#define CNG_RANDOM_SEED 12345
+/**
+ * Postfilter gain weighting factors scaled by 2^15
+ */
+static const int16_t ppf_gain_weight[2] = {0x1800, 0x2000};
+
+static const int16_t pitch_contrib[340] = {
+ 60, 0, 0, 2489, 60, 0, 0, 5217,
+ 1, 6171, 0, 3953, 0, 10364, 1, 9357,
+ -1, 8843, 1, 9396, 0, 5794, -1, 10816,
+ 2, 11606, -2, 12072, 0, 8616, 1, 12170,
+ 0, 14440, 0, 7787, -1, 13721, 0, 18205,
+ 0, 14471, 0, 15807, 1, 15275, 0, 13480,
+ -1, 18375, -1, 0, 1, 11194, -1, 13010,
+ 1, 18836, -2, 20354, 1, 16233, -1, 0,
+ 60, 0, 0, 12130, 0, 13385, 1, 17834,
+ 1, 20875, 0, 21996, 1, 0, 1, 18277,
+ -1, 21321, 1, 13738, -1, 19094, -1, 20387,
+ -1, 0, 0, 21008, 60, 0, -2, 22807,
+ 0, 15900, 1, 0, 0, 17989, -1, 22259,
+ 1, 24395, 1, 23138, 0, 23948, 1, 22997,
+ 2, 22604, -1, 25942, 0, 26246, 1, 25321,
+ 0, 26423, 0, 24061, 0, 27247, 60, 0,
+ -1, 25572, 1, 23918, 1, 25930, 2, 26408,
+ -1, 19049, 1, 27357, -1, 24538, 60, 0,
+ -1, 25093, 0, 28549, 1, 0, 0, 22793,
+ -1, 25659, 0, 29377, 0, 30276, 0, 26198,
+ 1, 22521, -1, 28919, 0, 27384, 1, 30162,
+ -1, 0, 0, 24237, -1, 30062, 0, 21763,
+ 1, 30917, 60, 0, 0, 31284, 0, 29433,
+ 1, 26821, 1, 28655, 0, 31327, 2, 30799,
+ 1, 31389, 0, 32322, 1, 31760, -2, 31830,
+ 0, 26936, -1, 31180, 1, 30875, 0, 27873,
+ -1, 30429, 1, 31050, 0, 0, 0, 31912,
+ 1, 31611, 0, 31565, 0, 25557, 0, 31357,
+ 60, 0, 1, 29536, 1, 28985, -1, 26984,
+ -1, 31587, 2, 30836, -2, 31133, 0, 30243,
+ -1, 30742, -1, 32090, 60, 0, 2, 30902,
+ 60, 0, 0, 30027, 0, 29042, 60, 0,
+ 0, 31756, 0, 24553, 0, 25636, -2, 30501,
+ 60, 0, -1, 29617, 0, 30649, 60, 0,
+ 0, 29274, 2, 30415, 0, 27480, 0, 31213,
+ -1, 28147, 0, 30600, 1, 31652, 2, 29068,
+ 60, 0, 1, 28571, 1, 28730, 1, 31422,
+ 0, 28257, 0, 24797, 60, 0, 0, 0,
+ 60, 0, 0, 22105, 0, 27852, 60, 0,
+ 60, 0, -1, 24214, 0, 24642, 0, 23305,
+ 60, 0, 60, 0, 1, 22883, 0, 21601,
+ 60, 0, 2, 25650, 60, 0, -2, 31253,
+ -2, 25144, 0, 17998
+};
+
+/**
+ * Size of the MP-MLQ fixed excitation codebooks
+ */
+static const int32_t max_pos[4] = {593775, 142506, 593775, 142506};
+
+/**
+ * 0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15
+ */
+static const int16_t postfilter_tbl[2][LPC_ORDER] = {
+ /* Zero */
+ {21299, 13844, 8999, 5849, 3802, 2471, 1606, 1044, 679, 441},
+ /* Pole */
+ {24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845}
+};
+
+static const int cng_adaptive_cb_lag[4] = { 1, 0, 1, 3 };
+
+static const int cng_filt[4] = { 273, 998, 499, 333 };
+
+static const int cng_bseg[3] = { 2048, 18432, 231233 };
+
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
{
- G723_1_Context *p = avctx->priv_data;
+ G723_1_Context *s = avctx->priv_data;
+
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ if (avctx->channels < 1 || avctx->channels > 2) {
+ av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
+ return AVERROR(EINVAL);
+ }
+ avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
+ for (int ch = 0; ch < avctx->channels; ch++) {
+ G723_1_ChannelContext *p = &s->ch[ch];
- avctx->channel_layout = AV_CH_LAYOUT_MONO;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- avctx->channels = 1;
- p->pf_gain = 1 << 12;
+ p->pf_gain = 1 << 12;
- memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
+ memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+ memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
- p->cng_random_seed = CNG_RANDOM_SEED;
- p->past_frame_type = SID_FRAME;
+ p->cng_random_seed = CNG_RANDOM_SEED;
+ p->past_frame_type = SID_FRAME;
+ }
return 0;
}
* @param buf pointer to the input buffer
* @param buf_size size of the input buffer
*/
-static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
+static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf,
int buf_size)
{
GetBitContext gb;
int ad_cb_len;
int temp, info_bits, i;
+ int ret;
- init_get_bits(&gb, buf, buf_size * 8);
