#include "dcadsp.h"
#include "fmtconvert.h"
+#if ARCH_ARM
+# include "arm/dca.h"
+#endif
+
//#define TRACE
#define DCA_PRIM_CHANNELS_MAX (7)
* All 2 channel configurations -> AV_CH_LAYOUT_STEREO
*/
-static const int64_t dca_core_channel_layout[] = {
+static const uint64_t dca_core_channel_layout[] = {
AV_CH_FRONT_CENTER, ///< 1, A
AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono)
AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo)
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
/* Frame header */
int frame_type; ///< type of the current frame
int samples_deficit; ///< deficit sample count
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
- float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
- DECLARE_ALIGNED(16, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
- DECLARE_ALIGNED(16, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
+ DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
+ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
+ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
int hist_index[DCA_PRIM_CHANNELS_MAX];
- DECLARE_ALIGNED(16, float, raXin)[32];
+ DECLARE_ALIGNED(32, float, raXin)[32];
int output; ///< type of output
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)[(DCA_PRIM_CHANNELS_MAX+1)*256];
+ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
+ DECLARE_ALIGNED(32, float, samples)[(DCA_PRIM_CHANNELS_MAX+1)*256];
const float *samples_chanptr[DCA_PRIM_CHANNELS_MAX+1];
uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE];
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
/* Frame header */
s->frame_type = get_bits(&s->gb, 1);
s->sample_blocks = get_bits(&s->gb, 7) + 1;
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 95)
- return -1;
+ return AVERROR_INVALIDDATA;
s->amode = get_bits(&s->gb, 6);
s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate)
- return -1;
+ return AVERROR_INVALIDDATA;
s->bit_rate_index = get_bits(&s->gb, 5);
s->bit_rate = dca_bit_rates[s->bit_rate_index];
if (!s->bit_rate)
- return -1;
+ return AVERROR_INVALIDDATA;
s->downmix = get_bits(&s->gb, 1);
s->dynrange = get_bits(&s->gb, 1);
int j, k;
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (!base_channel) {
s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
else if (s->bitalloc_huffman[j] == 7) {
av_log(s->avctx, AV_LOG_ERROR,
"Invalid bit allocation index\n");
- return -1;
+ return AVERROR_INVALIDDATA;
} else {
s->bitalloc[j][k] =
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
if (s->bitalloc[j][k] > 26) {
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
// j, k, s->bitalloc[j][k]);
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
}
}
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
for (j = base_channel; j < s->prim_channels; j++) {
const uint32_t *scale_table;
}
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
/* Scale factors for joint subband coding */
for (j = base_channel; j < s->prim_channels; j++) {
else /* Perfect reconstruction */
prCoeff = fir_32bands_perfect;
+ for (i = sb_act; i < 32; i++)
+ s->raXin[i] = 0.0;
+
/* Reconstructed channel sample index */
for (subindex = 0; subindex < 8; subindex++) {
/* Load in one sample from each subband and clear inactive subbands */
for (i = 0; i < sb_act; i++){
- uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ ((i-1)&2)<<30;
+ unsigned sign = (i - 1) & 2;
+ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
AV_WN32A(&s->raXin[i], v);
}
- for (; i < 32; i++)
- s->raXin[i] = 0.0;
s->synth.synth_filter_float(&s->imdct,
s->subband_fir_hist[chans], &s->hist_index[chans],
}
+#ifndef decode_blockcodes
/* Very compact version of the block code decoder that does not use table
* look-up but is slightly slower */
static int decode_blockcode(int code, int levels, int *values)
code = div;
}
- if (code == 0)
- return 0;
- else {
- av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
- return -1;
- }
+ return code;
+}
+
+static int decode_blockcodes(int code1, int code2, int levels, int *values)
+{
+ return decode_blockcode(code1, levels, values) |
+ decode_blockcode(code2, levels, values + 4);
}
+#endif
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 };
+#ifndef int8x8_fmul_int32
