view g729dec.c @ 11211:dfeaae916502 libavcodec

Since WavPack chunk can contain more samples than FFmpeg is guaranteed to hold, decode it in several iterations outputting as many samples as possible.
author kostya
date Fri, 19 Feb 2010 14:05:41 +0000
parents 62705926ba33
children 25e209f9153a
line wrap: on
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/*
 * G.729 decoder
 * 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 <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 "get_bits.h"

#include "g729.h"
#include "lsp.h"
#include "celp_math.h"
#include "acelp_filters.h"
#include "acelp_pitch_delay.h"
#include "acelp_vectors.h"
#include "g729data.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

/**
 * 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

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
} G729FormatDescription;

typedef struct {
    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
}  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,
};

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,
};

/**
 * \brief pseudo random number generator
 */
static inline uint16_t g729_prng(uint16_t value)
{
    return 31821 * value + 13849;
}

/**
 * Get parity bit of bit 2..7
 */
static inline int get_parity(uint8_t value)
{
   return (0x6996966996696996ULL >> (value >> 2)) & 1;
}

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;
    }

    /* Rotate past_quantizer_outputs. */
    memmove(past_quantizer_outputs + 1, past_quantizer_outputs, MA_NP * sizeof(int16_t*));
    past_quantizer_outputs[0] = quantizer_output;

    ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
}

static av_cold int decoder_init(AVCodecContext * avctx)
{
    G729Context* ctx = avctx->priv_data;
    int i,k;

    if (avctx->channels != 1) {
        av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
        return AVERROR_NOFMT;
    }

    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
    avctx->frame_size = SUBFRAME_SIZE << 1;

    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));

    return 0;
}

static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
                        AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int16_t *out_frame = data;
    GetBitContext gb;
    G729FormatDescription format;
    int frame_erasure = 0;    ///< frame erasure detected during decoding
    int bad_pitch = 0;        ///< parity check failed
    int i;
    G729Context *ctx = avctx->priv_data;
    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;         // pitch delay, integer part
    int pitch_delay_3x;          // pitch delay, multiplied by 3

    if (*data_size < SUBFRAME_SIZE << 2) {
        av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
        return AVERROR(EIO);
    }

    if (buf_size == 10) {
        format = format_g729_8k;
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
    } else if (buf_size == 8) {
        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_NOFMT);
    }

    for (i=0; i < buf_size; i++)
        frame_erasure |= buf[i];
    frame_erasure = !frame_erasure;

    init_get_bits(&gb, buf, buf_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);

    lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
               ma_predictor,
               quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);

    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++) {
        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 = 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);

        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  = (pitch_delay_3x + 1) / 3;

        ff_acelp_weighted_vector_sum(fc + pitch_delay_int,
                                     fc + pitch_delay_int,
                                     fc, 1 << 14,
                                     av_clip(ctx->gain_pitch, SHARP_MIN, SHARP_MAX),
                                     0, 14,
                                     SUBFRAME_SIZE - pitch_delay_int);

        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)

            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];

        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,
                                     1 << 13, 14, SUBFRAME_SIZE);

            ctx->pitch_delay_int_prev = pitch_delay_int;
    }

    *data_size = SUBFRAME_SIZE << 2;
    return buf_size;
}

AVCodec g729_decoder =
{
    "g729",
    CODEC_TYPE_AUDIO,
    CODEC_ID_G729,
    sizeof(G729Context),
    decoder_init,
    NULL,
    NULL,
    decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("G.729"),
};