->pitch_vector[i] *= 0.25;
-
ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
ctx->pitch_gain[4], fixed_gain, AMRWB_SUBFRAME_SIZE);
// emphasize pitch vector contribution in low bitrate modes
- if (ctx->pitch_gain[4] > 0.5 && !overflow && ctx->fr_cur_mode <= MODE_8k85) {
+ if (ctx->pitch_gain[4] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
+ int i;
float energy = ff_dot_productf(excitation, excitation,
AMRWB_SUBFRAME_SIZE);
@@ -915,14 +906,6 @@ static uint8_t synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
AMRWB_SUBFRAME_SIZE, LP_ORDER);
-
- // detect overflow
- for (i = 0; i < AMRWB_SUBFRAME_SIZE; i++)
- if (fabsf(samples[i]) > AMRWB_SAMPLE_BOUND) {
- return 1;
- }
-
- return 0;
}
/**
@@ -1083,6 +1066,9 @@ static void update_sub_state(AMRWBContext *ctx)
memmove(&ctx->pitch_gain[0], &ctx->pitch_gain[1], 4 * sizeof(float));
memmove(&ctx->fixed_gain[0], &ctx->fixed_gain[1], 4 * sizeof(float));
+
+ memmove(&ctx->samples_in[0], &ctx->samples_in[AMRWB_SUBFRAME_SIZE],
+ LP_ORDER * sizeof(float));
}
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
@@ -1174,7 +1160,6 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
}
/* Post-processing of excitation elements */
- /* XXX: Noise_enhancer was not tested yet, needs that 1.5dB check */
synth_fixed_gain = noise_enhancer(ctx->fixed_gain[4], &ctx->prev_tr_gain,
voice_fac, stab_fac);
@@ -1186,13 +1171,8 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
pitch_enhancer(synth_fixed_vector, voice_fac);
- if (synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
- synth_fixed_vector, &ctx->samples_in[LP_ORDER], 0))
- // overflow detected -> rerun synthesis scaling pitch vector down
- // by a factor of 4, skipping pitch vector contribution emphasis
- // and adaptive gain control
- synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
- synth_fixed_vector, &ctx->samples_in[LP_ORDER], 1);
+ synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
+ synth_fixed_vector, &ctx->samples_in[LP_ORDER]);
/* Synthesis speech post-processing */
de_emphasis(&ctx->samples_in[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
commit fe78e3f003f346c235c38403442d8f4f9f541afd
Author: Marcelo Povoa <marspeoplester(a)gmail.com>
Date: Mon Jul 26 17:57:28 2010 -0300
Change the stability factor multiplier to 512,
accurate according to some tests
diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
index 208c7c4..8768863 100644
--- a/libavcodec/amrwbdec.c
+++ b/libavcodec/amrwbdec.c
@@ -78,7 +78,7 @@ typedef struct {
float hpf_31_mem[4], hpf_400_mem[4]; ///< previous values in the high-pass filters
AVLFG prng; ///< random number generator for white noise excitation
- uint8_t first_frame; ///< flag active in the first frame decoded
+ uint8_t first_frame; ///< flag active during decoding of the first frame
} AMRWBContext;
static av_cold int amrwb_decode_init(AVCodecContext *avctx)
@@ -437,7 +437,7 @@ static void decode_pitch_vector(AMRWBContext *ctx,
LP_ORDER, AMRWB_SUBFRAME_SIZE);
/* Check which pitch signal path should be used.
- * 6k60 and 8k85 modes have ltp flag set to 0 */
+ * 6k60 and 8k85 modes have the ltp flag set to 0 */
if (amr_subframe->ltp) {
memcpy(ctx->pitch_vector, exc, AMRWB_SUBFRAME_SIZE * sizeof(float));
} else {
@@ -805,8 +805,8 @@ static float stability_factor(const float *isf, const float *isf_past)
acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
// XXX: I could not understand well this part from ref code
- // it made more sense changing the "/ 256" to "* 256"
- return FFMAX(0.0, 1.25 - acc * 0.8 * 256);
+ // it made more sense changing the "/ 256" to "* 512"
+ return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
}
/**
@@ -855,6 +855,9 @@ static void pitch_enhancer(float *fixed_vector, float voice_fac)
float cpe = 0.125 * (1 + voice_fac);
float last = fixed_vector[0]; // holds c(i - 1)
+ /* XXX: This procedure seems correct, but due to some roundings
+ * in the opencore code (line 1037 onwards) the resulting fixed_vector
+ * differs quite a bit between the two implementations */
fixed_vector[0] -= cpe * fixed_vector[1];
for (i = 1; i < AMRWB_SUBFRAME_SIZE - 1; i++) {
-----------------------------------------------------------------------
Summary of changes:
libavcodec/amrwbdata.h | 36 +++++++++++++++++--
libavcodec/amrwbdec.c | 95 +++++++++++++++++++++++++++---------------------
2 files changed, 86 insertions(+), 45 deletions(-)
hooks/post-receive
--
AMR-WB decoder