+ ret = init_get_bits8(&gb, buf, buf_size);
+ if (ret < 0)
+ return ret;
/* Extract frame type and rate info */
info_bits = get_bits(&gb, 2);
j = PULSE_MAX - pulses[index];
temp = subfrm->pulse_pos;
for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
- temp -= combinatorial_table[j][i];
+ temp -= ff_g723_1_combinatorial_table[j][i];
if (temp >= 0)
continue;
- temp += combinatorial_table[j++][i];
+ temp += ff_g723_1_combinatorial_table[j++][i];
if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
vector[subfrm->grid_index + GRID_SIZE * i] =
- -fixed_cb_gain[subfrm->amp_index];
+ -ff_g723_1_fixed_cb_gain[subfrm->amp_index];
} else {
vector[subfrm->grid_index + GRID_SIZE * i] =
- fixed_cb_gain[subfrm->amp_index];
+ ff_g723_1_fixed_cb_gain[subfrm->amp_index];
}
if (j == PULSE_MAX)
break;
if (subfrm->dirac_train == 1)
ff_g723_1_gen_dirac_train(vector, pitch_lag);
} else { /* 5300 bps */
- int cb_gain = fixed_cb_gain[subfrm->amp_index];
+ int cb_gain = ff_g723_1_fixed_cb_gain[subfrm->amp_index];
int cb_shift = subfrm->grid_index;
int cb_sign = subfrm->pulse_sign;
int cb_pos = subfrm->pulse_pos;
* @param ppf pitch postfilter parameters
* @param cur_rate current bitrate
*/
-static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
+static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
PPFParam *ppf, enum Rate cur_rate)
{
*
* @return residual interpolation index if voiced, 0 otherwise
*/
-static int comp_interp_index(G723_1_Context *p, int pitch_lag,
+static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
int *exc_eng, int *scale)
{
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
* @param buf postfiltered output vector
* @param energy input energy coefficient
*/
-static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
+static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
{
int num, denom, gain, bits1, bits2;
int i;
* @param buf input buffer
* @param dst output buffer
*/
-static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
+static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
int16_t *buf, int16_t *dst)
{
int16_t filter_coef[2][LPC_ORDER];
return (*state & 0x7FFF) * base >> 15;
}
-static int estimate_sid_gain(G723_1_Context *p)
+static int estimate_sid_gain(G723_1_ChannelContext *p)
{
int i, shift, seg, seg2, t, val, val_add, x, y;
if (p->sid_gain < 0) t = INT32_MIN;
else t = INT32_MAX;
} else
- t = p->sid_gain << shift;
+ t = p->sid_gain * (1 << shift);
+ } else if(shift < -31) {
+ t = (p->sid_gain < 0) ? -1 : 0;
}else
t = p->sid_gain >> -shift;
x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
return val;
}
-static void generate_noise(G723_1_Context *p)
+static void generate_noise(G723_1_ChannelContext *p)
{
int i, j, idx, t;
int off[SUBFRAMES];
static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
- G723_1_Context *p = avctx->priv_data;
+ G723_1_Context *s = avctx->priv_data;
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int16_t acb_vector[SUBFRAME_LEN];
int16_t *out;
int bad_frame = 0, i, j, ret;
- int16_t *audio = p->audio;
- if (buf_size < frame_size[dec_mode]) {
+ if (buf_size < frame_size[dec_mode] * avctx->channels) {
if (buf_size)
av_log(avctx, AV_LOG_WARNING,
"Expected %d bytes, got %d - skipping packet\n",
return buf_size;
}
- if (unpack_bitstream(p, buf, buf_size) < 0) {
- bad_frame = 1;
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_frame_type = ACTIVE_FRAME;
- else
- p->cur_frame_type = UNTRANSMITTED_FRAME;
- }
-
frame->nb_samples = FRAME_LEN;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- out = (int16_t *)frame->data[0];
-
- if (p->cur_frame_type == ACTIVE_FRAME) {
- if (!bad_frame)
- p->erased_frames = 0;
- else if (p->erased_frames != 3)
- p->erased_frames++;
-
- ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
- ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
-
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
-
- /* Generate the excitation for the frame */
- memcpy(p->excitation, p->prev_excitation,
- PITCH_MAX * sizeof(*p->excitation));
- if (!p->erased_frames) {
- int16_t *vector_ptr = p->excitation + PITCH_MAX;
-
- /* Update interpolation gain memory */
- p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
- p->subframe[3].amp_index) >> 1];
- for (i = 0; i < SUBFRAMES; i++) {
- gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
- p->pitch_lag[i >> 1], i);
- ff_g723_1_gen_acb_excitation(acb_vector,
- &p->excitation[SUBFRAME_LEN * i],
- p->pitch_lag[i >> 1],
- &p->subframe[i], p->cur_rate);
- /* Get the total excitation */
- for (j = 0; j < SUBFRAME_LEN; j++) {
- int v = av_clip_int16(vector_ptr[j] * 2);
- vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
- }
- vector_ptr += SUBFRAME_LEN;
- }
+ for (int ch = 0; ch < avctx->channels; ch++) {
+ G723_1_ChannelContext *p = &s->ch[ch];
+ int16_t *audio = p->audio;
+
+ if (unpack_bitstream(p, buf + ch * (buf_size / avctx->channels),
+ buf_size / avctx->channels) < 0) {
+ bad_frame = 1;
+ if (p->past_frame_type == ACTIVE_FRAME)
+ p->cur_frame_type = ACTIVE_FRAME;
+ else
+ p->cur_frame_type = UNTRANSMITTED_FRAME;
+ }
- vector_ptr = p->excitation + PITCH_MAX;
-
- p->interp_index = comp_interp_index(p, p->pitch_lag[1],
- &p->sid_gain, &p->cur_gain);
-
- /* Perform pitch postfiltering */
- if (p->postfilter) {
- i = PITCH_MAX;
- for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
- ppf + j, p->cur_rate);
-
- for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
- vector_ptr + i,
- vector_ptr + i + ppf[j].index,
- ppf[j].sc_gain,
- ppf[j].opt_gain,
- 1 << 14, 15, SUBFRAME_LEN);
- } else {
- audio = vector_ptr - LPC_ORDER;
- }
+ out = (int16_t *)frame->extended_data[ch];
+
+ if (p->cur_frame_type == ACTIVE_FRAME) {
+ if (!bad_frame)
+ p->erased_frames = 0;
+ else if (p->erased_frames != 3)
+ p->erased_frames++;
+
+ ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
+ ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
+
+ /* Save the lsp_vector for the next frame */
+ memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
+ /* Generate the excitation for the frame */
+ memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
- } else {
- p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
- if (p->erased_frames == 3) {
- /* Mute output */
- memset(p->excitation, 0,
- (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
- memset(p->prev_excitation, 0,
- PITCH_MAX * sizeof(*p->excitation));
- memset(frame->data[0], 0,
- (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
- } else {
- int16_t *buf = p->audio + LPC_ORDER;
+ if (!p->erased_frames) {
+ int16_t *vector_ptr = p->excitation + PITCH_MAX;
+
+ /* Update interpolation gain memory */
+ p->interp_gain = ff_g723_1_fixed_cb_gain[(p->subframe[2].amp_index +
+ p->subframe[3].amp_index) >> 1];
+ for (i = 0; i < SUBFRAMES; i++) {
+ gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
+ p->pitch_lag[i >> 1], i);
+ ff_g723_1_gen_acb_excitation(acb_vector,
+ &p->excitation[SUBFRAME_LEN * i],
+ p->pitch_lag[i >> 1],
+ &p->subframe[i], p->cur_rate);
+ /* Get the total excitation */
+ for (j = 0; j < SUBFRAME_LEN; j++) {
+ int v = av_clip_int16(vector_ptr[j] * 2);
+ vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
+ }
+ vector_ptr += SUBFRAME_LEN;
+ }
- /* Regenerate frame */
- residual_interp(p->excitation, buf, p->interp_index,
- p->interp_gain, &p->random_seed);
+ vector_ptr = p->excitation + PITCH_MAX;
+
+ p->interp_index = comp_interp_index(p, p->pitch_lag[1],
+ &p->sid_gain, &p->cur_gain);
+
+ /* Perform pitch postfiltering */
+ if (s->postfilter) {
+ i = PITCH_MAX;
+ for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
