/*
- * DCA compatible decoder
- * Copyright (C) 2004 Gildas Bazin
- * Copyright (C) 2004 Benjamin Zores
- * Copyright (C) 2006 Benjamin Larsson
- * Copyright (C) 2007 Konstantin Shishkov
- * Copyright (C) 2012 Paul B Mahol
- * Copyright (C) 2014 Niels Möller
+ * Copyright (C) 2016 foo86
*
* This file is part of FFmpeg.
*
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
-#include "libavutil/attributes.h"
-#include "libavutil/channel_layout.h"
-#include "libavutil/common.h"
-#include "libavutil/float_dsp.h"
-#include "libavutil/internal.h"
-#include "libavutil/intreadwrite.h"
-#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
+#include "libavutil/channel_layout.h"
-#include "avcodec.h"
-#include "dca.h"
+#include "dcadec.h"
+#include "dcamath.h"
#include "dca_syncwords.h"
-#include "dcadata.h"
-#include "dcadsp.h"
-#include "dcahuff.h"
-#include "fft.h"
-#include "fmtconvert.h"
-#include "get_bits.h"
-#include "internal.h"
-#include "mathops.h"
#include "profiles.h"
-#include "synth_filter.h"
-
-#if ARCH_ARM
-# include "arm/dca.h"
-#endif
-
-enum DCAMode {
- DCA_MONO = 0,
- DCA_CHANNEL,
- DCA_STEREO,
- DCA_STEREO_SUMDIFF,
- DCA_STEREO_TOTAL,
- DCA_3F,
- DCA_2F1R,
- DCA_3F1R,
- DCA_2F2R,
- DCA_3F2R,
- DCA_4F2R
-};
-
-
-enum DCAXxchSpeakerMask {
- DCA_XXCH_FRONT_CENTER = 0x0000001,
- DCA_XXCH_FRONT_LEFT = 0x0000002,
- DCA_XXCH_FRONT_RIGHT = 0x0000004,
- DCA_XXCH_SIDE_REAR_LEFT = 0x0000008,
- DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010,
- DCA_XXCH_LFE1 = 0x0000020,
- DCA_XXCH_REAR_CENTER = 0x0000040,
- DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080,
- DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100,
- DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200,
- DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400,
- DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800,
- DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000,
- DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000,
- DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000,
- DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000,
- DCA_XXCH_LFE2 = 0x0010000,
- DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000,
- DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000,
- DCA_XXCH_OVERHEAD = 0x0080000,
- DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000,
- DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000,
- DCA_XXCH_REAR_HIGH_CENTER = 0x0400000,
- DCA_XXCH_REAR_HIGH_LEFT = 0x0800000,
- DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000,
- DCA_XXCH_REAR_LOW_CENTER = 0x2000000,
- DCA_XXCH_REAR_LOW_LEFT = 0x4000000,
- DCA_XXCH_REAR_LOW_RIGHT = 0x8000000,
-};
-
-#define DCA_DOLBY 101 /* FIXME */
-
-#define DCA_CHANNEL_BITS 6
-#define DCA_CHANNEL_MASK 0x3F
-
-#define DCA_LFE 0x80
-
-#define HEADER_SIZE 14
-
-#define DCA_NSYNCAUX 0x9A1105A0
-
-/** Bit allocation */
-typedef struct BitAlloc {
- int offset; ///< code values offset
- int maxbits[8]; ///< max bits in VLC
- int wrap; ///< wrap for get_vlc2()
- VLC vlc[8]; ///< actual codes
-} BitAlloc;
-static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
-static BitAlloc dca_tmode; ///< transition mode VLCs
-static BitAlloc dca_scalefactor; ///< scalefactor VLCs
-static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
+#define MIN_PACKET_SIZE 16
+#define MAX_PACKET_SIZE 0x104000
-static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba,
- int idx)
+int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
{
- return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) +
- ba->offset;
-}
-
-static float dca_dmix_code(unsigned code);
-
-static av_cold void dca_init_vlcs(void)
-{
- static int vlcs_initialized = 0;
- int i, j, c = 14;
- static VLC_TYPE dca_table[23622][2];
-
- if (vlcs_initialized)
- return;
-
- dca_bitalloc_index.offset = 1;
- dca_bitalloc_index.wrap = 2;
- for (i = 0; i < 5; i++) {
- dca_bitalloc_index.vlc[i].table = &dca_table[ff_dca_vlc_offs[i]];
- dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_dca_vlc_offs[i];
- init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
- bitalloc_12_bits[i], 1, 1,
- bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
- dca_scalefactor.offset = -64;
- dca_scalefactor.wrap = 2;
- for (i = 0; i < 5; i++) {
- dca_scalefactor.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 5]];
- dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_dca_vlc_offs[i + 5];
- init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
- scales_bits[i], 1, 1,
- scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
- dca_tmode.offset = 0;
- dca_tmode.wrap = 1;
- for (i = 0; i < 4; i++) {
- dca_tmode.vlc[i].table = &dca_table[ff_dca_vlc_offs[i + 10]];
- dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_dca_vlc_offs[i + 10];
- init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
- tmode_bits[i], 1, 1,
- tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC);
- }
-
- for (i = 0; i < 10; i++)
- for (j = 0; j < 7; j++) {
- if (!bitalloc_codes[i][j])
- break;
- dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i];
- dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4);
- dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[ff_dca_vlc_offs[c]];
- dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_dca_vlc_offs[c];
-
- init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j],
- bitalloc_sizes[i],
- bitalloc_bits[i][j], 1, 1,
- bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC);
- c++;
- }
- vlcs_initialized = 1;
-}
-
-static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
-{
- while (len--)
- *dst++ = get_bits(gb, bits);
-}
-
-static inline int dca_xxch2index(DCAContext *s, int xxch_ch)
-{
- int i, base, mask;
-
- /* locate channel set containing the channel */
- for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1);
- i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i])
- base += av_popcount(mask);
-
- return base + av_popcount(mask & (xxch_ch - 1));
-}
-
-static int dca_parse_audio_coding_header(DCAContext *s, int base_channel,
- int xxch)
-{
- int i, j;
- static const uint8_t adj_table[4] = { 16, 18, 20, 23 };
- 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 };
- int hdr_pos = 0, hdr_size = 0;
- float scale_factor;
- int this_chans, acc_mask;
- int embedded_downmix;
- int nchans, mask[8];
- int coeff, ichan;
-
- /* xxch has arbitrary sized audio coding headers */
- if (xxch) {
- hdr_pos = get_bits_count(&s->gb);
- hdr_size = get_bits(&s->gb, 7) + 1;
- }
-
- nchans = get_bits(&s->gb, 3) + 1;
- if (xxch && nchans >= 3) {
- av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans);
- return AVERROR_INVALIDDATA;
- } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) {
- av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel);
- return AVERROR_INVALIDDATA;
- }
-
- s->audio_header.total_channels = nchans + base_channel;
- s->audio_header.prim_channels = s->audio_header.total_channels;
-
- /* obtain speaker layout mask & downmix coefficients for XXCH */
- if (xxch) {
- acc_mask = s->xxch_core_spkmask;
-
- this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6;
- s->xxch_spk_masks[s->xxch_chset] = this_chans;
- s->xxch_chset_nch[s->xxch_chset] = nchans;
-
- for (i = 0; i <= s->xxch_chset; i++)
- acc_mask |= s->xxch_spk_masks[i];
-
- /* check for downmixing information */
