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ffmpeg.wasm-core
Commit a71fff21 authored by Hendrik Leppkes's avatar Hendrik Leppkes
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Merge commit '58b42345' 

* commit '58b42345':
  dcadec: reorganise context data
Merged-by: 's avatarHendrik Leppkes <h.leppkes@gmail.com>
parents d3952510 58b42345
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Showing with 197 additions and 169 deletions
libavcodec/dca.h
View file @ a71fff21
...@@ -132,6 +132,47 @@ typedef struct QMF64_table { ...@@ -132,6 +132,47 @@ typedef struct QMF64_table {
float rsin[32]; float rsin[32];
} QMF64_table; } 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 { typedef struct DCAContext {
const AVClass *class; ///< class for AVOptions const AVClass *class; ///< class for AVOptions
AVCodecContext *avctx; AVCodecContext *avctx;
...@@ -165,28 +206,11 @@ typedef struct DCAContext { ...@@ -165,28 +206,11 @@ typedef struct DCAContext {
int dialog_norm; ///< dialog normalisation parameter int dialog_norm; ///< dialog normalisation parameter
/* Primary audio coding header */ /* Primary audio coding header */
int subframes; ///< number of subframes DCAAudioHeader audio_header;
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
/* Primary audio coding side information */ /* Primary audio coding side information */
int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count 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 float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///< stereo downmix coefficients
int dynrange_coef; ///< dynamic range coefficient int dynrange_coef; ///< dynamic range coefficient
...@@ -197,23 +221,17 @@ typedef struct DCAContext { ...@@ -197,23 +221,17 @@ typedef struct DCAContext {
uint8_t core_downmix_amode; ///< audio channel arrangement of embedded downmix 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) 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 float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data
int lfe_scale_factor; int lfe_scale_factor;
/* Subband samples history (for ADPCM) */ /* 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]; DECLARE_ALIGNED(32, float, raXin)[32];
DCAChan dca_chan[DCA_PRIM_CHANNELS_MAX];
int output; ///< type of output 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 *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1];
float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1]; float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1];
uint8_t *extra_channels_buffer; uint8_t *extra_channels_buffer;
......
libavcodec/dcadec.c
View file @ a71fff21
...@@ -238,8 +238,8 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, ...@@ -238,8 +238,8 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
return AVERROR_INVALIDDATA; return AVERROR_INVALIDDATA;
} }
s->total_channels = nchans + base_channel; s->audio_header.total_channels = nchans + base_channel;
s->prim_channels = s->total_channels; s->audio_header.prim_channels = s->audio_header.total_channels;
/* obtain speaker layout mask & downmix coefficients for XXCH */ /* obtain speaker layout mask & downmix coefficients for XXCH */
if (xxch) { if (xxch) {
...@@ -266,11 +266,11 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, ...@@ -266,11 +266,11 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
s->xxch_dmix_sf[s->xxch_chset] = scale_factor; s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
for (i = base_channel; i < s->prim_channels; i++) { for (i = base_channel; i < s->audio_header.prim_channels; i++) {
mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
} }
for (j = base_channel; j < s->prim_channels; j++) { for (j = base_channel; j < s->audio_header.prim_channels; j++) {
memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
s->xxch_dmix_embedded |= (embedded_downmix << j); s->xxch_dmix_embedded |= (embedded_downmix << j);
for (i = 0; i < s->xxch_nbits_spk_mask; i++) { for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
...@@ -294,40 +294,44 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, ...@@ -294,40 +294,44 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
} }
} }
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
s->prim_channels = DCA_PRIM_CHANNELS_MAX; s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
for (i = base_channel; i < s->prim_channels; i++) { for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->subband_activity[i] = get_bits(&s->gb, 5) + 2; s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
if (s->subband_activity[i] > DCA_SUBBANDS) if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
s->subband_activity[i] = DCA_SUBBANDS; s->audio_header.subband_activity[i] = DCA_SUBBANDS;
} }
for (i = base_channel; i < s->prim_channels; i++) { for (i = base_channel; i < s->audio_header.prim_channels; i++) {
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
if (s->vq_start_subband[i] > DCA_SUBBANDS) if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
