/*
- * G.729 decoder
+ * G.729, G729 Annex D decoders
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include <stdlib.h>
+
#include <inttypes.h>
-#include <limits.h>
-#include <stdio.h>
#include <string.h>
-#include <math.h>
-#include <assert.h>
#include "avcodec.h"
#include "libavutil/avutil.h"
#include "g729.h"
#include "lsp.h"
#include "celp_math.h"
+#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_pitch_delay.h"
#include "acelp_vectors.h"
#include "g729data.h"
+#include "g729postfilter.h"
/**
* minimum quantized LSF value (3.2.4)
*/
#define LSFQ_DIFF_MIN 321
+/// interpolation filter length
+#define INTERPOL_LEN 11
+
/**
* minimum gain pitch value (3.8, Equation 47)
* 0.2 in (1.14)
*/
#define MR_ENERGY 1018156
+#define DECISION_NOISE 0
+#define DECISION_INTERMEDIATE 1
+#define DECISION_VOICE 2
+
typedef enum {
FORMAT_G729_8K = 0,
FORMAT_G729D_6K4,
typedef struct {
DSPContext dsp;
+ /// past excitation signal buffer
+ int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
+
+ int16_t* exc; ///< start of past excitation data in buffer
int pitch_delay_int_prev; ///< integer part of previous subframe's pitch delay (4.1.3)
/// (2.13) LSP quantizer outputs
int16_t quant_energy[4]; ///< (5.10) past quantized energy
- /// (1.14) pitch gain of previous subframe
- int16_t gain_pitch;
+ /// previous speech data for LP synthesis filter
+ int16_t syn_filter_data[10];
+
+
+ /// residual signal buffer (used in long-term postfilter)
+ int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
+
+ /// previous speech data for residual calculation filter
+ int16_t res_filter_data[SUBFRAME_SIZE+10];
- /// (14.1) gain code from previous subframe
- int16_t gain_code;
+ /// previous speech data for short-term postfilter
+ int16_t pos_filter_data[SUBFRAME_SIZE+10];
+ /// (1.14) pitch gain of current and five previous subframes
+ int16_t past_gain_pitch[6];
+
+ /// (14.1) gain code from current and previous subframe
+ int16_t past_gain_code[2];
+
+ /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
+ int16_t voice_decision;
+
+ int16_t onset; ///< detected onset level (0-2)
+ int16_t was_periodic; ///< whether previous frame was declared as periodic or not (4.4)
+ int16_t ht_prev_data; ///< previous data for 4.2.3, equation 86
+ int gain_coeff; ///< (1.14) gain coefficient (4.2.4)
uint16_t rand_value; ///< random number generator value (4.4.4)
int ma_predictor_prev; ///< switched MA predictor of LSP quantizer from last good frame
+
+ /// (14.14) high-pass filter data (past input)
+ int hpf_f[2];
+
+ /// high-pass filter data (past output)
+ int16_t hpf_z[2];
} G729Context;
static const G729FormatDescription format_g729_8k = {
}
}
+/**
+ * Constructs new excitation signal and applies phase filter to it
+ * @param out[out] constructed speech signal
+ * @param in original excitation signal
+ * @param fc_cur (2.13) original fixed-codebook vector
+ * @param gain_code (14.1) gain code
+ * @param subframe_size length of the subframe
+ */
+static void g729d_get_new_exc(
+ int16_t* out,
+ const int16_t* in,
+ const int16_t* fc_cur,
+ int dstate,
+ int gain_code,
+ int subframe_size)
+{
+ int i;
+ int16_t fc_new[SUBFRAME_SIZE];
+
+ ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
+
+ for(i=0; i<subframe_size; i++)
+ {
+ out[i] = in[i];
+ out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
+ out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
+ }
+}
+
+/**
+ * Makes decision about onset in current subframe
+ * @param past_onset decision result of previous subframe
+ * @param past_gain_code gain code of current and previous subframe
+ *
+ * @return onset decision result for current subframe