- if (get_bits1(&s->gb)) {
- embedded_downmix = get_bits1(&s->gb);
- coeff = get_bits(&s->gb, 6);
-
- if (coeff<1 || coeff>61) {
- av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
- return AVERROR_INVALIDDATA;
- }
-
- scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3);
-
- s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
-
- for (i = base_channel; i < s->audio_header.prim_channels; i++) {
- mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
- }
-
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0]));
- s->xxch_dmix_embedded |= (embedded_downmix << j);
- for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
- if (mask[j] & (1 << i)) {
- if ((1 << i) == DCA_XXCH_LFE1) {
- av_log(s->avctx, AV_LOG_WARNING,
- "DCA-XXCH: dmix to LFE1 not supported.\n");
- continue;
- }
-
- coeff = get_bits(&s->gb, 7);
- ichan = dca_xxch2index(s, 1 << i);
- if ((coeff&63)<1 || (coeff&63)>61) {
- av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
- return AVERROR_INVALIDDATA;
- }
- s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);
- }
+ static const uint8_t dca2wav_norm[28] = {
+ 2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12,
+ 13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5,
+ };
+
+ static const uint8_t dca2wav_wide[28] = {
+ 2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12,
+ 13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5,
+ };
+
+ int dca_ch, wav_ch, nchannels = 0;
+
+ if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
+ for (dca_ch = 0; dca_ch < DCA_SPEAKER_COUNT; dca_ch++)
+ if (dca_mask & (1U << dca_ch))
+ ch_remap[nchannels++] = dca_ch;
+ avctx->channel_layout = dca_mask;
+ } else {
+ int wav_mask = 0;
+ int wav_map[18];
+ const uint8_t *dca2wav;
+ if (dca_mask == DCA_SPEAKER_LAYOUT_7POINT0_WIDE ||
+ dca_mask == DCA_SPEAKER_LAYOUT_7POINT1_WIDE)
+ dca2wav = dca2wav_wide;
+ else
+ dca2wav = dca2wav_norm;
+ for (dca_ch = 0; dca_ch < 28; dca_ch++) {
+ if (dca_mask & (1 << dca_ch)) {
+ wav_ch = dca2wav[dca_ch];
+ if (!(wav_mask & (1 << wav_ch))) {
+ wav_map[wav_ch] = dca_ch;
+ wav_mask |= 1 << wav_ch;
}
}
}
+ for (wav_ch = 0; wav_ch < 18; wav_ch++)
+ if (wav_mask & (1 << wav_ch))
+ ch_remap[nchannels++] = wav_map[wav_ch];
+ avctx->channel_layout = wav_mask;
}
- 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->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->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->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->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->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->audio_header.prim_channels; i++)
- s->audio_header.scalefactor_adj[i][j] = 16;
-
- for (j = 1; j < 11; j++)
- 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 (!xxch) {
- if (s->crc_present) {
- /* Audio header CRC check */
- get_bits(&s->gb, 16);
- }
- } else {
- /* Skip to the end of the header, also ignore CRC if present */
- i = get_bits_count(&s->gb);
- if (hdr_pos + 8 * hdr_size > i)
- skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
- }
-
- s->current_subframe = 0;
- s->current_subsubframe = 0;
-
- return 0;
+ avctx->channels = nchannels;
+ return nchannels;
}
-static int dca_parse_frame_header(DCAContext *s)
+static uint16_t crc16(const uint8_t *data, int size)
{
- init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
-
- /* Sync code */
- skip_bits_long(&s->gb, 32);
-
- /* Frame header */
- s->frame_type = get_bits(&s->gb, 1);
- s->samples_deficit = get_bits(&s->gb, 5) + 1;
- s->crc_present = 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 AVERROR_INVALIDDATA;
- s->amode = get_bits(&s->gb, 6);
- s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
- if (!s->sample_rate)
- return AVERROR_INVALIDDATA;
- s->bit_rate_index = get_bits(&s->gb, 5);
- s->bit_rate = ff_dca_bit_rates[s->bit_rate_index];
- if (!s->bit_rate)
- return AVERROR_INVALIDDATA;
+ static const uint16_t crctab[16] = {
+ 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7,
+ 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef,
+ };
- skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
- s->dynrange = get_bits(&s->gb, 1);
- s->timestamp = get_bits(&s->gb, 1);
- s->aux_data = get_bits(&s->gb, 1);
- s->hdcd = get_bits(&s->gb, 1);
- s->ext_descr = get_bits(&s->gb, 3);
- s->ext_coding = get_bits(&s->gb, 1);
- s->aspf = get_bits(&s->gb, 1);
- s->lfe = get_bits(&s->gb, 2);
- s->predictor_history = get_bits(&s->gb, 1);
-
- if (s->lfe > 2) {
- s->lfe = 0;
- av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
- return AVERROR_INVALIDDATA;
- }
-
- /* TODO: check CRC */
- if (s->crc_present)
- s->header_crc = get_bits(&s->gb, 16);
-
- s->multirate_inter = get_bits(&s->gb, 1);
- s->version = get_bits(&s->gb, 4);
- s->copy_history = get_bits(&s->gb, 2);
- s->source_pcm_res = get_bits(&s->gb, 3);
- s->front_sum = get_bits(&s->gb, 1);
- s->surround_sum = get_bits(&s->gb, 1);
- s->dialog_norm = get_bits(&s->gb, 4);
-
- /* FIXME: channels mixing levels */
- s->output = s->amode;
- if (s->lfe)
- s->output |= DCA_LFE;
+ uint16_t res = 0xffff;
+ int i;
- /* Primary audio coding header */
- s->audio_header.subframes = get_bits(&s->gb, 4) + 1;
+ for (i = 0; i < size; i++) {
+ res = (res << 4) ^ crctab[(data[i] >> 4) ^ (res >> 12)];
+ res = (res << 4) ^ crctab[(data[i] & 15) ^ (res >> 12)];
+ }
- return dca_parse_audio_coding_header(s, 0, 0);
+ return res;
}
-static inline int get_scale(GetBitContext *gb, int level, int value, int log2range)
+int ff_dca_check_crc(GetBitContext *s, int p1, int p2)
{
- if (level < 5) {
- /* huffman encoded */
- value += get_bitalloc(gb, &dca_scalefactor, level);
- value = av_clip(value, 0, (1 << log2range) - 1);
- } else if (level < 8) {
- if (level + 1 > log2range) {
- skip_bits(gb, level + 1 - log2range);
- value = get_bits(gb, log2range);
- } else {
- value = get_bits(gb, level + 1);
- }
- }
- return value;
+ if (((p1 | p2) & 7) || p1 < 0 || p2 > s->size_in_bits || p2 - p1 < 16)
+ return -1;
+ if (crc16(s->buffer + p1 / 8, (p2 - p1) / 8))
+ return -1;
+ return 0;
}
-static int dca_subframe_header(DCAContext *s, int base_channel, int block_index)
+void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples,
+ int *coeff_l, int nsamples, int ch_mask)
{
- /* Primary audio coding side information */
- int j, k;
-
- if (get_bits_left(&s->gb) < 0)
- return AVERROR_INVALIDDATA;
-
- if (!base_channel) {
- s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1;
- if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) {
- s->subsubframes[s->current_subframe] = 1;
- return AVERROR_INVALIDDATA;
- }
- s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
- }
-
- 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->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->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
- }
- }
- }
-
- /* Bit allocation index */
- 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->dca_chan[j].bitalloc[k] =
- get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
- }
-
- if (s->dca_chan[j].bitalloc[k] > 26) {
- ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
- j, k, s->dca_chan[j].bitalloc[k]);
- return AVERROR_INVALIDDATA;
- }
- }
- }
-
- /* Transition mode */
- 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->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]);
- }