s->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->audio_header.joint_intensity + base_channel,
get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2); s->audio_header.prim_channels - base_channel, 3);
get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3); get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3); 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 */ /* Get codebooks quantization indexes */
if (!base_channel) 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 (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++) for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); s->audio_header.quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
/* Get scale factor adjustment */ /* Get scale factor adjustment */
for (j = 0; j < 11; j++) for (j = 0; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++) for (i = base_channel; i < s->audio_header.prim_channels; i++)
s->scalefactor_adj[i][j] = 1; s->audio_header.scalefactor_adj[i][j] = 1;
for (j = 1; j < 11; j++) for (j = 1; j < 11; j++)
for (i = base_channel; i < s->prim_channels; i++) for (i = base_channel; i < s->audio_header.prim_channels; i++)
if (s->quant_index_huffman[i][j] < thr[j]) if (s->audio_header.quant_index_huffman[i][j] < thr[j])
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
if (!xxch) { if (!xxch) {
if (s->crc_present) { if (s->crc_present) {
...@@ -406,7 +410,7 @@ static int dca_parse_frame_header(DCAContext *s) ...@@ -406,7 +410,7 @@ static int dca_parse_frame_header(DCAContext *s)
s->output |= DCA_LFE; s->output |= DCA_LFE;
/* Primary audio coding header */ /* 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, 0); return dca_parse_audio_coding_header(s, 0, 0);
} }
...@@ -445,53 +449,53 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) ...@@ -445,53 +449,53 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
} }
for (j = base_channel; j < s->prim_channels; j++) { for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) for (k = 0; k < s->audio_header.subband_activity[j]; k++)
s->prediction_mode[j][k] = get_bits(&s->gb, 1); s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
} }
/* Get prediction codebook */ /* Get prediction codebook */
for (j = base_channel; j < s->prim_channels; j++) { for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (s->prediction_mode[j][k] > 0) { if (s->dca_chan[j].prediction_mode[k] > 0) {
/* (Prediction coefficient VQ address) */ /* (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 */ /* Bit allocation index */
for (j = base_channel; j < s->prim_channels; j++) { for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->vq_start_subband[j]; k++) { for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
if (s->bitalloc_huffman[j] == 6) if (s->audio_header.bitalloc_huffman[j] == 6)
s->bitalloc[j][k] = get_bits(&s->gb, 5); s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
else if (s->bitalloc_huffman[j] == 5) else if (s->audio_header.bitalloc_huffman[j] == 5)
s->bitalloc[j][k] = get_bits(&s->gb, 4); s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
else if (s->bitalloc_huffman[j] == 7) { else if (s->audio_header.bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR, av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n"); "Invalid bit allocation index\n");
return AVERROR_INVALIDDATA; return AVERROR_INVALIDDATA;
} else { } else {
s->bitalloc[j][k] = s->dca_chan[j].bitalloc[k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); 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", 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; return AVERROR_INVALIDDATA;
} }
} }
} }
/* Transition mode */ /* Transition mode */
for (j = base_channel; j < s->prim_channels; j++) { for (j = base_channel; j < s->audio_header.prim_channels; j++) {
for (k = 0; k < s->subband_activity[j]; k++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
s->transition_mode[j][k] = 0; s->dca_chan[j].transition_mode[k] = 0;
if (s->subsubframes[s->current_subframe] > 1 && if (s->subsubframes[s->current_subframe] > 1 &&
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
s->transition_mode[j][k] = s->dca_chan[j].transition_mode[k] =
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
} }
} }
} }
...@@ -499,14 +503,14 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) ...@@ -499,14 +503,14 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
if (get_bits_left(&s->gb) < 0) if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA; 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; const uint32_t *scale_table;
int scale_sum, log_size; int scale_sum, log_size;
memset(s->scale_factor[j], 0, memset(s->dca_chan[j].scale_factor, 0,
s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); 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; scale_table = ff_dca_scale_factor_quant7;
log_size = 7; log_size = 7;
} else { } else {
...@@ -517,45 +521,46 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) ...@@ -517,45 +521,46 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
