comparison qdm2.c @ 3043:583020ce54a8 libavcodec

Fix a bunch of spelling/grammar mistakes in doxygen comments and output.

3042:a434a3752e78

/**
 * @file qdm2.c
 * QDM2 decoder
 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
 * The decoder is not perfect yet, there are still some distorions expecially
 * on files encoded with 16 or 8 subbands
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

    unsigned int size;   ///< subpacket size
    const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
} QDM2SubPacket;

/**
 * A node in subpacket list
 */
typedef struct _QDM2SubPNode {
    QDM2SubPacket *packet;      ///< packet
    struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;

    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];

    // Flags
    int has_errors;         ///< packet have errors
    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
    int do_synth_filter;    ///< used to perform or skip synthesis filter

    int sub_packet;
    int noise_idx; ///< Index for dithering noise table
} QDM2Context;


static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];

 *
 * @param data      pointer to data to be checksum'ed
 * @param length    data length
 * @param value     checksum value
 *
 * @return          0 if checksum is ok
 */
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
    int i;

    for (i=0; i < length; i++)

    return (uint16_t)(value & 0xffff);
}


/**
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer
 *
 * @param gb            bitreader context
 * @param sub_packet    packet under analysis
 */
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)

          sub_packet->type |= (get_bits (gb, 8) << 8);

      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
    }

    av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}


/**
 * Return node pointer to first packet of requested type in list
 *
 * @param list    list of subpacket to be scanned
 * @param type    type of searched subpacket
 * @return        node pointer for subpacket if found, else NULL
 */
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
{

    return NULL;
}


/**
 * Replaces 8 elements with their average value
 * Called by qdm2_decode_superblock before starting subblocks decoding
 *
 * @param q       context
 */
static void average_quantized_coeffs (QDM2Context *q)
{

        }
}


/**
 * Build subband samples with noise weighted by q->tone_level
 * Called by synthfilt_build_sb_samples
 *
 * @param q     context
 * @param sb    subband index
 */
static void build_sb_samples_from_noise (QDM2Context *q, int sb)

        }
}


/**
 * Called while processing data from subpackets 11 and 12
 * Used after making changes to coding_method array
 *
 * @param sb               subband index
 * @param channels         number of channels
 * @param coding_method    q->coding_method[0][0][0]
 */

 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bit
 * @param sb_min    lower subband processed (sb_min included)
 * @param sb_max    higher subband processed (sb_max excluded)
 */
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
{

    } // subband loop
}


/**
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
 * same VLC tables as process_subpacket_9 are used
 *
 * @param q         context
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
 * @param gb        bitreader context
 * @param length    packet length in bit
 */
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
{
    int i, k, run, level, diff;

 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bit
 */
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
{
    int sb, j, k, n, ch;

/**
 * Process subpacket 10 if not null, else
 *
 * @param q         context
 * @param node      pointer to node with packet
 * @param length    packet length in bit
 */
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
{
    GetBitContext gb;

/**
 * Process subpacket 12
 *
 * @param q         context
 * @param node      pointer to node with packet
 * @param length    packet length in bit
 */
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
{
    GetBitContext gb;

        process_subpacket_12(q, NULL, 0);
}


/*
 * Decode superblock, fill packet lists
 *
 * @param q    context
 */
static void qdm2_decode_super_block (QDM2Context *q)
{

            if (next_index >= header.size)
                break;
        }

        /* decode sub packet */
        packet = &q->sub_packets[i];
        qdm2_decode_sub_packet_header(&gb, packet);
        next_index = packet->size + get_bits_count(&gb) / 8;
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;

            packet->size += packet_bytes - sub_packet_size;
        }

        packet_bytes -= sub_packet_size;

        /* add sub packet to 'all sub packets' list */
        q->sub_packet_list_A[i].packet = packet;

        /* add sub packet to related list */
        if (packet->type == 8) {
            SAMPLES_NEEDED_2("packet type 8");
            return;
        } else if (packet->type >= 9 && packet->type <= 12) {
            /* packets for MPEG Audio like Synthesis Filter */

    /* reset minimum indices for FFT coefficients */
    q->fft_coefs_index = 0;
    for (i=0; i < 5; i++)
        q->fft_coefs_min_index[i] = -1;

    /* process sub packets ordered by type, largest type first */
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
        QDM2SubPacket *packet;

        /* find sub packet with largest type less than max */
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
            value = q->sub_packet_list_B[j].packet->type;
            if (value > min && value < max) {
                min = value;
                packet = q->sub_packet_list_B[j].packet;

    const int n2 = n >> 1;
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
    float c, s, f0, f1, f2, f3;
    int i, j;

    /* pre rotation (or something like that) */
    for (i=1; i < n2; i++) {
        j  = (n - i);
        c = q->exptab[i].re;
        s = -q->exptab[i].im;
        f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;

    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));

    /* decode block of QDM2 compressed data */
    if (q->sub_packet == 0) {
        q->has_errors = 0; // zero it for a new super block
        av_log(NULL,AV_LOG_DEBUG,"Super block follows\n");
        qdm2_decode_super_block(q);
    }

    /* parse sub packets */
    if (!q->has_errors) {
        if (q->sub_packet == 2)
            qdm2_decode_fft_packets(q);

        qdm2_fft_tone_synthesizer(q, q->sub_packet);