- }
- }
-
- if (get_bits_left(&s->gb) < 0)
- return AVERROR_INVALIDDATA;
-
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- const uint32_t *scale_table;
- int scale_sum, log_size;
-
- 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->audio_header.scalefactor_huffman[j] == 6) {
- scale_table = ff_dca_scale_factor_quant7;
- log_size = 7;
- } else {
- scale_table = ff_dca_scale_factor_quant6;
- log_size = 6;
- }
-
- /* When huffman coded, only the difference is encoded */
- scale_sum = 0;
-
- 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->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
- /* Get second scale factor */
- 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->audio_header.prim_channels; j++) {
- /* Transmitted only if joint subband coding enabled */
- 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->audio_header.prim_channels; j++) {
- int source_channel;
+ int pos, spkr, max_spkr = av_log2(ch_mask);
+ int *coeff_r = coeff_l + av_popcount(ch_mask);
- /* Transmitted only if joint subband coding enabled */
- if (s->audio_header.joint_intensity[j] > 0) {
- int scale = 0;
- source_channel = s->audio_header.joint_intensity[j] - 1;
+ av_assert0(DCA_HAS_STEREO(ch_mask));
- /* When huffman coded, only the difference is encoded
- * (is this valid as well for joint scales ???) */
+ // Scale left and right channels
+ pos = (ch_mask & DCA_SPEAKER_MASK_C);
+ dcadsp->dmix_scale(samples[DCA_SPEAKER_L], coeff_l[pos ], nsamples);
+ dcadsp->dmix_scale(samples[DCA_SPEAKER_R], coeff_r[pos + 1], nsamples);
- 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]; */
- }
+ // Downmix remaining channels
+ for (spkr = 0; spkr <= max_spkr; spkr++) {
+ if (!(ch_mask & (1U << spkr)))
+ continue;
- if (!(s->debug_flag & 0x02)) {
- av_log(s->avctx, AV_LOG_DEBUG,
- "Joint stereo coding not supported\n");
- s->debug_flag |= 0x02;
- }
- }
- }
+ if (*coeff_l && spkr != DCA_SPEAKER_L)
+ dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr],
+ *coeff_l, nsamples);
- /* Dynamic range coefficient */
- if (!base_channel && s->dynrange)
- s->dynrange_coef = get_bits(&s->gb, 8);
+ if (*coeff_r && spkr != DCA_SPEAKER_R)
+ dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr],
+ *coeff_r, nsamples);
- /* Side information CRC check word */
- if (s->crc_present) {
- get_bits(&s->gb, 16);
+ coeff_l++;
+ coeff_r++;
}
-
- /*
- * Primary audio data arrays
- */
-
- /* VQ encoded high frequency subbands */
- 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->dca_chan[j].high_freq_vq[k] = get_bits(&s->gb, 10);
-
- /* Low frequency effect data */
- if (!base_channel && s->lfe) {
- int quant7;
- /* LFE samples */
- int lfe_samples = 2 * s->lfe * (4 + block_index);
- int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]);
- float lfe_scale;
-
- for (j = lfe_samples; j < lfe_end_sample; j++) {
- /* Signed 8 bits int */
- s->lfe_data[j] = get_sbits(&s->gb, 8);
- }
-
- /* Scale factor index */
- quant7 = get_bits(&s->gb, 8);
- if (quant7 > 127) {
- avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");
- return AVERROR_INVALIDDATA;
- }
- s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7];
-
- /* Quantization step size * scale factor */
- lfe_scale = 0.035 * s->lfe_scale_factor;
-
- for (j = lfe_samples; j < lfe_end_sample; j++)
- s->lfe_data[j] *= lfe_scale;
- }
-
- return 0;
}
-static void qmf_32_subbands(DCAContext *s, int chans,
- float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
- float scale)
+void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples,
+ int *coeff_l, int nsamples, int ch_mask)
{
- const float *prCoeff;
+ int pos, spkr, max_spkr = av_log2(ch_mask);
+ int *coeff_r = coeff_l + av_popcount(ch_mask);
+ const float scale = 1.0f / (1 << 15);
- int sb_act = s->audio_header.subband_activity[chans];
+ av_assert0(DCA_HAS_STEREO(ch_mask));
- scale *= sqrt(1 / 8.0);
+ // Scale left and right channels
+ pos = (ch_mask & DCA_SPEAKER_MASK_C);
+ fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_L], samples[DCA_SPEAKER_L],
+ coeff_l[pos ] * scale, nsamples);
+ fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_R], samples[DCA_SPEAKER_R],
+ coeff_r[pos + 1] * scale, nsamples);
- /* Select filter */
- if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = ff_dca_fir_32bands_nonperfect;
- else /* Perfect reconstruction */
- prCoeff = ff_dca_fir_32bands_perfect;
+ // Downmix remaining channels
+ for (spkr = 0; spkr <= max_spkr; spkr++) {
+ if (!(ch_mask & (1U << spkr)))
+ continue;
- s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
- 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);
-}
-
-static QMF64_table *qmf64_precompute(void)
-{
- unsigned i, j;
- QMF64_table *table = av_malloc(sizeof(*table));
- if (!table)
- return NULL;
-
- for (i = 0; i < 32; i++)
- for (j = 0; j < 32; j++)
- table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
- for (i = 0; i < 32; i++)
- for (j = 0; j < 32; j++)
- table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
-
- /* FIXME: Is the factor 0.125 = 1/8 right? */
- for (i = 0; i < 32; i++)
- table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
- for (i = 0; i < 32; i++)
- table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
-
- return table;
-}
+ if (*coeff_l && spkr != DCA_SPEAKER_L)
+ fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr],
+ *coeff_l * scale, nsamples);
-/* FIXME: Totally unoptimized. Based on the reference code and
- * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
- * for doubling the size. */
-static void qmf_64_subbands(DCAContext *s, int chans,
- float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
- float *samples_out, float scale)
-{
- float raXin[64];
- float A[32], B[32];
- 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->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
- raXin[i] = 0.0;
- for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
- for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
-
- for (k = 0; k < 32; k++) {
- A[k] = 0.0;
- for (i = 0; i < 32; i++)
- A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
- }
- for (k = 0; k < 32; k++) {
- B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
- for (i = 1; i < 32; i++)
- B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
- }
- for (k = 0; k < 32; k++) {
- raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
- raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
- }
-
- for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
- float out = raZ[i];
- for (j = 0; j < 1024; j += 128)
- out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
- *samples_out++ = out * scale;
- }
-
- for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
- float hist = 0.0;
- for (j = 0; j < 1024; j += 128)
- hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
-
- raZ[i] = hist;
- }
+ if (*coeff_r && spkr != DCA_SPEAKER_R)
+ fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr],
+ *coeff_r * scale, nsamples);
- /* FIXME: Make buffer circular, to avoid this move. */
- memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
+ coeff_l++;
+ coeff_r++;
}
}
-static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
- float *samples_out)
+static int convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size)
{
- /* samples_in: An array holding decimated samples.