/* When huffman coded, only the difference is encoded */ /* When huffman coded, only the difference is encoded */
scale_sum = 0; scale_sum = 0;
for (k = 0; k < s->subband_activity[j]; k++) { for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][0] = scale_table[scale_sum]; 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 */ /* Get second scale factor */
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
s->scale_factor[j][k][1] = scale_table[scale_sum]; s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
} }
} }
} }
/* Joint subband scale factor codebook select */ /* 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 */ /* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0) if (s->audio_header.joint_intensity[j] > 0)
s->joint_huff[j] = get_bits(&s->gb, 3); s->dca_chan[j].joint_huff = get_bits(&s->gb, 3);
} }
if (get_bits_left(&s->gb) < 0) if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA; return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */ /* 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; int source_channel;
/* Transmitted only if joint subband coding enabled */ /* Transmitted only if joint subband coding enabled */
if (s->joint_intensity[j] > 0) { if (s->audio_header.joint_intensity[j] > 0) {
int scale = 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 /* When huffman coded, only the difference is encoded
* (is this valid as well for joint scales ???) */ * (is this valid as well for joint scales ???) */
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { for (k = s->audio_header.subband_activity[j];
scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7); k < s->audio_header.subband_activity[source_channel]; k++) {
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ 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)) { if (!(s->debug_flag & 0x02)) {
...@@ -580,10 +585,10 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) ...@@ -580,10 +585,10 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
*/ */
/* VQ encoded high frequency subbands */ /* VQ encoded high frequency subbands */
for (j = base_channel; j < s->prim_channels; j++) for (j = base_channel; j < s->audio_header.prim_channels; j++)
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
/* 1 vector -> 32 samples */ /* 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 */ /* Low frequency effect data */
if (!base_channel && s->lfe) { if (!base_channel && s->lfe) {
...@@ -622,7 +627,7 @@ static void qmf_32_subbands(DCAContext *s, int chans, ...@@ -622,7 +627,7 @@ static void qmf_32_subbands(DCAContext *s, int chans,
{ {
const float *prCoeff; const float *prCoeff;
int sb_act = s->subband_activity[chans]; int sb_act = s->audio_header.subband_activity[chans];
scale *= sqrt(1 / 8.0); scale *= sqrt(1 / 8.0);
...@@ -633,9 +638,9 @@ static void qmf_32_subbands(DCAContext *s, int chans, ...@@ -633,9 +638,9 @@ static void qmf_32_subbands(DCAContext *s, int chans,
prCoeff = ff_dca_fir_32bands_perfect; prCoeff = ff_dca_fir_32bands_perfect;
s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct, s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
s->subband_fir_hist[chans], s->dca_chan[chans].subband_fir_hist,
&s->hist_index[chans], &s->dca_chan[chans].hist_index,
s->subband_fir_noidea[chans], prCoeff, s->dca_chan[chans].subband_fir_noidea, prCoeff,
samples_out, s->raXin, scale); samples_out, s->raXin, scale);
} }
...@@ -670,14 +675,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL ...@@ -670,14 +675,14 @@ static void qmf_64_subbands(DCAContext *s, int chans, float samples_in[64][SAMPL
{ {
float raXin[64]; float raXin[64];
float A[32], B[32]; float A[32], B[32];
float *raX = s->subband_fir_hist[chans]; float *raX = s->dca_chan[chans].subband_fir_hist;
float *raZ = s->subband_fir_noidea[chans]; float *raZ = s->dca_chan[chans].subband_fir_noidea;
unsigned i, j, k, subindex; 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; raXin[i] = 0.0;
for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) { 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]; raXin[i] = samples_in[i][subindex];
for (k = 0; k < 32; k++) { for (k = 0; k < 32; k++) {
...@@ -866,8 +871,6 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -866,8 +871,6 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
const float *quant_step_table; 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]); LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]);
/* /*
...@@ -880,17 +883,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -880,17 +883,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
else else
quant_step_table = ff_dca_lossy_quant_d; 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]; float rscale[DCA_SUBBANDS];
if (get_bits_left(&s->gb) < 0) if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA; 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; int m;
/* Select the mid-tread linear quantizer */ /* 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]; float quant_step_size = quant_step_table[abits];
