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FFmpeg/libavcodec/sipr.c
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00

571 lines
18 KiB
C
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/*
* SIPR / ACELP.NET decoder
*
* Copyright (c) 2008 Vladimir Voroshilov
* Copyright (c) 2009 Vitor Sessak
*
* 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 <math.h>
#include <stdint.h>
#include <string.h>
#include "libavutil/mathematics.h"
#include "avcodec.h"
#define ALT_BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_math.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "acelp_filters.h"
#include "celp_filters.h"
#define MAX_SUBFRAME_COUNT 5
#include "sipr.h"
#include "siprdata.h"
typedef struct {
const char *mode_name;
uint16_t bits_per_frame;
uint8_t subframe_count;
uint8_t frames_per_packet;
float pitch_sharp_factor;
/* bitstream parameters */
uint8_t number_of_fc_indexes;
uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
/** size in bits of the i-th stage vector of quantizer */
uint8_t vq_indexes_bits[5];
/** size in bits of the adaptive-codebook index for every subframe */
uint8_t pitch_delay_bits[5];
uint8_t gp_index_bits;
uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
} SiprModeParam;
static const SiprModeParam modes[MODE_COUNT] = {
[MODE_16k] = {
.mode_name = "16k",
.bits_per_frame = 160,
.subframe_count = SUBFRAME_COUNT_16k,
.frames_per_packet = 1,
.pitch_sharp_factor = 0.00,
.number_of_fc_indexes = 10,
.ma_predictor_bits = 1,
.vq_indexes_bits = {7, 8, 7, 7, 7},
.pitch_delay_bits = {9, 6},
.gp_index_bits = 4,
.fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
.gc_index_bits = 5
},
[MODE_8k5] = {
.mode_name = "8k5",
.bits_per_frame = 152,
.subframe_count = 3,
.frames_per_packet = 1,
.pitch_sharp_factor = 0.8,
.number_of_fc_indexes = 3,
.ma_predictor_bits = 0,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {9, 9, 9},
.gc_index_bits = 7
},
[MODE_6k5] = {
.mode_name = "6k5",
.bits_per_frame = 232,
.subframe_count = 3,
.frames_per_packet = 2,
.pitch_sharp_factor = 0.8,
.number_of_fc_indexes = 3,
.ma_predictor_bits = 0,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {5, 5, 5},
.gc_index_bits = 7
},
[MODE_5k0] = {
.mode_name = "5k0",
.bits_per_frame = 296,
.subframe_count = 5,
.frames_per_packet = 2,
.pitch_sharp_factor = 0.85,
.number_of_fc_indexes = 1,
.ma_predictor_bits = 0,
.vq_indexes_bits = {6, 7, 7, 7, 5},
.pitch_delay_bits = {8, 5, 8, 5, 5},
.gp_index_bits = 0,
.fc_index_bits = {10},
.gc_index_bits = 7
}
};
const float ff_pow_0_5[] = {
1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
};
static void dequant(float *out, const int *idx, const float *cbs[])
{
int i;
int stride = 2;
int num_vec = 5;
for (i = 0; i < num_vec; i++)
memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
}
static void lsf_decode_fp(float *lsfnew, float *lsf_history,
const SiprParameters *parm)
{
int i;
float lsf_tmp[LP_FILTER_ORDER];
dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
for (i = 0; i < LP_FILTER_ORDER; i++)
lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
/* Note that a minimum distance is not enforced between the last value and
the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
for (i = 0; i < LP_FILTER_ORDER - 1; i++)
lsfnew[i] = cos(lsfnew[i]);
lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
}
/** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
static void pitch_sharpening(int pitch_lag_int, float beta,
float *fixed_vector)
{
int i;
for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
}
/**
* Extract decoding parameters from the input bitstream.