- * Samples in current subframe starts from samples_in[0],
- * while samples_in[-1], samples_in[-2], ..., stores samples
- * from last subframe as history.
- *
- * samples_out: An array holding interpolated samples
- */
-
- int idx;
- const float *prCoeff;
- int deciindex;
-
- /* Select decimation filter */
- if (s->lfe == 1) {
- idx = 1;
- prCoeff = ff_dca_lfe_fir_128;
- } else {
- idx = 0;
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
- prCoeff = ff_dca_lfe_xll_fir_64;
- else
- prCoeff = ff_dca_lfe_fir_64;
- }
- /* Interpolation */
- for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
- s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
- samples_in++;
- samples_out += 2 * 32 * (1 + idx);
+ switch (AV_RB32(src)) {
+ case DCA_SYNCWORD_CORE_BE:
+ case DCA_SYNCWORD_SUBSTREAM:
+ memcpy(dst, src, src_size);
+ return src_size;
+ case DCA_SYNCWORD_CORE_LE:
+ case DCA_SYNCWORD_CORE_14B_BE:
+ case DCA_SYNCWORD_CORE_14B_LE:
+ return avpriv_dca_convert_bitstream(src, src_size, dst, max_size);
+ default:
+ return AVERROR_INVALIDDATA;
}
}
-/* downmixing routines */
-#define MIX_REAR1(samples, s1, rs, coef) \
- samples[0][i] += samples[s1][i] * coef[rs][0]; \
- samples[1][i] += samples[s1][i] * coef[rs][1];
-
-#define MIX_REAR2(samples, s1, s2, rs, coef) \
- samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \
- samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1];
-
-#define MIX_FRONT3(samples, coef) \
- t = samples[c][i]; \
- u = samples[l][i]; \
- v = samples[r][i]; \
- samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \
- samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1];
-
-#define DOWNMIX_TO_STEREO(op1, op2) \
- for (i = 0; i < 256; i++) { \
- op1 \
- op2 \
- }
-
-static void dca_downmix(float **samples, int srcfmt, int lfe_present,
- float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
- const int8_t *channel_mapping)
+static int dcadec_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
- int c, l, r, sl, sr, s;
- int i;
- float t, u, v;
+ DCAContext *s = avctx->priv_data;
+ AVFrame *frame = data;
+ uint8_t *input = avpkt->data;
+ int input_size = avpkt->size;
+ int i, ret, prev_packet = s->packet;
- switch (srcfmt) {
- case DCA_MONO:
- case DCA_4F2R:
- av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
- break;
- case DCA_CHANNEL:
- case DCA_STEREO:
- case DCA_STEREO_TOTAL:
- case DCA_STEREO_SUMDIFF:
- break;
- case DCA_3F:
- c = channel_mapping[0];
- l = channel_mapping[1];
- r = channel_mapping[2];
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), );
- break;
- case DCA_2F1R:
- s = channel_mapping[2];
- DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), );
- break;
- case DCA_3F1R:
- c = channel_mapping[0];
- l = channel_mapping[1];
- r = channel_mapping[2];
- s = channel_mapping[3];
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR1(samples, s, 3, coef));
- break;
- case DCA_2F2R:
- sl = channel_mapping[2];
- sr = channel_mapping[3];
- DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), );
- break;
- case DCA_3F2R:
- c = channel_mapping[0];
- l = channel_mapping[1];
- r = channel_mapping[2];
- sl = channel_mapping[3];
- sr = channel_mapping[4];
- DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
- MIX_REAR2(samples, sl, sr, 3, coef));
- break;
- }
- if (lfe_present) {
- int lf_buf = ff_dca_lfe_index[srcfmt];
- int lf_idx = ff_dca_channels[srcfmt];
- for (i = 0; i < 256; i++) {
- samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
- samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
- }
+ if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n");
+ return AVERROR_INVALIDDATA;
}
-}
-#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, int32_t *values)
-{
- int i;
- int offset = (levels - 1) >> 1;
-
- for (i = 0; i < 4; i++) {
- int div = FASTDIV(code, levels);
- values[i] = code - offset - div * levels;
- code = div;
- }
+ av_fast_malloc(&s->buffer, &s->buffer_size,
+ FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_SIZE);
+ if (!s->buffer)
+ return AVERROR(ENOMEM);
- return code;
-}
+ for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++)
+ ret = convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size);
-static int decode_blockcodes(int code1, int code2, int levels, int32_t *values)
-{
- return decode_blockcode(code1, levels, values) |
- decode_blockcode(code2, levels, values + 4);
-}
-#endif
+ if (ret < 0)
+ return ret;
-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 };
+ input = s->buffer;
+ input_size = ret;
-static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
-{
- int k, l;
- int subsubframe = s->current_subsubframe;
- const uint32_t *quant_step_table;
-
- /*
- * Audio data
- */
-
- /* Select quantization step size table */
- if (s->bit_rate_index == 0x1f)
- quant_step_table = ff_dca_lossless_quant;
- else
- quant_step_table = ff_dca_lossy_quant;
-
- for (k = base_channel; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
-
- if (get_bits_left(&s->gb) < 0)
- return AVERROR_INVALIDDATA;
-
- for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
- int m;
-
- /* Select the mid-tread linear quantizer */
- int abits = s->dca_chan[k].bitalloc[l];
-
- uint32_t quant_step_size = quant_step_table[abits];
-
- /*
- * Extract bits from the bit stream
- */
- if (!abits)
- memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
- sizeof(subband_samples[l][0]));
- else {
- uint32_t rscale;
- /* Deal with transients */
- int sfi = s->dca_chan[k].transition_mode[l] &&
- subsubframe >= s->dca_chan[k].transition_mode[l];
- /* Determine quantization index code book and its type.