...@@ -899,7 +903,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -899,7 +903,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
*/ */
/* Select quantization index code book */ /* 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 * Extract bits from the bit stream
...@@ -909,9 +913,10 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -909,9 +913,10 @@ 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])); memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0]));
} else { } else {
/* Deal with transients */ /* Deal with transients */
int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l]; int sfi = s->dca_chan[k].transition_mode[l] &&
rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] * subsubframe >= s->dca_chan[k].transition_mode[l];
s->scalefactor_adj[k][sel]; 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 >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
if (abits <= 7) { if (abits <= 7) {
...@@ -944,54 +949,61 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -944,54 +949,61 @@ 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], s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0],
block, rscale, SAMPLES_PER_SUBBAND * s->vq_start_subband[k]); 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; int m;
/* /*
* Inverse ADPCM if in prediction mode * Inverse ADPCM if in prediction mode
*/ */
if (s->prediction_mode[k][l]) { if (s->dca_chan[k].prediction_mode[l]) {
int n; int n;
if (s->predictor_history) if (s->predictor_history)
subband_samples[k][l][0] += (ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] * subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
s->subband_samples_hist[k][l][3] + s->dca_chan[k].subband_samples_hist[l][3] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][1] * ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
s->subband_samples_hist[k][l][2] + s->dca_chan[k].subband_samples_hist[l][2] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][2] * ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
s->subband_samples_hist[k][l][1] + s->dca_chan[k].subband_samples_hist[l][1] +
ff_dca_adpcm_vb[s->prediction_vq[k][l]][3] * ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
s->subband_samples_hist[k][l][0]) * s->dca_chan[k].subband_samples_hist[l][0]) *
(1.0f / 8192); (1.0f / 8192);
for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
float sum = ff_dca_adpcm_vb[s->prediction_vq[k][l]][0] * float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
subband_samples[k][l][m - 1]; subband_samples[l][m - 1];
for (n = 2; n <= 4; n++) for (n = 2; n <= 4; n++)
if (m >= n) if (m >= n)
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] * sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
subband_samples[k][l][m - n]; subband_samples[l][m - n];
else if (s->predictor_history) else if (s->predictor_history)
sum += ff_dca_adpcm_vb[s->prediction_vq[k][l]][n - 1] * sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
s->subband_samples_hist[k][l][m - n + 4]; s->dca_chan[k].subband_samples_hist[l][m - n + 4];
subband_samples[k][l][m] += sum * (1.0f / 8192); 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 * 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)) { if (!(s->debug_flag & 0x01)) {
av_log(s->avctx, AV_LOG_DEBUG, av_log(s->avctx, AV_LOG_DEBUG,
"Stream with high frequencies VQ coding\n"); "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01; 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, ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND,
s->scale_factor[k], s->vq_start_subband[k], s->dca_chan[k].scale_factor,
s->subband_activity[k]); s->audio_header.vq_start_subband[k],
s->audio_header.subband_activity[k]);
} }
} }
...@@ -1003,17 +1015,11 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ...@@ -1003,17 +1015,11 @@ 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; return 0;
} }
static int dca_filter_channels(DCAContext *s, int block_index, int upsample) 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; int k;
if (upsample) { if (upsample) {
...@@ -1024,18 +1030,22 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) ...@@ -1024,18 +1030,22 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
} }
/* 64 subbands QMF */ /* 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) 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]], s->samples_chanptr[s->channel_order_tab[k]],
/* Upsampling needs a factor 2 here. */ /* Upsampling needs a factor 2 here. */
M_SQRT2 / 32768.0); M_SQRT2 / 32768.0);
} }
} else { } else {
/* 32 subbands QMF */ /* 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) 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]], s->samples_chanptr[s->channel_order_tab[k]],
M_SQRT1_2 / 32768.0); M_SQRT1_2 / 32768.0);
} }
...@@ -1062,7 +1072,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) ...@@ -1062,7 +1072,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
/* FIXME: This downmixing is probably broken with upsample. /* FIXME: This downmixing is probably broken with upsample.
