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dcadec: reorganise context data

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place primary audio coding header data into DCAAudioHeader
structure to make DCAContext clearer
and move channel related data to DCAChan structure to make
them easier to use by extensions

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
pull/154/head
Alexandra Khirnova 9 years ago committed by Luca Barbato
parent 3a4d369ea4
commit 58b42345b3
2 changed files with 187 additions and 159 deletions
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  1. 70
      libavcodec/dca.h
  2. 276
      libavcodec/dcadec.c

70
libavcodec/dca.h
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@ -130,6 +130,47 @@ typedef struct QMF64_table {
float rsin[32];
} QMF64_table;
/* Primary audio coding header */
typedef struct DCAAudioHeader {
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
int subframes; ///< number of subframes
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
} DCAAudioHeader;
typedef struct DCAChan {
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][8];
/* Subband samples history (for ADPCM) */
DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_SUBBANDS][4];
int hist_index;
/* Half size is sufficient for core decoding, but for 96 kHz data
* we need QMF with 64 subbands and 1024 samples. */
DECLARE_ALIGNED(32, float, subband_fir_hist)[1024];
DECLARE_ALIGNED(32, float, subband_fir_noidea)[64];
/* Primary audio coding side information */
int prediction_mode[DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_SUBBANDS]; ///< bit allocation index
int transition_mode[DCA_SUBBANDS]; ///< transition mode (transients)
int32_t scale_factor[DCA_SUBBANDS][2];///< scale factors (2 if transient)
int joint_huff; ///< joint subband scale factors codebook
int joint_scale_factor[DCA_SUBBANDS]; ///< joint subband scale factors
int32_t high_freq_vq[DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
} DCAChan;
typedef struct DCAContext {
AVClass *class; ///< class for AVOptions
AVCodecContext *avctx;
@ -163,28 +204,11 @@ typedef struct DCAContext {
int dialog_norm; ///< dialog normalisation parameter
/* Primary audio coding header */
int subframes; ///< number of subframes
int total_channels; ///< number of channels including extensions
int prim_channels; ///< number of primary audio channels
int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
DCAAudioHeader audio_header;
/* Primary audio coding side information */
int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
int32_t scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];///< scale factors (2 if transient)
int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///< stereo downmix coefficients
int dynrange_coef; ///< dynamic range coefficient
@ -195,23 +219,17 @@ typedef struct DCAContext {
uint8_t core_downmix_amode; ///< audio channel arrangement of embedded downmix
uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4]; ///< embedded downmix coefficients (9-bit codes)
int32_t high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
/* Half size is sufficient for core decoding, but for 96 kHz data
* we need QMF with 64 subbands and 1024 samples. */
DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][1024];
DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][64];
int hist_index[DCA_PRIM_CHANNELS_MAX];
DECLARE_ALIGNED(32, float, raXin)[32];
DCAChan dca_chan[DCA_PRIM_CHANNELS_MAX];
int output; ///< type of output
DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t *extra_channels_buffer;

276
libavcodec/dcadec.c
Unescape View File

@ -229,43 +229,47 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel)
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
s->total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
s->prim_channels = s->total_channels;
s->audio_header.total_channels = get_bits(&s->gb, 3) + 1 + base_channel;
s->audio_header.prim_channels = s->audio_header.total_channels;
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX;
if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
for (i = base_channel; i < s->prim_channels; i++) {
s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS;
for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
s->audio_header.subband_activity[i] = DCA_SUBBANDS;
}
for (i = base_channel; i < s->prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS)
s->vq_start_subband[i] = DCA_SUBBANDS;
for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
s->audio_header.vq_start_subband[i] = DCA_SUBBANDS;
}
get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3);
get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2);
get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3);
get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3);
get_array(&s->gb, s->audio_header.joint_intensity + base_channel,
s->audio_header.prim_channels - base_channel, 3);
get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 2);
get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 3);
get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
s->audio_header.prim_channels - base_channel, 3);
/* Get codebooks quantization indexes */
