comparison dca.c @ 11908:9b1095b2616a libavcodec

Support DTS-ES extension (XCh) in dca: move subband_samples into context structure Patch by Nick Brereton, nick at nbrereton dot net

11907:defe337cf643

#define DCA_PRIM_CHANNELS_MAX (5)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32)      /* Should be 28 */
#define DCA_SUBSUBFRAMES_MAX (4)
#define DCA_LFE_MAX (3)

enum DCAMode {
    DCA_MONO = 0,
    DCA_CHANNEL,

    int output;                 ///< type of output
    float add_bias;             ///< output bias
    float scale_bias;           ///< output scale

    DECLARE_ALIGNED(16, float, samples)[1536];  /* 6 * 256 = 1536, might only need 5 */
    const float *samples_chanptr[6];

    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
    int dca_buffer_size;        ///< how much data is in the dca_buffer

}

static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

static int dca_subsubframe(DCAContext * s)
{
    int k, l;
    int subsubframe = s->current_subsubframe;

    const float *quant_step_table;

    /* FIXME */
    LOCAL_ALIGNED_16(float, subband_samples, [DCA_PRIM_CHANNELS_MAX], [DCA_SUBBANDS][8]);
    LOCAL_ALIGNED_16(int, block, [8]);

    /*
     * Audio data
     */

 * Decode a dca frame block
 *
 * @param s     pointer to the DCAContext
 */

static int dca_decode_block(DCAContext * s)
{

    /* Sanity check */
    if (s->current_subframe >= s->subframes) {
        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

    /* Read subsubframe */
#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
    if (dca_subsubframe(s))
        return -1;

    /* Update state */
    s->current_subsubframe++;
    if (s->current_subsubframe >= s->subsubframes) {

    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
        return -1;
    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
    for (i = 0; i < (s->sample_blocks / 8); i++) {
        dca_decode_block(s);
        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
        samples += 256 * channels;
    }

    return buf_size;

11908:9b1095b2616a

#define DCA_PRIM_CHANNELS_MAX (5)
#define DCA_SUBBANDS (32)
#define DCA_ABITS_MAX (32)      /* Should be 28 */
#define DCA_SUBSUBFRAMES_MAX (4)
#define DCA_BLOCKS_MAX (16)
#define DCA_LFE_MAX (3)

enum DCAMode {
    DCA_MONO = 0,
    DCA_CHANNEL,

    int output;                 ///< type of output
    float add_bias;             ///< output bias
    float scale_bias;           ///< output scale

    DECLARE_ALIGNED(16, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
    DECLARE_ALIGNED(16, float, samples)[1536];  /* 6 * 256 = 1536, might only need 5 */
    const float *samples_chanptr[6];

    uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
    int dca_buffer_size;        ///< how much data is in the dca_buffer

}

static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };

static int dca_subsubframe(DCAContext * s, int block_index)
{
    int k, l;
    int subsubframe = s->current_subsubframe;

    const float *quant_step_table;

    /* FIXME */
    float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index];
    LOCAL_ALIGNED_16(int, block, [8]);

    /*
     * Audio data
     */

 * Decode a dca frame block
 *
 * @param s     pointer to the DCAContext
 */

static int dca_decode_block(DCAContext * s, int block_index)
{

    /* Sanity check */
    if (s->current_subframe >= s->subframes) {
        av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",

    /* Read subsubframe */
#ifdef TRACE
    av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
    if (dca_subsubframe(s, block_index))
        return -1;

    /* Update state */
    s->current_subsubframe++;
    if (s->current_subsubframe >= s->subsubframes) {

    if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
        return -1;
    *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
    for (i = 0; i < (s->sample_blocks / 8); i++) {
        dca_decode_block(s, i);
        s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
        samples += 256 * channels;
    }

    return buf_size;