comparison qdm2.c @ 6273:39cd8d3d5b9e libavcodec

const

6272:ec8c482d17c1

typedef struct {
    float level;
    float *samples_im;
    float *samples_re;
    float *table;
    int   phase;
    int   phase_shift;
    int   duration;
    short time_index;
    short cutoff;

    FFTContext fft_ctx;
    FFTComplex exptab[128];
    QDM2FFT fft;

    /// I/O data
    uint8_t *compressed_data;
    int compressed_size;
    float output_buffer[1024];

    /// Synthesis filter
    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);

 * @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++)
        value -= data[i];

                        tone.cutoff = (offset >= 60) ? 3 : 2;

                    tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
                    tone.samples_im = &q->fft.samples_im[ch][offset];
                    tone.samples_re = &q->fft.samples_re[ch][offset];
                    tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
                    tone.duration = i;
                    tone.time_index = 0;

    return 0;
}


static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
{
    int ch, i;
    const int frame_size = (q->frame_size * q->channels);

    /* select input buffer */

}


static int qdm2_decode_frame(AVCodecContext *avctx,
            void *data, int *data_size,
            uint8_t *buf, int buf_size)
{
    QDM2Context *s = avctx->priv_data;

    if(!buf)
        return 0;

6273:39cd8d3d5b9e

typedef struct {
    float level;
    float *samples_im;
    float *samples_re;
    const float *table;
    int   phase;
    int   phase_shift;
    int   duration;
    short time_index;
    short cutoff;

    FFTContext fft_ctx;
    FFTComplex exptab[128];
    QDM2FFT fft;

    /// I/O data
    const uint8_t *compressed_data;
    int compressed_size;
    float output_buffer[1024];

    /// Synthesis filter
    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);

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

    for (i=0; i < length; i++)
        value -= data[i];

                        tone.cutoff = (offset >= 60) ? 3 : 2;

                    tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
                    tone.samples_im = &q->fft.samples_im[ch][offset];
                    tone.samples_re = &q->fft.samples_re[ch][offset];
                    tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
                    tone.duration = i;
                    tone.time_index = 0;

    return 0;
}


static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
{
    int ch, i;
    const int frame_size = (q->frame_size * q->channels);

    /* select input buffer */

}


static int qdm2_decode_frame(AVCodecContext *avctx,
            void *data, int *data_size,
            const uint8_t *buf, int buf_size)
{
    QDM2Context *s = avctx->priv_data;

    if(!buf)
        return 0;