if (!base_channel)
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman));
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */
for (j = 0; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
s->scalefactor_adj[i][j] = 1;
for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->audio_header.scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
for (i = base_channel; i < s->audio_header.prim_channels; i++)
if (s->audio_header.quant_index_huffman[i][j] < thr[j])
s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (s->crc_present) {
/* Audio header CRC check */
@ -336,7 +340,7 @@ static int dca_parse_frame_header(DCAContext *s)
s->output |= DCA_LFE;
/* Primary audio coding header */
s->subframes = get_bits(&s->gb, 4) + 1;
s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
return dca_parse_audio_coding_header(s, 0);
}
@ -371,53 +375,53 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
}
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++)
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++)
s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
}
/* Get prediction codebook */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
if (s->prediction_mode[j][k] > 0) {
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (s->dca_chan[j].prediction_mode[k] > 0) {
/* (Prediction coefficient VQ address) */
s->prediction_vq[j][k] = get_bits(&s->gb, 12);
s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
}
}
}
/* Bit allocation index */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->vq_start_subband[j]; k++) {
if (s->bitalloc_huffman[j] == 6)
s->bitalloc[j][k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4);
else if (s->bitalloc_huffman[j] == 7) {
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
if (s->audio_header.bitalloc_huffman[j] == 6)
s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
else if (s->audio_header.bitalloc_huffman[j] == 5)
s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
else if (s->audio_header.bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
return AVERROR_INVALIDDATA;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
s->dca_chan[j].bitalloc[k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
}
if (s->bitalloc[j][k] > 26) {
if (s->dca_chan[j].bitalloc[k] > 26) {
ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
j, k, s->bitalloc[j][k]);
j, k, s->dca_chan[j].bitalloc[k]);
return AVERROR_INVALIDDATA;
}
}
}
/* Transition mode */
for (j = base_channel; j < s->prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) {
s->transition_mode[j][k] = 0;
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
s->dca_chan[j].transition_mode[k] = 0;
if (s->subsubframes[s->current_subframe] > 1 &&
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
s->transition_mode[j][k] =
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
s->dca_chan[j].transition_mode[k] =
get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
}
}
}
@ -425,14 +429,14 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (j = base_channel; j < s->prim_channels; j++) {
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
const uint32_t *scale_table;
int scale_sum, log_size;
memset(s->scale_factor[j], 0,
s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
memset(s->dca_chan[j].scale_factor, 0,
s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
if (s->scalefactor_huffman[j] == 6) {
if (s->audio_header.scalefactor_huffman[j] == 6) {
scale_table = ff_dca_scale_factor_quant7;
log_size = 7;
} else {
@ -443,45 +447,46 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
/* When huffman coded, only the difference is encoded */
scale_sum = 0;
for (k = 0; k < s->subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][0] = scale_table[scale_sum];
for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
}
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
/* Get second scale factor */
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][1] = scale_table[scale_sum];
scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
}
}
}
/* Joint subband scale factor codebook select */
for (j = base_channel; j < s->prim_channels; j++) {
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3);
if (s->audio_header.joint_intensity[j] > 0)
s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->prim_channels; j++) {
for (j = base_channel; j < s->audio_header.prim_channels; j++) {
int source_channel;
/* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0) {
if (s->audio_header.joint_intensity[j] > 0) {
int scale = 0;
source_channel = s->joint_intensity[j] - 1;
source_channel = s->audio_header.joint_intensity[j] - 1;
/* When huffman coded, only the difference is encoded
* (is this valid as well for joint scales ???) */
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7);
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
for (k = s->audio_header.subband_activity[j];
k < s->audio_header.subband_activity[source_channel]; k++) {
scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
s->dca_chan[j].joint_scale_factor[k] = scale; /*joint_scale_table[scale]; */
}
if (!(s->debug_flag & 0x02)) {
@ -506,10 +511,10 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
*/