+ */
+static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
+{
+ if((past_gain_code[0] >> 1) > past_gain_code[1])
+ return 2;
+ else
+ return FFMAX(past_onset-1, 0);
+}
+
+/**
+ * Makes decision about voice presence in current subframe
+ * @param onset onset level
+ * @param prev_voice_decision voice decision result from previous subframe
+ * @param past_gain_pitch pitch gain of current and previous subframes
+ *
+ * @return voice decision result for current subframe
+ */
+static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
+{
+ int i, low_gain_pitch_cnt, voice_decision;
+
+ if(past_gain_pitch[0] >= 14745) // 0.9
+ voice_decision = DECISION_VOICE;
+ else if (past_gain_pitch[0] <= 9830) // 0.6
+ voice_decision = DECISION_NOISE;
+ else
+ voice_decision = DECISION_INTERMEDIATE;
+
+ for(i=0, low_gain_pitch_cnt=0; i<6; i++)
+ if(past_gain_pitch[i] < 9830)
+ low_gain_pitch_cnt++;
+
+ if(low_gain_pitch_cnt > 2 && !onset)
+ voice_decision = DECISION_NOISE;
+
+ if(!onset && voice_decision > prev_voice_decision + 1)
+ voice_decision--;
+
+ if(onset && voice_decision < DECISION_VOICE)
+ voice_decision++;
+
+ return voice_decision;
+}
+
+static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order, int shift)
+{
+ int res = 0;
+
+ while (order--)
+ res += (*v1++ * *v2++) >> shift;
+
+ return res;
+}
+
static av_cold int decoder_init(AVCodecContext * avctx)
{
G729Context* ctx = avctx->priv_data;
av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
return AVERROR(EINVAL);
}
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
avctx->frame_size = SUBFRAME_SIZE << 1;
+ ctx->gain_coeff = 16384; // 1.0 in (1.14)
+
for (k = 0; k < MA_NP + 1; k++) {
ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
for (i = 1; i < 11; i++)
ctx->lsp[1] = ctx->lsp_buf[1];
memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
+ ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
+
/* random seed initialization */
ctx->rand_value = 21845;
ctx->quant_energy[i] = -14336; // -14 in (5.10)
dsputil_init(&ctx->dsp, avctx);
+ ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
return 0;
}
int buf_size = avpkt->size;
int16_t *out_frame = data;
GetBitContext gb;
- G729FormatDescription format;
+ const G729FormatDescription *format;
int frame_erasure = 0; ///< frame erasure detected during decoding
int bad_pitch = 0; ///< parity check failed
int i;
uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
- int pitch_delay_int; // pitch delay, integer part
+ int pitch_delay_int[2]; // pitch delay, integer part
int pitch_delay_3x; // pitch delay, multiplied by 3
int16_t fc[SUBFRAME_SIZE]; // fixed-codebook vector
+ int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
+ int j;
+ int gain_before, gain_after;
+ int is_periodic = 0; // whether one of the subframes is declared as periodic or not
if (*data_size < SUBFRAME_SIZE << 2) {
av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
if (buf_size == 10) {
packet_type = FORMAT_G729_8K;
- format = format_g729_8k;
+ format = &format_g729_8k;
+ //Reset voice decision
+ ctx->onset = 0;
+ ctx->voice_decision = DECISION_VOICE;
av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
} else if (buf_size == 8) {
packet_type = FORMAT_G729D_6K4;
- format = format_g729d_6k4;
+ format = &format_g729d_6k4;
av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
} else {
av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
frame_erasure |= buf[i];
frame_erasure = !frame_erasure;
- init_get_bits(&gb, buf, buf_size);
+ init_get_bits(&gb, buf, 8*buf_size);
ma_predictor = get_bits(&gb, 1);
quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