+static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
+{
+ float fscale = scale / 16.0;
+ int i;
+ for (i = 0; i < 8; i++)
+ dst[i] = src[i] * fscale;
+}
+#endif
+
static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
{
int k, l;
for (k = base_channel; k < s->prim_channels; k++) {
if (get_bits_left(&s->gb) < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
for (l = 0; l < s->vq_start_subband[k]; l++) {
int m;
if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
if (abits <= 7){
/* Block code */
- int block_code1, block_code2, size, levels;
+ int block_code1, block_code2, size, levels, err;
size = abits_sizes[abits-1];
levels = abits_levels[abits-1];
block_code1 = get_bits(&s->gb, size);
- /* FIXME Should test return value */
- decode_blockcode(block_code1, levels, block);
block_code2 = get_bits(&s->gb, size);
- decode_blockcode(block_code2, levels, &block[4]);
+ err = decode_blockcodes(block_code1, block_code2,
+ levels, block);
+ if (err) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "ERROR: block code look-up failed\n");
+ return AVERROR_INVALIDDATA;
+ }
}else{
/* no coding */
for (m = 0; m < 8; m++)
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
/* 1 vector -> 32 samples but we only need the 8 samples
* for this subsubframe. */
- int m;
+ int hfvq = s->high_freq_vq[k][l];
if (!s->debug_flag & 0x01) {
av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
s->debug_flag |= 0x01;
}
- for (m = 0; m < 8; m++) {
- subband_samples[k][l][m] =
- high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
- m]
- * (float) s->scale_factor[k][l][0] / 16.0;
- }
+ int8x8_fmul_int32(subband_samples[k][l],
+ &high_freq_vq[hfvq][subsubframe * 8],
+ s->scale_factor[k][l][0]);
}
}
/* presumably optional information only appears in the core? */
if (!base_channel) {
if (s->timestamp)
- get_bits(&s->gb, 32);
+ skip_bits_long(&s->gb, 32);
if (s->aux_data)
aux_data_count = get_bits(&s->gb, 6);
static int dca_decode_block(DCAContext * s, int base_channel, int block_index)
{
+ int ret;
/* Sanity check */
if (s->current_subframe >= s->subframes) {
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
s->current_subframe, s->subframes);
- return -1;
+ return AVERROR_INVALIDDATA;
}
if (!s->current_subsubframe) {
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
#endif
/* Read subframe header */
- if (dca_subframe_header(s, base_channel, block_index))
- return -1;
+ if ((ret = dca_subframe_header(s, base_channel, block_index)))
+ return ret;
}
/* Read subsubframe */
#ifdef TRACE
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
#endif
- if (dca_subsubframe(s, base_channel, block_index))
- return -1;
+ if ((ret = dca_subsubframe(s, base_channel, block_index)))
+ return ret;
/* Update state */
s->current_subsubframe++;
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
#endif
/* Read subframe footer */
- if (dca_subframe_footer(s, base_channel))
- return -1;
+ if ((ret = dca_subframe_footer(s, base_channel)))
+ return ret;
}
return 0;
PutBitContext pb;
if ((unsigned)src_size > (unsigned)max_size) {
-// av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
+// av_log(NULL, AV_LOG_ERROR, "Input frame size larger than DCA_MAX_FRAME_SIZE!\n");
// return -1;
src_size = max_size;
}
flush_put_bits(&pb);
return (put_bits_count(&pb) + 7) >> 3;
default:
- return -1;
+ return AVERROR_INVALIDDATA;
}
}
{
int ss_index;
int blownup;
- int header_size;
- int hd_size;
int num_audiop = 1;
int num_assets = 1;
int active_ss_mask[8];
ss_index = get_bits(&s->gb, 2);
blownup = get_bits1(&s->gb);
- header_size = get_bits(&s->gb, 8 + 4 * blownup) + 1;
- hd_size = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;
+ skip_bits(&s->gb, 8 + 4 * blownup); // header_size
+ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size
s->static_fields = get_bits1(&s->gb);
if (s->static_fields) {
* Main frame decoding function
* FIXME add arguments
*/
-static int dca_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int dca_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int lfe_samples;
int num_core_channels = 0;
- int i;
- int16_t *samples = data;
+ int i, ret;
+ float *samples_flt;
+ int16_t *samples_s16;
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer,
DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
- if (s->dca_buffer_size == -1) {
+ if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
- if (dca_parse_frame_header(s) < 0) {
+ if ((ret = dca_parse_frame_header(s)) < 0) {
//seems like the frame is corrupt, try with the next one
- *data_size=0;
- return buf_size;
+ return ret;
}
//set AVCodec values with parsed data
avctx->sample_rate = s->sample_rate;
avctx->bit_rate = s->bit_rate;
+ avctx->frame_size = s->sample_blocks * 32;
s->profile = FF_PROFILE_DTS;
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, 0, i);
+ if ((ret = dca_decode_block(s, 0, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding block\n");
+ return ret;
+ }
}
/* record number of core channels incase less than max channels are requested */
dca_parse_audio_coding_header(s, s->xch_base_channel);
for (i = 0; i < (s->sample_blocks / 8); i++) {
- dca_decode_block(s, s->xch_base_channel, i);
+ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) {
+ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n");
+ continue;
+ }
}
s->xch_present = 1;
if (channels > !!s->lfe &&
s->channel_order_tab[channels - 1 - !!s->lfe] < 0)
- return -1;
+ return AVERROR_INVALIDDATA;
if (avctx->request_channels == 2 && s->prim_channels > 2) {
channels = 2;
}
} else {
av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode);
- return -1;
+ return AVERROR_INVALIDDATA;
}
/* There is nothing that prevents a dts frame to change channel configuration
- but FFmpeg doesn't support that so only set the channels if it is previously
+ but Libav doesn't support that so only set the channels if it is previously
unset. Ideally during the first probe for channels the crc should be checked
and only set avctx->channels when the crc is ok. Right now the decoder could
set the channels based on a broken first frame.*/
if (avctx->channels != channels) {
av_log(avctx, AV_LOG_ERROR, "DCA decoder does not support number of "
"channels changing in stream. Skipping frame.\n");
- return -1;
+ return AVERROR_PATCHWELCOME;
}
- 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;
+ /* get output buffer */
+ s->frame.nb_samples = 256 * (s->sample_blocks / 8);
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples_flt = (float *)s->frame.data[0];
+ samples_s16 = (int16_t *)s->frame.data[0];
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
float* back_chan = s->samples + s->channel_order_tab[s->xch_base_channel] * 256;
float* lt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 2] * 256;
float* rt_chan = s->samples + s->channel_order_tab[s->xch_base_channel - 1] * 256;
- int j;
- for(j = 0; j < 256; ++j) {
- lt_chan[j] -= back_chan[j] * M_SQRT1_2;
- rt_chan[j] -= back_chan[j] * M_SQRT1_2;
- }
+ s->dsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
+ s->dsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
}
- s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
- samples += 256 * channels;
+ if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
+ s->fmt_conv.float_interleave(samples_flt, s->samples_chanptr, 256,
+ channels);
+ samples_flt += 256 * channels;
+ } else {
+ s->fmt_conv.float_to_int16_interleave(samples_s16,
+ s->samples_chanptr, 256,
+ channels);
+ samples_s16 += 256 * channels;
+ }
}
/* update lfe history */
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
}
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return buf_size;
}
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
- s->scale_bias = 1.0;
+ if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+ s->scale_bias = 1.0 / 32768.0;
+ } else {
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->scale_bias = 1.0;
+ }
/* allow downmixing to stereo */
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
avctx->channels = avctx->request_channels;
}
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
.decode = dca_decode_frame,
.close = dca_decode_end,
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .capabilities = CODEC_CAP_CHANNEL_CONF,
+ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
+ .sample_fmts = (const enum AVSampleFormat[]) {
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
+ },
.profiles = NULL_IF_CONFIG_SMALL(profiles),
};
projects / ffmpeg / blobdiff