- Select quantization index code book */
- int sel = s->audio_header.quant_index_huffman[k][abits];
-
- rscale = (s->dca_chan[k].scale_factor[l][sfi] *
- s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
-
- if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) {
- if (abits <= 7) {
- /* Block code */
- 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);
- block_code2 = get_bits(&s->gb, size);
- err = decode_blockcodes(block_code1, block_code2,
- levels, subband_samples[l]);
- 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 < SAMPLES_PER_SUBBAND; m++)
- subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
- }
- } else {
- /* Huffman coded */
- for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
- subband_samples[l][m] = get_bitalloc(&s->gb,
- &dca_smpl_bitalloc[abits], sel);
- }
- s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
- }
- }
+ s->packet = 0;
- for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
- int m;
- /*
- * Inverse ADPCM if in prediction mode
- */
- if (s->dca_chan[k].prediction_mode[l]) {
- int n;
- if (s->predictor_history)
- subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
- (1 << 12) >> 13;
- for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
- int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
- (int64_t)subband_samples[l][m - 1];
- for (n = 2; n <= 4; n++)
- if (m >= n)
- sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
- (int64_t)subband_samples[l][m - n];
- else if (s->predictor_history)
- sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
- subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
- }
- }
+ // Parse backward compatible core sub-stream
+ if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) {
+ int frame_size;
+ if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0) {
+ s->core_residual_valid = 0;
+ return ret;
}
- /* 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->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, s->dca_chan[k].high_freq_vq,
- ff_dca_high_freq_vq,
- subsubframe * SAMPLES_PER_SUBBAND,
- s->dca_chan[k].scale_factor,
- s->audio_header.vq_start_subband[k],
- s->audio_header.subband_activity[k]);
- }
- }
+ s->packet |= DCA_PACKET_CORE;
- /* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
- if (get_bits(&s->gb, 16) != 0xFFFF) {
- av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
- return AVERROR_INVALIDDATA;
+ // EXXS data must be aligned on 4-byte boundary
+ frame_size = FFALIGN(s->core.frame_size, 4);
+ if (input_size - 4 > frame_size) {
+ input += frame_size;
+ input_size -= frame_size;
}
}
- return 0;
-}
-
-static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
-{
- int k;
-
- if (upsample) {
- LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
+ if (!s->core_only) {
+ DCAExssAsset *asset = NULL;
- if (!s->qmf64_table) {
- s->qmf64_table = qmf64_precompute();
- if (!s->qmf64_table)
- return AVERROR(ENOMEM);
- }
-
- /* 64 subbands QMF */
- for (k = 0; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
- s->dca_chan[k].subband_samples[block_index];
-
- s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
- DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
-
- if (s->channel_order_tab[k] >= 0)
- qmf_64_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
- /* Upsampling needs a factor 2 here. */
- M_SQRT2 / 32768.0);
- }
- } else {
- /* 32 subbands QMF */
- LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
-
- for (k = 0; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
- s->dca_chan[k].subband_samples[block_index];
-
- s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
- DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
-
- if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
- M_SQRT1_2 / 32768.0);
- }
- }
-
- /* Generate LFE samples for this subsubframe FIXME!!! */
- if (s->lfe) {
- float *samples = s->samples_chanptr[s->lfe_index];
- lfe_interpolation_fir(s,
- s->lfe_data + 2 * s->lfe * (block_index + 4),
- samples);
- if (upsample) {
- unsigned i;
- /* Should apply the filter in Table 6-11 when upsampling. For
- * now, just duplicate. */
- for (i = 255; i > 0; i--) {
- samples[2 * i] =
- samples[2 * i + 1] = samples[i];
+ // Parse extension sub-stream (EXSS)
+ if (AV_RB32(input) == DCA_SYNCWORD_SUBSTREAM) {
+ if ((ret = ff_dca_exss_parse(&s->exss, input, input_size)) < 0) {
+ if (avctx->err_recognition & AV_EF_EXPLODE)
+ return ret;
+ } else {
+ s->packet |= DCA_PACKET_EXSS;
+ asset = &s->exss.assets[0];
}
- samples[1] = samples[0];
}
- }
-
- /* FIXME: This downmixing is probably broken with upsample.
- * Probably totally broken also with XLL in general. */
- /* Downmixing to Stereo */
- 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);
- }
-
- return 0;
-}
-
-static int dca_subframe_footer(DCAContext *s, int base_channel)
-{
- int in, out, aux_data_count, aux_data_end, reserved;
- uint32_t nsyncaux;
-
- /*
- * Unpack optional information
- */
-
- /* presumably optional information only appears in the core? */
- if (!base_channel) {
- if (s->timestamp)
- skip_bits_long(&s->gb, 32);
-
- if (s->aux_data) {
- aux_data_count = get_bits(&s->gb, 6);
-
- // align (32-bit)
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-
- aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
-
- if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
- av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
- nsyncaux);
- return AVERROR_INVALIDDATA;
- }
-
- if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
- avpriv_request_sample(s->avctx,
- "Auxiliary Decode Time Stamp Flag");
- // align (4-bit)
- skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
- // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
- skip_bits_long(&s->gb, 44);
- }
-
- if ((s->core_downmix = get_bits1(&s->gb))) {
- int am = get_bits(&s->gb, 3);
- switch (am) {
- case 0:
- s->core_downmix_amode = DCA_MONO;
- break;
- case 1:
- s->core_downmix_amode = DCA_STEREO;
- break;
- case 2:
- s->core_downmix_amode = DCA_STEREO_TOTAL;
- break;
- case 3:
- s->core_downmix_amode = DCA_3F;
- break;
- case 4:
- s->core_downmix_amode = DCA_2F1R;
- break;
- case 5:
- s->core_downmix_amode = DCA_2F2R;
- break;
- case 6:
- s->core_downmix_amode = DCA_3F1R;
- break;
- default:
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid mode %d for embedded downmix coefficients\n",
- am);
- return AVERROR_INVALIDDATA;
- }
- for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
- 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,
- "Invalid downmix coefficient code %"PRIu16"\n",
- tmp);
- return AVERROR_INVALIDDATA;
- }
- s->core_downmix_codes[in][out] = tmp;
- }
- }
- }
- align_get_bits(&s->gb); // byte align
- skip_bits(&s->gb, 16); // nAUXCRC16
-
- /*
- * additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
- *
- * Note: don't check for overreads, aux_data_count can't be trusted.