* Probably totally broken also with XLL in general. */ * Probably totally broken also with XLL in general. */
/* Downmixing to Stereo */ /* 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) { s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef, dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
s->channel_order_tab); s->channel_order_tab);
...@@ -1139,7 +1149,7 @@ static int dca_subframe_footer(DCAContext *s, int base_channel) ...@@ -1139,7 +1149,7 @@ static int dca_subframe_footer(DCAContext *s, int base_channel)
return AVERROR_INVALIDDATA; return AVERROR_INVALIDDATA;
} }
for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) { 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); uint16_t tmp = get_bits(&s->gb, 9);
if ((tmp & 0xFF) > 241) { if ((tmp & 0xFF) > 241) {
av_log(s->avctx, AV_LOG_ERROR, av_log(s->avctx, AV_LOG_ERROR,
...@@ -1185,9 +1195,9 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index) ...@@ -1185,9 +1195,9 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
int ret; int ret;
/* Sanity check */ /* 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", 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; return AVERROR_INVALIDDATA;
} }
...@@ -1207,7 +1217,7 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index) ...@@ -1207,7 +1217,7 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
s->current_subsubframe = 0; s->current_subsubframe = 0;
s->current_subframe++; s->current_subframe++;
} }
if (s->current_subframe >= s->subframes) { if (s->current_subframe >= s->audio_header.subframes) {
/* Read subframe footer */ /* Read subframe footer */
if ((ret = dca_subframe_footer(s, base_channel))) if ((ret = dca_subframe_footer(s, base_channel)))
return ret; return ret;
...@@ -1260,7 +1270,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) ...@@ -1260,7 +1270,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s)
/* loop over the channel data sets */ /* loop over the channel data sets */
/* only decode as many channels as we've decoded base data for */ /* only decode as many channels as we've decoded base data for */
for(chset = 0, chan_base = 0; for(chset = 0, chan_base = 0;
chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels; chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels;
chan_base += n_xbr_ch[chset++]) { chan_base += n_xbr_ch[chset++]) {
int start_posn = get_bits_count(&s->gb); int start_posn = get_bits_count(&s->gb);
int subsubframe = 0; int subsubframe = 0;
...@@ -1293,7 +1303,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) ...@@ -1293,7 +1303,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s)
int nbits; int nbits;
int scale_table_size; int scale_table_size;
if (s->scalefactor_huffman[chan_base+i] == 6) { if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) {
scale_table = ff_dca_scale_factor_quant7; scale_table = ff_dca_scale_factor_quant7;
scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
} else { } else {
...@@ -1312,7 +1322,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s) ...@@ -1312,7 +1322,7 @@ int ff_dca_xbr_parse_frame(DCAContext *s)
} }
scale_table_high[i][j][0] = scale_table[index]; scale_table_high[i][j][0] = scale_table[index];
if(xbr_tmode && s->transition_mode[i][j]) { if(xbr_tmode && s->dca_chan[i].transition_mode[j]) {
int index = get_bits(&s->gb, nbits); int index = get_bits(&s->gb, nbits);
if (index >= scale_table_size) { if (index >= scale_table_size) {
av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
...@@ -1330,9 +1340,9 @@ int ff_dca_xbr_parse_frame(DCAContext *s) ...@@ -1330,9 +1340,9 @@ int ff_dca_xbr_parse_frame(DCAContext *s)
for(j = 0; j < active_bands[chset][i]; j++) { for(j = 0; j < active_bands[chset][i]; j++) {
const int xbr_abits = abits_high[i][j]; const int xbr_abits = abits_high[i][j];
const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits]; const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits];
const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j]; const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j];
const float rscale = quant_step_size * scale_table_high[i][j][sfi]; const float rscale = quant_step_size * scale_table_high[i][j][sfi];