* @param parms parameters structure
* @param pgb pointer to initialized GetBitContext structure
*/
static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
const SiprModeParam *p)
{
int i, j;
if (p->ma_predictor_bits)
parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
for (i = 0; i < 5; i++)
parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
for (i = 0; i < p->subframe_count; i++) {
parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
if (p->gp_index_bits)
parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
for (j = 0; j < p->number_of_fc_indexes; j++)
parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
}
}
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
int num_subfr)
{
double lsfint[LP_FILTER_ORDER];
int i,j;
float t, t0 = 1.0 / num_subfr;
t = t0 * 0.5;
for (i = 0; i < num_subfr; i++) {
for (j = 0; j < LP_FILTER_ORDER; j++)
lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
Az += LP_FILTER_ORDER;
t += t0;
}
}
/**
* Evaluate the adaptive impulse response.
*/
static void eval_ir(const float *Az, int pitch_lag, float *freq,
float pitch_sharp_factor)
{
float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
int i;
tmp1[0] = 1.;
for (i = 0; i < LP_FILTER_ORDER; i++) {
tmp1[i+1] = Az[i] * ff_pow_0_55[i];
tmp2[i ] = Az[i] * ff_pow_0_7 [i];
}
memset(tmp1 + 11, 0, 37 * sizeof(float));
ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
LP_FILTER_ORDER);
pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
}
/**
* Evaluate the convolution of a vector with a sparse vector.
*/
static void convolute_with_sparse(float *out, const AMRFixed *pulses,
const float *shape, int length)
{
int i, j;
memset(out, 0, length*sizeof(float));
for (i = 0; i < pulses->n; i++)
for (j = pulses->x[i]; j < length; j++)
out[j] += pulses->y[i] * shape[j - pulses->x[i]];
}
/**
* Apply postfilter, very similar to AMR one.
*/
static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
{
float buf[SUBFR_SIZE + LP_FILTER_ORDER];
float *pole_out = buf + LP_FILTER_ORDER;
float lpc_n[LP_FILTER_ORDER];
float lpc_d[LP_FILTER_ORDER];
int i;
for (i = 0; i < LP_FILTER_ORDER; i++) {
lpc_d[i] = lpc[i] * ff_pow_0_75[i];
lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
};
memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
LP_FILTER_ORDER*sizeof(float));
ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
LP_FILTER_ORDER);
memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
LP_FILTER_ORDER*sizeof(float));
ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
LP_FILTER_ORDER*sizeof(*pole_out));
memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
LP_FILTER_ORDER*sizeof(*pole_out));
ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
LP_FILTER_ORDER);
}
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
SiprMode mode, int low_gain)
{
int i;
switch (mode) {
case MODE_6k5:
for (i = 0; i < 3; i++) {
fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
}
fixed_sparse->n = 3;
break;
case MODE_8k5:
for (i = 0; i < 3; i++) {
fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
fixed_sparse->y[2*i + 1] =
(fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
-fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
}
fixed_sparse->n = 6;
break;
case MODE_5k0:
default:
if (low_gain) {
int offset = (pulses[0] & 0x200) ? 2 : 0;
int val = pulses[0];
for (i = 0; i < 3; i++) {
int index = (val & 0x7) * 6 + 4 - i*2;
fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
fixed_sparse->x[i] = index;
val >>= 3;
}
fixed_sparse->n = 3;
} else {
int pulse_subset = (pulses[0] >> 8) & 1;
fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
fixed_sparse->y[1] = -fixed_sparse->y[0];
fixed_sparse->n = 2;
}
break;
}
}
static void decode_frame(SiprContext *ctx, SiprParameters *params,
float *out_data)
{
int i, j;
int subframe_count = modes[ctx->mode].subframe_count;
int frame_size = subframe_count * SUBFR_SIZE;
float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT];
float *excitation;
float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
float lsf_new[LP_FILTER_ORDER];
float *impulse_response = ir_buf + LP_FILTER_ORDER;
float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
// memory alignment
int t0_first = 0;
AMRFixed fixed_cb;
memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
lsf_decode_fp(lsf_new, ctx->lsf_history, params);
sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
for (i = 0; i < subframe_count; i++) {
