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FFmpeg/libavcodec/g729dec.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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/*
* G.729, G729 Annex D decoders
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include <string.h>
#include "avcodec.h"
#include "libavutil/avutil.h"
#include "libavutil/mem.h"
#include "get_bits.h"
#include "audiodsp.h"
#include "codec_internal.h"
#include "decode.h"
#include "g729.h"
#include "lsp.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)
* 0.005 in Q13
*/
#define LSFQ_MIN 40
/**
* maximum quantized LSF value (3.2.4)
* 3.135 in Q13
*/
#define LSFQ_MAX 25681
/**
* minimum LSF distance (3.2.4)
* 0.0391 in Q13
*/
#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 SHARP_MIN 3277
/**
* maximum gain pitch value (3.8, Equation 47)
* (EE) This does not comply with the specification.
* Specification says about 0.8, which should be
* 13107 in (1.14), but reference C code uses
* 13017 (equals to 0.7945) instead of it.
*/
#define SHARP_MAX 13017
/**
* MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26 * subframe_size) in (7.13)
*/
#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,
FORMAT_COUNT,
} G729Formats;
typedef struct {
uint8_t ac_index_bits[2]; ///< adaptive codebook index for second subframe (size in bits)
uint8_t parity_bit; ///< parity bit for pitch delay
uint8_t gc_1st_index_bits; ///< gain codebook (first stage) index (size in bits)
uint8_t gc_2nd_index_bits; ///< gain codebook (second stage) index (size in bits)
uint8_t fc_signs_bits; ///< number of pulses in fixed-codebook vector
uint8_t fc_indexes_bits; ///< size (in bits) of fixed-codebook index entry
uint8_t block_size;
} G729FormatDescription;
typedef struct {
/// 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 past_quantizer_output_buf[MA_NP + 1][10];
int16_t* past_quantizer_outputs[MA_NP + 1];
int16_t lsfq[10]; ///< (2.13) quantized LSF coefficients from previous frame
int16_t lsp_buf[2][10]; ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
int16_t *lsp[2]; ///< pointers to lsp_buf
int16_t quant_energy[4]; ///< (5.10) past quantized energy
/// 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];
/// 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];
} G729ChannelContext;
typedef struct {
AudioDSPContext adsp;
G729ChannelContext *channel_context;
} G729Context;
static const G729FormatDescription format_g729_8k = {
.ac_index_bits = {8,5},
.parity_bit = 1,
.gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
.gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
.fc_signs_bits = 4,
.fc_indexes_bits = 13,
.block_size = G729_8K_BLOCK_SIZE,
};
static const G729FormatDescription format_g729d_6k4 = {
.ac_index_bits = {8,4},
.parity_bit = 0,
.gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
.gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
.fc_signs_bits = 2,
.fc_indexes_bits = 9,
.block_size = G729D_6K4_BLOCK_SIZE,
};
/**
* @brief pseudo random number generator
*/
static inline uint16_t g729_prng(uint16_t value)
{
return 31821 * value + 13849;
}
/**
* Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
* @param[out] lsfq (2.13) quantized LSF coefficients
* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
* @param ma_predictor switched MA predictor of LSP quantizer
* @param vq_1st first stage vector of quantizer
* @param vq_2nd_low second stage lower vector of LSP quantizer
* @param vq_2nd_high second stage higher vector of LSP quantizer
*/
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
int16_t ma_predictor,
int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
{
int i,j;
static const uint8_t min_distance[2]={10, 5}; //(2.13)
int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
for (i = 0; i < 5; i++) {
quantizer_output[i] = cb_lsp_1st[vq_1st][i ] + cb_lsp_2nd[vq_2nd_low ][i ];
quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
}
for (j = 0; j < 2; j++) {
for (i = 1; i < 10; i++) {
int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
if (diff > 0) {
quantizer_output[i - 1] -= diff;
quantizer_output[i ] += diff;
}
}
}
for (i = 0; i < 10; i++) {
int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
for (j = 0; j < MA_NP; j++)
sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
lsfq[i] = sum >> 15;
}
ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
}
/**
* Restores past LSP quantizer output using LSF from previous frame
* @param[in,out] lsfq (2.13) quantized LSF coefficients
* @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
* @param ma_predictor_prev MA predictor from previous frame
* @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
*/
static void lsf_restore_from_previous(int16_t* lsfq,
int16_t* past_quantizer_outputs[MA_NP + 1],
int ma_predictor_prev)
{
int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
int i,k;
for (i = 0; i < 10; i++) {
int tmp = lsfq[i] << 15;
for (k = 0; k < MA_NP; k++)
tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
}
}
/**
* 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;
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)
{
int64_t res = 0;
while (order--)
res += *v1++ * *v2++;
if (res > INT32_MAX) return INT32_MAX;
else if (res < INT32_MIN) return INT32_MIN;
return res;
}
static av_cold int decoder_init(AVCodecContext * avctx)
{
G729Context *s = avctx->priv_data;
G729ChannelContext *ctx;
int channels = avctx->ch_layout.nb_channels;
int c,i,k;
if (channels < 1 || channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", channels);
return AVERROR(EINVAL);
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
/* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
avctx->frame_size = SUBFRAME_SIZE << 1;
ctx =
s->channel_context = av_mallocz(sizeof(G729ChannelContext) * channels);
if (!ctx)
return AVERROR(ENOMEM);
for (c = 0; c < channels; c++) {
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->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
}
ctx->lsp[0] = ctx->lsp_buf[0];
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];
ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
/* random seed initialization */