/* VQ encoded high frequency subbands */
for (j = base_channel; j < s->prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
for (j = base_channel; j < s->audio_header.prim_channels; j++)
for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
/* 1 vector -> 32 samples */
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
s->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
/* Low frequency effect data */
if (!base_channel && s->lfe) {
@ -543,7 +548,7 @@ static void qmf_32_subbands(DCAContext *s, int chans,
{
const float *prCoeff;
int sb_act = s->subband_activity[chans];
int sb_act = s->audio_header.subband_activity[chans];
scale *= sqrt(1 / 8.0);
@ -554,9 +559,9 @@ static void qmf_32_subbands(DCAContext *s, int chans,
prCoeff = ff_dca_fir_32bands_perfect;
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
s->subband_fir_hist[chans],
&s->hist_index[chans],
s->subband_fir_noidea[chans], prCoeff,
s->dca_chan[chans].subband_fir_hist,
&s->dca_chan[chans].hist_index,
s->dca_chan[chans].subband_fir_noidea, prCoeff,
samples_out, s->raXin, scale);
}
@ -591,14 +596,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
{
float raXin[64];
float A[32], B[32];
float *raX = s->subband_fir_hist[chans];
float *raZ = s->subband_fir_noidea[chans];
float *raX = s->dca_chan[chans].subband_fir_hist;
float *raZ = s->dca_chan[chans].subband_fir_noidea;
unsigned i, j, k, subindex;
for (i = s->subband_activity[chans]; i < 64; i++)
for (i = s->audio_header.subband_activity[chans]; i < 64; i++)
raXin[i] = 0.0;
for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
for (i = 0; i < s->subband_activity[chans]; i++)
for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
raXin[i] = samples_in[i][subindex];
for (k = 0; k < 32; k++) {
@ -787,8 +792,6 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
const float *quant_step_table;
/* FIXME */
float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
/*
@ -801,17 +804,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
else
quant_step_table = ff_dca_lossy_quant_d;
for (k = base_channel; k < s->prim_channels; k++) {
for (k = base_channel; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
float rscale[DCA_SUBBANDS];
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (l = 0; l < s->vq_start_subband[k]; l++) {
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/* Select the mid-tread linear quantizer */
int abits = s->bitalloc[k][l];
int abits = s->dca_chan[k].bitalloc[l];
float quant_step_size = quant_step_table[abits];
@ -820,7 +824,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
*/
/* Select quantization index code book */
int sel = s->quant_index_huffman[k][abits];
int sel = s->audio_header.quant_index_huffman[k][abits];
/*
* Extract bits from the bit stream
@ -830,9 +834,10 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
} else {
/* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l];
rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] *
s->scalefactor_adj[k][sel];
int sfi = s->dca_chan[k].transition_mode[l] &&
subsubframe >= s->dca_chan[k].transition_mode[l];
rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] *
s->audio_header.scalefactor_adj[k][sel];
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
if (abits <= 7) {
@ -865,54 +870,61 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
}
}
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[k][0],
block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]);
s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0],
block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]);
for (l = 0; l < s->vq_start_subband[k]; l++) {
for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
int m;
/*
* Inverse ADPCM if in prediction mode
*/
if (s->prediction_mode[k][l]) {
if (s->dca_chan[k].prediction_mode[l]) {
int n;
if (s->predictor_history)
subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
s->subband_samples_hist[k][l][3] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] *
s->subband_samples_hist[k][l][2] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] *
s->subband_samples_hist[k][l][1] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] *
s->subband_samples_hist[k][l][0]) *
subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
s->dca_chan[k].subband_samples_hist[l][3] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
s->dca_chan[k].subband_samples_hist[l][2] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
s->dca_chan[k].subband_samples_hist[l][1] +
ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
s->dca_chan[k].subband_samples_hist[l][0]) *
(1.0f / 8192);
for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] *
subband_samples[k][l][m - 1];
float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
subband_samples[l][m - 1];
for (n = 2; n <= 4; n++)
if (m >= n)
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
subband_samples[k][l][m - n];
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
subband_samples[l][m - n];
else if (s->predictor_history)
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] *
s->subband_samples_hist[k][l][m - n + 4];
subband_samples[k][l][m] += sum * 1.0f / 8192;
sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