float *pAz = Az + i*LP_FILTER_ORDER;
float fixed_vector[SUBFR_SIZE];
int T0,T0_frac;
float pitch_gain, gain_code, avg_energy;
ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
ctx->mode == MODE_5k0, 6);
if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
t0_first = T0;
ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
ff_b60_sinc, 6,
2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
SUBFR_SIZE);
decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
ctx->past_pitch_gain < 0.8);
eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
SUBFR_SIZE);
avg_energy =
(0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
SUBFR_SIZE;
ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
avg_energy, ctx->energy_history,
34 - 15.0/(0.05*M_LN10/M_LN2),
pred);
ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
pitch_gain, gain_code, SUBFR_SIZE);
pitch_gain *= 0.5 * pitch_gain;
pitch_gain = FFMIN(pitch_gain, 0.4);
ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
gain_code *= ctx->gain_mem;
for (j = 0; j < SUBFR_SIZE; j++)
fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
if (ctx->mode == MODE_5k0) {
postfilter_5k0(ctx, pAz, fixed_vector);
ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
pAz, excitation, SUBFR_SIZE,
LP_FILTER_ORDER);
}
ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
SUBFR_SIZE, LP_FILTER_ORDER);
excitation += SUBFR_SIZE;
}
memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
LP_FILTER_ORDER * sizeof(float));
if (ctx->mode == MODE_5k0) {
for (i = 0; i < subframe_count; i++) {
float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
SUBFR_SIZE);
ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
&synth[i * SUBFR_SIZE], energy,
SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
}
memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
LP_FILTER_ORDER*sizeof(float));
}
memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
(PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
ff_acelp_apply_order_2_transfer_function(out_data, synth,
(const float[2]) {-1.99997 , 1.000000000},
(const float[2]) {-1.93307352, 0.935891986},
0.939805806,
ctx->highpass_filt_mem,
frame_size);
}
static av_cold int sipr_decoder_init(AVCodecContext * avctx)
{
SiprContext *ctx = avctx->priv_data;
int i;
switch (avctx->block_align) {
case 20: ctx->mode = MODE_16k; break;
case 19: ctx->mode = MODE_8k5; break;
case 29: ctx->mode = MODE_6k5; break;
case 37: ctx->mode = MODE_5k0; break;
default:
av_log(avctx, AV_LOG_ERROR, "Invalid block_align: %d\n", avctx->block_align);
return AVERROR(EINVAL);
}
av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
if (ctx->mode == MODE_16k) {
ff_sipr_init_16k(ctx);
ctx->decode_frame = ff_sipr_decode_frame_16k;
} else {
ctx->decode_frame = decode_frame;
}
for (i = 0; i < LP_FILTER_ORDER; i++)
ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
for (i = 0; i < 4; i++)
ctx->energy_history[i] = -14;
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
avcodec_get_frame_defaults(&ctx->frame);
avctx->coded_frame = &ctx->frame;
return 0;
}
static int sipr_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
SiprContext *ctx = avctx->priv_data;
const uint8_t *buf=avpkt->data;
SiprParameters parm;
const SiprModeParam *mode_par = &modes[ctx->mode];
GetBitContext gb;
float *samples;
int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
int i, ret;
ctx->avctx = avctx;
if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
av_log(avctx, AV_LOG_ERROR,
"Error processing packet: packet size (%d) too small\n",
avpkt->size);
return -1;
}
/* get output buffer */
ctx->frame.nb_samples = mode_par->frames_per_packet * subframe_size *
mode_par->subframe_count;
if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (float *)ctx->frame.data[0];
init_get_bits(&gb, buf, mode_par->bits_per_frame);
for (i = 0; i < mode_par->frames_per_packet; i++) {
decode_parameters(&parm, &gb, mode_par);
ctx->decode_frame(ctx, &parm, samples);
samples += subframe_size * mode_par->subframe_count;
}
*got_frame_ptr = 1;
*(AVFrame *)data = ctx->frame;
return mode_par->bits_per_frame >> 3;
}
AVCodec ff_sipr_decoder = {
.name = "sipr",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_SIPR,
.priv_data_size = sizeof(SiprContext),
.init = sipr_decoder_init,
.decode = sipr_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
};
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