- */
- if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
- avpriv_request_sample(s->avctx,
- "Core auxiliary data reserved content");
- skip_bits_long(&s->gb, reserved);
+ // Parse XLL component in EXSS
+ if (asset && (asset->extension_mask & DCA_EXSS_XLL)) {
+ if ((ret = ff_dca_xll_parse(&s->xll, input, asset)) < 0) {
+ // Conceal XLL synchronization error
+ if (ret == AVERROR(EAGAIN)
+ && (prev_packet & DCA_PACKET_XLL)
+ && (s->packet & DCA_PACKET_CORE))
+ s->packet |= DCA_PACKET_XLL | DCA_PACKET_RECOVERY;
+ else if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
+ return ret;
+ } else {
+ s->packet |= DCA_PACKET_XLL;
}
}
- if (s->crc_present && s->dynrange)
- get_bits(&s->gb, 16);
- }
-
- return 0;
-}
-
-/**
- * Decode a dca frame block
- *
- * @param s pointer to the DCAContext
- */
-
-static int dca_decode_block(DCAContext *s, int base_channel, int block_index)
-{
- int ret;
-
- /* Sanity check */
- if (s->current_subframe >= s->audio_header.subframes) {
- av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
- s->current_subframe, s->audio_header.subframes);
- return AVERROR_INVALIDDATA;
- }
-
- if (!s->current_subsubframe) {
- /* Read subframe header */
- if ((ret = dca_subframe_header(s, base_channel, block_index)))
+ // Parse core extensions in EXSS or backward compatible core sub-stream
+ if ((s->packet & DCA_PACKET_CORE)
+ && (ret = ff_dca_core_parse_exss(&s->core, input, asset)) < 0)
return ret;
}
- /* Read subsubframe */
- if ((ret = dca_subsubframe(s, base_channel, block_index)))
- return ret;
+ // Filter the frame
+ if (s->packet & DCA_PACKET_XLL) {
+ if (s->packet & DCA_PACKET_CORE) {
+ int x96_synth = -1;
- /* Update state */
- s->current_subsubframe++;
- if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) {
- s->current_subsubframe = 0;
- s->current_subframe++;
- }
- if (s->current_subframe >= s->audio_header.subframes) {
- /* Read subframe footer */
- if ((ret = dca_subframe_footer(s, base_channel)))
- return ret;
- }
-
- return 0;
-}
+ // Enable X96 synthesis if needed
+ if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000)
+ x96_synth = 1;
-int ff_dca_xbr_parse_frame(DCAContext *s)
-{
- int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2];
- int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX];
- int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS];
- int anctemp[DCA_CHSET_CHANS_MAX];
- int chset_fsize[DCA_CHSETS_MAX];
- int n_xbr_ch[DCA_CHSETS_MAX];
- int hdr_size, num_chsets, xbr_tmode, hdr_pos;
- int i, j, k, l, chset, chan_base;
-
- av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n");
-
- /* get bit position of sync header */
- hdr_pos = get_bits_count(&s->gb) - 32;
-
- hdr_size = get_bits(&s->gb, 6) + 1;
- num_chsets = get_bits(&s->gb, 2) + 1;
-
- for(i = 0; i < num_chsets; i++)
- chset_fsize[i] = get_bits(&s->gb, 14) + 1;
-
- xbr_tmode = get_bits1(&s->gb);
-
- for(i = 0; i < num_chsets; i++) {
- n_xbr_ch[i] = get_bits(&s->gb, 3) + 1;
- k = get_bits(&s->gb, 2) + 5;
- for(j = 0; j < n_xbr_ch[i]; j++) {
- active_bands[i][j] = get_bits(&s->gb, k) + 1;
- if (active_bands[i][j] > DCA_SUBBANDS) {
- av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]);
- return AVERROR_INVALIDDATA;
- }
- }
- }
-
- /* skip to the end of the header */
- i = get_bits_count(&s->gb);
- if(hdr_pos + hdr_size * 8 > i)
- skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
-
- /* loop over the channel data sets */
- /* only decode as many channels as we've decoded base data for */
- for(chset = 0, chan_base = 0;
- chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels;
- chan_base += n_xbr_ch[chset++]) {
- int start_posn = get_bits_count(&s->gb);
- int subsubframe = 0;
- int subframe = 0;
-
- /* loop over subframes */
- for (k = 0; k < (s->sample_blocks / 8); k++) {
- /* parse header if we're on first subsubframe of a block */
- if(subsubframe == 0) {
- /* Parse subframe header */
- for(i = 0; i < n_xbr_ch[chset]; i++) {
- anctemp[i] = get_bits(&s->gb, 2) + 2;
- }
-
- for(i = 0; i < n_xbr_ch[chset]; i++) {
- get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]);
- }
-
- for(i = 0; i < n_xbr_ch[chset]; i++) {
- anctemp[i] = get_bits(&s->gb, 3);
- if(anctemp[i] < 1) {
- av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n");
- return AVERROR_INVALIDDATA;
- }
- }
-
- /* generate scale factors */
- for(i = 0; i < n_xbr_ch[chset]; i++) {
- const uint32_t *scale_table;
- int nbits;
- int scale_table_size;
-
- if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) {
- scale_table = ff_dca_scale_factor_quant7;
- scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
- } else {
- scale_table = ff_dca_scale_factor_quant6;
- scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
- }
-
- nbits = anctemp[i];
-
- for(j = 0; j < active_bands[chset][i]; j++) {
- if(abits_high[i][j] > 0) {
- int index = get_bits(&s->gb, nbits);
- if (index >= scale_table_size) {
- av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
- return AVERROR_INVALIDDATA;
- }
- scale_table_high[i][j][0] = scale_table[index];
-
- if(xbr_tmode && s->dca_chan[i].transition_mode[j]) {
- int index = get_bits(&s->gb, nbits);
- if (index >= scale_table_size) {
- av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
- return AVERROR_INVALIDDATA;
- }
- scale_table_high[i][j][1] = scale_table[index];
- }
- }
- }
- }
+ if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) {
+ s->core_residual_valid = 0;
+ return ret;
}
- /* decode audio array for this block */
- for(i = 0; i < n_xbr_ch[chset]; i++) {
- for(j = 0; j < active_bands[chset][i]; j++) {
- const int xbr_abits = abits_high[i][j];
- const uint32_t quant_step_size = ff_dca_lossless_quant[xbr_abits];
- const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j];
- const uint32_t rscale = scale_table_high[i][j][sfi];
- int32_t *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j];
- int32_t block[SAMPLES_PER_SUBBAND];
-
- if(xbr_abits <= 0)
- continue;
-
- if(xbr_abits > 7) {
- get_array(&s->gb, block, SAMPLES_PER_SUBBAND, xbr_abits - 3);
- } else {
- int block_code1, block_code2, size, levels, err;
-
- size = abits_sizes[xbr_abits - 1];
- levels = abits_levels[xbr_abits - 1];
-
- block_code1 = get_bits(&s->gb, size);
- block_code2 = get_bits(&s->gb, size);
- err = decode_blockcodes(block_code1, block_code2,
- levels, block);
- if (err) {
- av_log(s->avctx, AV_LOG_ERROR,
- "ERROR: DTS-XBR: block code look-up failed\n");
- return AVERROR_INVALIDDATA;
- }
- }
-
- /* scale & sum into subband */
- s->dcadsp.dequantize(block, quant_step_size, rscale);
- for(l = 0; l < SAMPLES_PER_SUBBAND; l++)
- subband_samples[l] += block[l];
- }
- }
-
- /* check DSYNC marker */
- if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) {
- if(get_bits(&s->gb, 16) != 0xffff) {
- av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n");
- return AVERROR_INVALIDDATA;
- }
- }
-
- /* advance sub-sub-frame index */
- if(++subsubframe >= s->subsubframes[subframe]) {
- subsubframe = 0;
- subframe++;
+ // Force lossy downmixed output on the first core frame filtered.