ctx->rand_value = 21845;
/* quantized prediction error */
for (i = 0; i < 4; i++)
ctx->quant_energy[i] = -14336; // -14 in (5.10)
ctx++;
}
ff_audiodsp_init(&s->adsp);
s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
return 0;
}
static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t *out_frame;
GetBitContext gb;
const G729FormatDescription *format;
int c, i;
int16_t *tmp;
G729Formats packet_type;
G729Context *s = avctx->priv_data;
G729ChannelContext *ctx = s->channel_context;
int channels = avctx->ch_layout.nb_channels;
int16_t lp[2][11]; // (3.12)
uint8_t ma_predictor; ///< switched MA predictor of LSP quantizer
uint8_t quantizer_1st; ///< first stage vector of quantizer
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[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, ret;
int gain_before, gain_after;
frame->nb_samples = SUBFRAME_SIZE<<1;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels) == 0) {
packet_type = 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 == G729D_6K4_BLOCK_SIZE * channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
packet_type = 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);
return AVERROR_INVALIDDATA;
}
for (c = 0; c < channels; c++) {
int frame_erasure = 0; ///< frame erasure detected during decoding
int bad_pitch = 0; ///< parity check failed
int is_periodic = 0; ///< whether one of the subframes is declared as periodic or not
out_frame = (int16_t*)frame->data[c];
if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
if (*buf != ((avctx->ch_layout.nb_channels - 1 - c) * 0x80 | 2))
avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
buf++;
}
for (i = 0; i < format->block_size; i++)
frame_erasure |= buf[i];
frame_erasure = !frame_erasure;
init_get_bits8(&gb, buf, format->block_size);
ma_predictor = get_bits(&gb, 1);
quantizer_1st = get_bits(&gb, VQ_1ST_BITS);
quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
if (frame_erasure) {
lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
ctx->ma_predictor_prev);
} else {
lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
ma_predictor,
quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
ctx->ma_predictor_prev = ma_predictor;
}
tmp = ctx->past_quantizer_outputs[MA_NP];
memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
MA_NP * sizeof(int16_t*));
ctx->past_quantizer_outputs[0] = tmp;
ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
for (i = 0; i < 2; i++) {
int gain_corr_factor;
uint8_t ac_index; ///< adaptive codebook index
uint8_t pulses_signs; ///< fixed-codebook vector pulse signs
int fc_indexes; ///< fixed-codebook indexes
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)
bad_pitch = av_parity(ac_index >> 2) == 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);
if (frame_erasure) {
pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
} else if (!i) {
if (bad_pitch) {
pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
} else {
pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
}
} else {
int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
if (packet_type == FORMAT_G729D_6K4) {
pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
} else {
pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
}
}
/* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
pitch_delay_int[i] = (pitch_delay_3x + 1) / 3;
if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
pitch_delay_int[i] = PITCH_DELAY_MAX;
}
if (frame_erasure) {
ctx->rand_value = g729_prng(ctx->rand_value);
fc_indexes = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
ctx->rand_value = g729_prng(ctx->rand_value);
pulses_signs = ctx->rand_value;
}
memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
switch (packet_type) {
case FORMAT_G729_8K:
ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
ff_fc_4pulses_8bits_track_4,
fc_indexes, pulses_signs, 3, 3);
break;
case FORMAT_G729D_6K4:
ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
ff_fc_2pulses_9bits_track2_gray,
fc_indexes, pulses_signs, 1, 4);
break;
}
/*
This filter enhances harmonic components of the fixed-codebook vector to
improve the quality of the reconstructed speech.
/ fc_v[i], i < pitch_delay
fc_v[i] = <
\ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
*/
if (SUBFRAME_SIZE > pitch_delay_int[i])
ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
fc + pitch_delay_int[i],
fc, 1 << 14,
av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
0, 14,
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->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 {
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 occur 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->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, 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 divided 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,
(!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 occurred, 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(
&s->adsp,
&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[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));
buf += format->block_size;
ctx++;
}
*got_frame_ptr = 1;
return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels;
}
static av_cold int decode_close(AVCodecContext *avctx)
{
G729Context *s = avctx->priv_data;
av_freep(&s->channel_context);
return 0;
}
const FFCodec ff_g729_decoder = {
.p.name = "g729",
CODEC_LONG_NAME("G.729"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_G729,
.priv_data_size = sizeof(G729Context),
.init = decoder_init,
FF_CODEC_DECODE_CB(decode_frame),
.close = decode_close,
.p.capabilities =
#if FF_API_SUBFRAMES
AV_CODEC_CAP_SUBFRAMES |
#endif
AV_CODEC_CAP_DR1,
};
const FFCodec ff_acelp_kelvin_decoder = {
.p.name = "acelp.kelvin",
CODEC_LONG_NAME("Sipro ACELP.KELVIN"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ACELP_KELVIN,
.priv_data_size = sizeof(G729Context),
.init = decoder_init,
FF_CODEC_DECODE_CB(decode_frame),
.close = decode_close,
.p.capabilities =
#if FF_API_SUBFRAMES
AV_CODEC_CAP_SUBFRAMES |
#endif
AV_CODEC_CAP_DR1,
};
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