s->dca_chan[k].subband_samples_hist[l][m - n + 4];
subband_samples[l][m] += sum * 1.0f / 8192;
}
}
}
/* Backup predictor history for adpcm */
for (l = 0; l < DCA_SUBBANDS; l++)
AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
/*
* Decode VQ encoded high frequencies
*/
if (s->subband_activity[k] > s->vq_start_subband[k]) {
if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG,
"Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->scale_factor[k], s->vq_start_subband[k],
s->subband_activity[k]);
s->dca_chan[k].scale_factor,
s->audio_header.vq_start_subband[k],
s->audio_header.subband_activity[k]);
}
}
@ -924,17 +936,11 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
}
}
/* Backup predictor history for adpcm */
for (k = base_channel; k < s->prim_channels; k++)
for (l = 0; l < s->vq_start_subband[k]; l++)
AV_COPY128(s->subband_samples_hist[k][l], &subband_samples[k][l][4]);
return 0;
}
static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
{
float (*subband_samples)[DCA_SUBBANDS][SAMPLES_PER_SUBBAND] = s->subband_samples[block_index];
int k;
if (upsample) {
@ -945,18 +951,22 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
}
/* 64 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
for (k = 0; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
if (s->channel_order_tab[k] >= 0)
qmf_64_subbands(s, k, subband_samples[k],
qmf_64_subbands(s, k, subband_samples,
s->samples_chanptr[s->channel_order_tab[k]],
/* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0);
}
} else {
/* 32 subbands QMF */
for (k = 0; k < s->prim_channels; k++) {
for (k = 0; k < s->audio_header.prim_channels; k++) {
float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index];
if (s->channel_order_tab[k] >= 0)
qmf_32_subbands(s, k, subband_samples[k],
qmf_32_subbands(s, k, subband_samples,
s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0);
}
@ -983,7 +993,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
/* FIXME: This downmixing is probably broken with upsample.
* Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */
if (s->prim_channels + !!s->lfe > 2 &&
if (s->audio_header.prim_channels + !!s->lfe > 2 &&
s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
s->channel_order_tab);
@ -1060,7 +1070,7 @@ static int dca_subframe_footer(DCAContext *s, int base_channel)
return AVERROR_INVALIDDATA;
}
for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
for (in = 0; in < s->prim_channels + !!s->lfe; in++) {
for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
uint16_t tmp = get_bits(&s->gb, 9);
if ((tmp & 0xFF) > 241) {
av_log(s->avctx, AV_LOG_ERROR,
@ -1106,9 +1116,9 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
if (s->current_subframe >= s->audio_header.subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
s->current_subframe, s->audio_header.subframes);
return AVERROR_INVALIDDATA;
}
@ -1128,7 +1138,7 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
s->current_subsubframe = 0;
s->current_subframe++;
}
if (s->current_subframe >= s->subframes) {
if (s->current_subframe >= s->audio_header.subframes) {
/* Read subframe footer */
if ((ret = dca_subframe_footer(s, base_channel)))
return ret;
@ -1169,7 +1179,7 @@ static int scan_for_extensions(AVCodecContext *avctx)
case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize;
s->xch_base_channel = s->prim_channels;
s->xch_base_channel = s->audio_header.prim_channels;
/* validate sync word using XCHFSIZE field */
xch_fsize = show_bits(&s->gb, 10);
@ -1254,7 +1264,7 @@ static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_
if (s->amode < 16) {
avctx->channel_layout = dca_core_channel_layout[s->amode];
if (s->prim_channels + !!s->lfe > 2 &&
if (s->audio_header.prim_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
/*
* Neither the core's auxiliary data nor our default tables contain
@ -1289,7 +1299,7 @@ static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_
if (num_core_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
channels = 2;
s->output = s->prim_channels == 2 ? s->amode : DCA_STEREO;
s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
/* Stereo downmix coefficients
@ -1315,7 +1325,7 @@ static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_
if (num_core_channels + !!s->lfe >
FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
avpriv_request_sample(s->avctx, "Downmixing %d channels",
s->prim_channels + !!s->lfe);
s->audio_header.prim_channels + !!s->lfe);
return AVERROR_PATCHWELCOME;
}
for (i = 0; i < num_core_channels + !!s->lfe; i++) {
@ -1387,7 +1397,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
}
/* record number of core channels incase less than max channels are requested */
num_core_channels = s->prim_channels;
num_core_channels = s->audio_header.prim_channels;
if (s->ext_coding)
s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr];
@ -1398,7 +1408,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
avctx->profile = s->profile;
full_channels = channels = s->prim_channels + !!s->lfe;
full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
ret = set_channel_layout(avctx, channels, num_core_channels);
if (ret < 0)

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