uint8_t gc_1st_index; ///< gain codebook (first stage) index
uint8_t gc_2nd_index; ///< gain codebook (second stage) index
- ac_index = get_bits(&gb, format.ac_index_bits[i]);
- if(!i && format.parity_bit)
+ ac_index = get_bits(&gb, format->ac_index_bits[i]);
+ if(!i && format->parity_bit)
bad_pitch = get_parity(ac_index) == get_bits1(&gb);
- fc_indexes = get_bits(&gb, format.fc_indexes_bits);
- pulses_signs = get_bits(&gb, format.fc_signs_bits);
- gc_1st_index = get_bits(&gb, format.gc_1st_index_bits);
- gc_2nd_index = get_bits(&gb, format.gc_2nd_index_bits);
+ fc_indexes = get_bits(&gb, format->fc_indexes_bits);
+ pulses_signs = get_bits(&gb, format->fc_signs_bits);
+ gc_1st_index = get_bits(&gb, format->gc_1st_index_bits);
+ gc_2nd_index = get_bits(&gb, format->gc_2nd_index_bits);
if (frame_erasure)
pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
}
/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
- pitch_delay_int = (pitch_delay_3x + 1) / 3;
+ pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
if (frame_erasure) {
ctx->rand_value = g729_prng(ctx->rand_value);
- fc_indexes = ctx->rand_value & ((1 << format.fc_indexes_bits) - 1);
+ fc_indexes = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
ctx->rand_value = g729_prng(ctx->rand_value);
pulses_signs = ctx->rand_value;
fc_v[i] = <
\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
*/
- ff_acelp_weighted_vector_sum(fc + pitch_delay_int,
- fc + pitch_delay_int,
+ ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
+ fc + pitch_delay_int[i],
fc, 1 << 14,
- av_clip(ctx->gain_pitch, SHARP_MIN, SHARP_MAX),
+ av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
0, 14,
- SUBFRAME_SIZE - pitch_delay_int);
+ SUBFRAME_SIZE - pitch_delay_int[i]);
+
+ memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
+ ctx->past_gain_code[1] = ctx->past_gain_code[0];
if (frame_erasure) {
- ctx->gain_pitch = (29491 * ctx->gain_pitch) >> 15; // 0.90 (0.15)
- ctx->gain_code = ( 2007 * ctx->gain_code ) >> 11; // 0.98 (0.11)
+ ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
+ ctx->past_gain_code[0] = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
gain_corr_factor = 0;
} else {
- ctx->gain_pitch = cb_gain_1st_8k[gc_1st_index][0] +
- cb_gain_2nd_8k[gc_2nd_index][0];
- gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
- cb_gain_2nd_8k[gc_2nd_index][1];
+ if (packet_type == FORMAT_G729D_6K4) {
+ ctx->past_gain_pitch[0] = cb_gain_1st_6k4[gc_1st_index][0] +
+ cb_gain_2nd_6k4[gc_2nd_index][0];
+ gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
+ cb_gain_2nd_6k4[gc_2nd_index][1];
+
+ /* Without check below overflow can occure in ff_acelp_update_past_gain.
+ It is not issue for G.729, because gain_corr_factor in it's case is always
+ greater than 1024, while in G.729D it can be even zero. */
+ gain_corr_factor = FFMAX(gain_corr_factor, 1024);
+#ifndef G729_BITEXACT
+ gain_corr_factor >>= 1;
+#endif
+ } else {
+ ctx->past_gain_pitch[0] = cb_gain_1st_8k[gc_1st_index][0] +
+ cb_gain_2nd_8k[gc_2nd_index][0];
+ gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
+ cb_gain_2nd_8k[gc_2nd_index][1];
+ }
/* Decode the fixed-codebook gain. */
- ctx->gain_code = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
- fc, MR_ENERGY,
- ctx->quant_energy,
- ma_prediction_coeff,
- SUBFRAME_SIZE, 4);
+ ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
+ fc, MR_ENERGY,
+ ctx->quant_energy,
+ ma_prediction_coeff,
+ SUBFRAME_SIZE, 4);
+#ifdef G729_BITEXACT
+ /*
+ This correction required to get bit-exact result with
+ reference code, because gain_corr_factor in G.729D is
+ two times larger than in original G.729.