float *subband_samples = s->subband_samples[k][chan_base+i][j]; float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j];
int block[8]; int block[8];
if(xbr_abits <= 0) if(xbr_abits <= 0)
...@@ -1421,7 +1431,7 @@ int ff_dca_xxch_decode_frame(DCAContext *s) ...@@ -1421,7 +1431,7 @@ int ff_dca_xxch_decode_frame(DCAContext *s)
for (chset = 0; chset < num_chsets; chset++) { for (chset = 0; chset < num_chsets; chset++) {
chstart = get_bits_count(&s->gb); chstart = get_bits_count(&s->gb);
base_channel = s->prim_channels; base_channel = s->audio_header.prim_channels;
s->xxch_chset = chset; s->xxch_chset = chset;
/* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
...@@ -1479,7 +1489,7 @@ static int scan_for_extensions(AVCodecContext *avctx) ...@@ -1479,7 +1489,7 @@ static int scan_for_extensions(AVCodecContext *avctx)
case DCA_SYNCWORD_XCH: { case DCA_SYNCWORD_XCH: {
int ext_amode, xch_fsize; 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 */ /* validate sync word using XCHFSIZE field */
xch_fsize = show_bits(&s->gb, 10); xch_fsize = show_bits(&s->gb, 10);
...@@ -1576,7 +1586,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core ...@@ -1576,7 +1586,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core
if (s->amode < 16) { if (s->amode < 16) {
avctx->channel_layout = ff_dca_core_channel_layout[s->amode]; avctx->channel_layout = ff_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) { avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
/* /*
* Neither the core's auxiliary data nor our default tables contain * Neither the core's auxiliary data nor our default tables contain
...@@ -1622,7 +1632,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core ...@@ -1622,7 +1632,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core
if (num_core_channels + !!s->lfe > 2 && if (num_core_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
*channels = 2; *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; avctx->channel_layout = AV_CH_LAYOUT_STEREO;
} }
else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
...@@ -1642,7 +1652,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core ...@@ -1642,7 +1652,7 @@ static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core
channel_mask = s->xxch_core_spkmask; channel_mask = s->xxch_core_spkmask;
{ {
*channels = s->prim_channels + !!s->lfe; *channels = s->audio_header.prim_channels + !!s->lfe;
for (i = 0; i < s->xxch_chset; i++) { for (i = 0; i < s->xxch_chset; i++) {
channel_mask |= s->xxch_spk_masks[i]; channel_mask |= s->xxch_spk_masks[i];
} }
...@@ -1751,9 +1761,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, ...@@ -1751,9 +1761,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
} }
/* record number of core channels incase less than max channels are requested */ /* 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->prim_channels + !!s->lfe > 2 && if (s->audio_header.prim_channels + !!s->lfe > 2 &&
avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
/* Stereo downmix coefficients /* Stereo downmix coefficients
* *
...@@ -1778,7 +1788,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, ...@@ -1778,7 +1788,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
if (num_core_channels + !!s->lfe > if (num_core_channels + !!s->lfe >
FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) { FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
avpriv_request_sample(s->avctx, "Downmixing %d channels", avpriv_request_sample(s->avctx, "Downmixing %d channels",
s->prim_channels + !!s->lfe); s->audio_header.prim_channels + !!s->lfe);
return AVERROR_PATCHWELCOME; return AVERROR_PATCHWELCOME;
} }
for (i = 0; i < num_core_channels + !!s->lfe; i++) { for (i = 0; i < num_core_channels + !!s->lfe; i++) {
...@@ -1805,7 +1815,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, ...@@ -1805,7 +1815,7 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data,
avctx->profile = s->profile; 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); ret = set_channel_layout(avctx, &channels, num_core_channels);
if (ret < 0) if (ret < 0)
......
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