+ // This prevents audible clicks when seeking and is consistent with
+ // what reference decoder does when there are multiple channel sets.
+ if (!s->core_residual_valid) {
+ if (s->xll.nreschsets > 0 && s->xll.nchsets > 1)
+ s->packet |= DCA_PACKET_RECOVERY;
+ s->core_residual_valid = 1;
}
}
- /* skip to next channel set */
- i = get_bits_count(&s->gb);
- if(start_posn + chset_fsize[chset] * 8 != i) {
- j = start_posn + chset_fsize[chset] * 8 - i;
- if(j < 0 || j >= 8)
- av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set,"
- " skipping further than expected (%d bits)\n", j);
- skip_bits_long(&s->gb, j);
- }
- }
-
- return 0;
-}
-
-
-/* parse initial header for XXCH and dump details */
-int ff_dca_xxch_decode_frame(DCAContext *s)
-{
- int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos;
- int i, chset, base_channel, chstart, fsize[8];
-
- /* assume header word has already been parsed */
- hdr_pos = get_bits_count(&s->gb) - 32;
- hdr_size = get_bits(&s->gb, 6) + 1;
- /*chhdr_crc =*/ skip_bits1(&s->gb);
- spkmsk_bits = get_bits(&s->gb, 5) + 1;
- num_chsets = get_bits(&s->gb, 2) + 1;
-
- for (i = 0; i < num_chsets; i++)
- fsize[i] = get_bits(&s->gb, 14) + 1;
-
- core_spk = get_bits(&s->gb, spkmsk_bits);
- s->xxch_core_spkmask = core_spk;
- s->xxch_nbits_spk_mask = spkmsk_bits;
- s->xxch_dmix_embedded = 0;
-
- /* skip to the end of the header */
- i = get_bits_count(&s->gb);
- if (hdr_pos + hdr_size * 8 > i)
- skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i);
-
- for (chset = 0; chset < num_chsets; chset++) {
- chstart = get_bits_count(&s->gb);
- base_channel = s->audio_header.prim_channels;
- s->xxch_chset = chset;
-
- /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs.
- 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */
- dca_parse_audio_coding_header(s, base_channel, 1);
-
- /* decode channel data */
- for (i = 0; i < (s->sample_blocks / 8); i++) {
- if (dca_decode_block(s, base_channel, i)) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Error decoding DTS-XXCH extension\n");
- continue;
+ if ((ret = ff_dca_xll_filter_frame(&s->xll, frame)) < 0) {
+ // Fall back to core unless hard error
+ if (!(s->packet & DCA_PACKET_CORE))
+ return ret;
+ if (ret != AVERROR_INVALIDDATA || (avctx->err_recognition & AV_EF_EXPLODE))
+ return ret;
+ if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) {
+ s->core_residual_valid = 0;
+ return ret;
}
}
-
- /* skip to end of this section */
- i = get_bits_count(&s->gb);
- if (chstart + fsize[chset] * 8 > i)
- skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i);
- }
- s->xxch_chset = num_chsets;
-
- return 0;
-}
-
-static float dca_dmix_code(unsigned code)
-{
- int sign = (code >> 8) - 1;
- code &= 0xff;
- return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));
-}
-
-static int scan_for_extensions(AVCodecContext *avctx)
-{
- DCAContext *s = avctx->priv_data;
- int core_ss_end, ret = 0;
-
- core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8;
-
- /* only scan for extensions if ext_descr was unknown or indicated a
- * supported XCh extension */
- if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) {
- /* if ext_descr was unknown, clear s->core_ext_mask so that the
- * extensions scan can fill it up */
- s->core_ext_mask = FFMAX(s->core_ext_mask, 0);
-
- /* extensions start at 32-bit boundaries into bitstream */
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-
- while (core_ss_end - get_bits_count(&s->gb) >= 32) {
- uint32_t bits = get_bits_long(&s->gb, 32);
- int i;
-
- switch (bits) {
- case DCA_SYNCWORD_XCH: {
- int ext_amode, xch_fsize;
-
- s->xch_base_channel = s->audio_header.prim_channels;
-
- /* validate sync word using XCHFSIZE field */
- xch_fsize = show_bits(&s->gb, 10);
- if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) &&
- (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1))
- continue;
-
- /* skip length-to-end-of-frame field for the moment */
- skip_bits(&s->gb, 10);
-
- s->core_ext_mask |= DCA_EXT_XCH;
-
- /* extension amode(number of channels in extension) should be 1 */
- /* AFAIK XCh is not used for more channels */
- if ((ext_amode = get_bits(&s->gb, 4)) != 1) {
- av_log(avctx, AV_LOG_ERROR,
- "XCh extension amode %d not supported!\n",
- ext_amode);
- continue;
- }
-
- if (s->xch_base_channel < 2) {
- avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");
- continue;
- }
-
- /* much like core primary audio coding header */
- dca_parse_audio_coding_header(s, s->xch_base_channel, 0);
-
- for (i = 0; i < (s->sample_blocks / 8); 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;
- break;
- }
- case DCA_SYNCWORD_XXCH:
- /* XXCh: extended channels */
- /* usually found either in core or HD part in DTS-HD HRA streams,
- * but not in DTS-ES which contains XCh extensions instead */
- s->core_ext_mask |= DCA_EXT_XXCH;
- ff_dca_xxch_decode_frame(s);
- break;
-
- case 0x1d95f262: {
- int fsize96 = show_bits(&s->gb, 12) + 1;
- if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96)
- continue;
-
- av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n",
- get_bits_count(&s->gb));
- skip_bits(&s->gb, 12);
- av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96);
- av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4));
-
- s->core_ext_mask |= DCA_EXT_X96;
- break;
- }
- }
-
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
+ } else if (s->packet & DCA_PACKET_CORE) {
+ if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) {
+ s->core_residual_valid = 0;
+ return ret;
}
+ s->core_residual_valid = !!(s->core.filter_mode & DCA_FILTER_MODE_FIXED);
} else {
- /* no supported extensions, skip the rest of the core substream */
- skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb));
+ return AVERROR_INVALIDDATA;
}
- if (s->core_ext_mask & DCA_EXT_X96)
- s->profile = FF_PROFILE_DTS_96_24;
- else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH))
- s->profile = FF_PROFILE_DTS_ES;
-
- /* check for ExSS (HD part) */
- if (s->dca_buffer_size - s->frame_size > 32 &&
- get_bits_long(&s->gb, 32) == DCA_SYNCWORD_SUBSTREAM)
- ff_dca_exss_parse_header(s);
+ *got_frame_ptr = 1;
- return ret;
+ return avpkt->size;
}
-static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels)
+static av_cold void dcadec_flush(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
- int i, j, chset, mask;
- int channel_layout, channel_mask;
- int posn, lavc;
-
- /* If we have XXCH then the channel layout is managed differently */
- /* note that XLL will also have another way to do things */
- if (!(s->core_ext_mask & DCA_EXT_XXCH)) {
- /* xxx should also do MA extensions */
- if (s->amode < 16) {
- avctx->channel_layout = ff_dca_core_channel_layout[s->amode];
-
- 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
- * downmix coefficients for the additional channel coded in the XCh
- * extension, so when we're doing a Stereo downmix, don't decode it.