+
+ If bit-exact result is not issue then gain_corr_factor
+ can be simpler devided by 2 before call to g729_get_gain_code
+ instead of using correction below.
+ */
+ if (packet_type == FORMAT_G729D_6K4) {
+ gain_corr_factor >>= 1;
+ ctx->past_gain_code[0] >>= 1;
+ }
+#endif
}
ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
+ /* Routine requires rounding to lowest. */
+ ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
+ ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
+ ff_acelp_interp_filter, 6,
+ (pitch_delay_3x % 3) << 1,
+ 10, SUBFRAME_SIZE);
+
ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
ctx->exc + i * SUBFRAME_SIZE, fc,
- (!voicing && frame_erasure) ? 0 : ctx->gain_pitch,
- ( voicing && frame_erasure) ? 0 : ctx->gain_code,
+ (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
+ ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
1 << 13, 14, SUBFRAME_SIZE);
+ memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
+
+ if (ff_celp_lp_synthesis_filter(
+ synth+10,
+ &lp[i][1],
+ ctx->exc + i * SUBFRAME_SIZE,
+ SUBFRAME_SIZE,
+ 10,
+ 1,
+ 0,
+ 0x800))
+ /* Overflow occured, downscale excitation signal... */
+ for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
+ ctx->exc_base[j] >>= 2;
+
+ /* ... and make synthesis again. */
+ if (packet_type == FORMAT_G729D_6K4) {
+ int16_t exc_new[SUBFRAME_SIZE];
+
+ ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
+ ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
+
+ g729d_get_new_exc(exc_new, ctx->exc + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
+
+ ff_celp_lp_synthesis_filter(
+ synth+10,
+ &lp[i][1],
+ exc_new,
+ SUBFRAME_SIZE,
+ 10,
+ 0,
+ 0,
+ 0x800);
+ } else {
+ ff_celp_lp_synthesis_filter(
+ synth+10,
+ &lp[i][1],
+ ctx->exc + i * SUBFRAME_SIZE,
+ SUBFRAME_SIZE,
+ 10,
+ 0,
+ 0,
+ 0x800);
+ }
+ /* Save data (without postfilter) for use in next subframe. */
+ memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
+
+ /* Calculate gain of unfiltered signal for use in AGC. */
+ gain_before = 0;
+ for (j = 0; j < SUBFRAME_SIZE; j++)
+ gain_before += FFABS(synth[j+10]);
+
+ /* Call postfilter and also update voicing decision for use in next frame. */
+ ff_g729_postfilter(
+ &ctx->dsp,
+ &ctx->ht_prev_data,
+ &is_periodic,
+ &lp[i][0],
+ pitch_delay_int[0],
+ ctx->residual,
+ ctx->res_filter_data,
+ ctx->pos_filter_data,
+ synth+10,
+ SUBFRAME_SIZE);
+
+ /* Calculate gain of filtered signal for use in AGC. */
+ gain_after = 0;
+ for(j=0; j<SUBFRAME_SIZE; j++)
+ gain_after += FFABS(synth[j+10]);
+
+ ctx->gain_coeff = ff_g729_adaptive_gain_control(
+ gain_before,
+ gain_after,
+ synth+10,
+ SUBFRAME_SIZE,
+ ctx->gain_coeff);
+
if (frame_erasure)
ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
else
- ctx->pitch_delay_int_prev = pitch_delay_int;
+ ctx->pitch_delay_int_prev = pitch_delay_int[i];
+
+ memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
+ ff_acelp_high_pass_filter(
+ out_frame + i*SUBFRAME_SIZE,
+ ctx->hpf_f,
+ synth+10,
+ SUBFRAME_SIZE);
+ memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
}
+ ctx->was_periodic = is_periodic;
+
+ /* Save signal for use in next frame. */
+ memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
+
*data_size = SUBFRAME_SIZE << 2;
return buf_size;
}
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