3043:583020ce54a8

/**
 * @file qdm2.c
 * QDM2 decoder
 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
 * The decoder is not perfect yet, there are still some distortions
 * especially on files encoded with 16 or 8 subbands.
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

    unsigned int size;   ///< subpacket size
    const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
} QDM2SubPacket;

/**
 * A node in the subpacket list
 */
typedef struct _QDM2SubPNode {
    QDM2SubPacket *packet;      ///< packet
    struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;

    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];

    // Flags
    int has_errors;         ///< packet has errors
    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
    int do_synth_filter;    ///< used to perform or skip synthesis filter

    int sub_packet;
    int noise_idx; ///< index for dithering noise table
} QDM2Context;


static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];

 *
 * @param data      pointer to data to be checksum'ed
 * @param length    data length
 * @param value     checksum value
 *
 * @return          0 if checksum is OK
 */
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
    int i;

    for (i=0; i < length; i++)

    return (uint16_t)(value & 0xffff);
}


/**
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
 *
 * @param gb            bitreader context
 * @param sub_packet    packet under analysis
 */
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)

          sub_packet->type |= (get_bits (gb, 8) << 8);

      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
    }

    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}


/**
 * Return node pointer to first packet of requested type in list.
 *
 * @param list    list of subpackets to be scanned
 * @param type    type of searched subpacket
 * @return        node pointer for subpacket if found, else NULL
 */
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
{

    return NULL;
}


/**
 * Replaces 8 elements with their average value.
 * Called by qdm2_decode_superblock before starting subblock decoding.
 *
 * @param q       context
 */
static void average_quantized_coeffs (QDM2Context *q)
{

        }
}


/**
 * Build subband samples with noise weighted by q->tone_level.
 * Called by synthfilt_build_sb_samples.
 *
 * @param q     context
 * @param sb    subband index
 */
static void build_sb_samples_from_noise (QDM2Context *q, int sb)

        }
}


/**
 * Called while processing data from subpackets 11 and 12.
 * Used after making changes to coding_method array.
 *
 * @param sb               subband index
 * @param channels         number of channels
 * @param coding_method    q->coding_method[0][0][0]
 */

 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bits
 * @param sb_min    lower subband processed (sb_min included)
 * @param sb_max    higher subband processed (sb_max excluded)
 */
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
{

    } // subband loop
}


/**
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
 * same VLC tables as process_subpacket_9 are used.
 *
 * @param q         context
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
 * @param gb        bitreader context
 * @param length    packet length in bits
 */
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
{
    int i, k, run, level, diff;

 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
 *
 * @param q         context
 * @param gb        bitreader context
 * @param length    packet length in bits
 */
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
{
    int sb, j, k, n, ch;

/**
 * Process subpacket 10 if not null, else
 *
 * @param q         context
 * @param node      pointer to node with packet
 * @param length    packet length in bits
 */
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
{
    GetBitContext gb;

/**
 * Process subpacket 12
 *
 * @param q         context
 * @param node      pointer to node with packet
 * @param length    packet length in bits
 */
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
{
    GetBitContext gb;

        process_subpacket_12(q, NULL, 0);
}


/*
 * Decode superblock, fill packet lists.
 *
 * @param q    context
 */
static void qdm2_decode_super_block (QDM2Context *q)
{

            if (next_index >= header.size)
                break;
        }

        /* decode subpacket */
        packet = &q->sub_packets[i];
        qdm2_decode_sub_packet_header(&gb, packet);
        next_index = packet->size + get_bits_count(&gb) / 8;
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;

            packet->size += packet_bytes - sub_packet_size;
        }

        packet_bytes -= sub_packet_size;

        /* add subpacket to 'all subpackets' list */
        q->sub_packet_list_A[i].packet = packet;

        /* add subpacket to related list */
        if (packet->type == 8) {
            SAMPLES_NEEDED_2("packet type 8");
            return;
        } else if (packet->type >= 9 && packet->type <= 12) {
            /* packets for MPEG Audio like Synthesis Filter */

    /* reset minimum indices for FFT coefficients */
    q->fft_coefs_index = 0;
    for (i=0; i < 5; i++)
        q->fft_coefs_min_index[i] = -1;

    /* process subpackets ordered by type, largest type first */
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
        QDM2SubPacket *packet;

        /* find subpacket with largest type less than max */
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
            value = q->sub_packet_list_B[j].packet->type;
            if (value > min && value < max) {
                min = value;
                packet = q->sub_packet_list_B[j].packet;

    const int n2 = n >> 1;
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
    float c, s, f0, f1, f2, f3;
    int i, j;

    /* prerotation (or something like that) */
    for (i=1; i < n2; i++) {
        j  = (n - i);
        c = q->exptab[i].re;
        s = -q->exptab[i].im;
        f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;

    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));

    /* decode block of QDM2 compressed data */
    if (q->sub_packet == 0) {
        q->has_errors = 0; // zero it for a new super block
        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
        qdm2_decode_super_block(q);
    }

    /* parse subpackets */
    if (!q->has_errors) {
        if (q->sub_packet == 2)
            qdm2_decode_fft_packets(q);

        qdm2_fft_tone_synthesizer(q, q->sub_packet);