+ ppf + j, p->cur_rate);
+
+ for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
+ vector_ptr + i,
+ vector_ptr + i + ppf[j].index,
+ ppf[j].sc_gain,
+ ppf[j].opt_gain,
+ 1 << 14, 15, SUBFRAME_LEN);
+ } else {
+ audio = vector_ptr - LPC_ORDER;
+ }
/* Save the excitation for the next frame */
- memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
+ memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
PITCH_MAX * sizeof(*p->excitation));
+ } else {
+ p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
+ if (p->erased_frames == 3) {
+ /* Mute output */
+ memset(p->excitation, 0,
+ (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
+ memset(p->prev_excitation, 0,
+ PITCH_MAX * sizeof(*p->excitation));
+ memset(frame->data[0], 0,
+ (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
+ } else {
+ int16_t *buf = p->audio + LPC_ORDER;
+
+ /* Regenerate frame */
+ residual_interp(p->excitation, buf, p->interp_index,
+ p->interp_gain, &p->random_seed);
+
+ /* Save the excitation for the next frame */
+ memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
+ PITCH_MAX * sizeof(*p->excitation));
+ }
+ }
+ p->cng_random_seed = CNG_RANDOM_SEED;
+ } else {
+ if (p->cur_frame_type == SID_FRAME) {
+ p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
+ ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
+ } else if (p->past_frame_type == ACTIVE_FRAME) {
+ p->sid_gain = estimate_sid_gain(p);
}
- }
- p->cng_random_seed = CNG_RANDOM_SEED;
- } else {
- if (p->cur_frame_type == SID_FRAME) {
- p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
- ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
- } else if (p->past_frame_type == ACTIVE_FRAME) {
- p->sid_gain = estimate_sid_gain(p);
- }
- if (p->past_frame_type == ACTIVE_FRAME)
- p->cur_gain = p->sid_gain;
- else
- p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
- generate_noise(p);
- ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
- /* Save the lsp_vector for the next frame */
- memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
- }
+ if (p->past_frame_type == ACTIVE_FRAME)
+ p->cur_gain = p->sid_gain;
+ else
+ p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
+ generate_noise(p);
+ ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
+ /* Save the lsp_vector for the next frame */
+ memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
+ }
- p->past_frame_type = p->cur_frame_type;
+ p->past_frame_type = p->cur_frame_type;
- memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
- for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
- ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
- audio + i, SUBFRAME_LEN, LPC_ORDER,
- 0, 1, 1 << 12);
- memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
+ memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
+ for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
+ audio + i, SUBFRAME_LEN, LPC_ORDER,
+ 0, 1, 1 << 12);
+ memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
- if (p->postfilter) {
- formant_postfilter(p, lpc, p->audio, out);
- } else { // if output is not postfiltered it should be scaled by 2
- for (i = 0; i < FRAME_LEN; i++)
- out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
+ if (s->postfilter) {
+ formant_postfilter(p, lpc, p->audio, out);
+ } else { // if output is not postfiltered it should be scaled by 2
+ for (i = 0; i < FRAME_LEN; i++)
+ out[i] = av_clip_int16(2 * p->audio[LPC_ORDER + i]);
+ }
}
*got_frame_ptr = 1;
- return frame_size[dec_mode];
+ return frame_size[dec_mode] * avctx->channels;
}
#define OFFSET(x) offsetof(G723_1_Context, x)
.version = LIBAVUTIL_VERSION_INT,
};
-AVCodec ff_g723_1_decoder = {
+const AVCodec ff_g723_1_decoder = {
.name = "g723_1",
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.type = AVMEDIA_TYPE_AUDIO,
projects / ffmpeg / blobdiff