- */
- s->xch_disable = 1;
- }
-
- if (s->xch_present && !s->xch_disable) {
- if (avctx->channel_layout & AV_CH_BACK_CENTER) {
- avpriv_request_sample(avctx, "XCh with Back center channel");
- return AVERROR_INVALIDDATA;
- }
- avctx->channel_layout |= AV_CH_BACK_CENTER;
- if (s->lfe) {
- avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
- s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
- } else {
- s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode];
- }
- if (s->channel_order_tab[s->xch_base_channel] < 0)
- return AVERROR_INVALIDDATA;
- } else {
- *channels = num_core_channels + !!s->lfe;
- s->xch_present = 0; /* disable further xch processing */
- if (s->lfe) {
- avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
- s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode];
- } else
- s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode];
- }
-
- if (*channels > !!s->lfe &&
- s->channel_order_tab[*channels - 1 - !!s->lfe] < 0)
- return AVERROR_INVALIDDATA;
-
- if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) {
- av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout));
- return AVERROR_INVALIDDATA;
- }
-
- if (num_core_channels + !!s->lfe > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- *channels = 2;
- s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO;
- avctx->channel_layout = AV_CH_LAYOUT_STEREO;
- }
- else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
- static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
- s->channel_order_tab = dca_channel_order_native;
- }
- s->lfe_index = ff_dca_lfe_index[s->amode];
- } else {
- av_log(avctx, AV_LOG_ERROR,
- "Non standard configuration %d !\n", s->amode);
- return AVERROR_INVALIDDATA;
- }
-
- s->xxch_dmix_embedded = 0;
- } else {
- /* we only get here if an XXCH channel set can be added to the mix */
- channel_mask = s->xxch_core_spkmask;
-
- {
- *channels = s->audio_header.prim_channels + !!s->lfe;
- for (i = 0; i < s->xxch_chset; i++) {
- channel_mask |= s->xxch_spk_masks[i];
- }
- }
-
- /* Given the DTS spec'ed channel mask, generate an avcodec version */
- channel_layout = 0;
- for (i = 0; i < s->xxch_nbits_spk_mask; ++i) {
- if (channel_mask & (1 << i)) {
- channel_layout |= ff_dca_map_xxch_to_native[i];
- }
- }
-
- /* make sure that we have managed to get equivalent dts/avcodec channel
- * masks in some sense -- unfortunately some channels could overlap */
- if (av_popcount(channel_mask) != av_popcount(channel_layout)) {
- av_log(avctx, AV_LOG_DEBUG,
- "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n");
- return AVERROR_INVALIDDATA;
- }
- avctx->channel_layout = channel_layout;
-
- if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) {
- /* Estimate DTS --> avcodec ordering table */
- for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) {
- mask = chset >= 0 ? s->xxch_spk_masks[chset]
- : s->xxch_core_spkmask;
- for (i = 0; i < s->xxch_nbits_spk_mask; i++) {
- if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) {
- lavc = ff_dca_map_xxch_to_native[i];
- posn = av_popcount(channel_layout & (lavc - 1));
- s->xxch_order_tab[j++] = posn;
- }
- }
-
- }
-
- s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1));
- } else { /* native ordering */
- for (i = 0; i < *channels; i++)
- s->xxch_order_tab[i] = i;
-
- s->lfe_index = *channels - 1;
- }
+ ff_dca_core_flush(&s->core);
+ ff_dca_xll_flush(&s->xll);
- s->channel_order_tab = s->xxch_order_tab;
- }
-
- return 0;
+ s->core_residual_valid = 0;
}
-/**
- * Main frame decoding function
- * FIXME add arguments
- */
-static int dca_decode_frame(AVCodecContext *avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
+static av_cold int dcadec_close(AVCodecContext *avctx)
{
- AVFrame *frame = data;
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
- int lfe_samples;
- int num_core_channels = 0;
- int i, ret;
- float **samples_flt;
- float *src_chan;
- float *dst_chan;
DCAContext *s = avctx->priv_data;
- int channels, full_channels;
- float scale;
- int achan;
- int chset;
- int mask;
- int j, k;
- int endch;
- int upsample = 0;
-
- s->exss_ext_mask = 0;
- s->xch_present = 0;
-
- s->dca_buffer_size = AVERROR_INVALIDDATA;
- for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++)
- s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, s->dca_buffer,
- DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE);
-
- if (s->dca_buffer_size == AVERROR_INVALIDDATA) {
- av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
- return AVERROR_INVALIDDATA;
- }
-
- if ((ret = dca_parse_frame_header(s)) < 0) {
- // seems like the frame is corrupt, try with the next one
- return ret;
- }
- // set AVCodec values with parsed data
- avctx->sample_rate = s->sample_rate;
-
- s->profile = FF_PROFILE_DTS;
-
- for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); 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 */
- num_core_channels = s->audio_header.prim_channels;
-
- if (s->audio_header.prim_channels + !!s->lfe > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- /* Stereo downmix coefficients
- *
- * The decoder can only downmix to 2-channel, so we need to ensure
- * embedded downmix coefficients are actually targeting 2-channel.
- */
- if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
- s->core_downmix_amode == DCA_STEREO_TOTAL)) {
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- /* Range checked earlier */
- s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
- s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
- }
- s->output = s->core_downmix_amode;
- } else {
- int am = s->amode & DCA_CHANNEL_MASK;
- if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid channel mode %d\n", am);
- return AVERROR_INVALIDDATA;
- }
- if (num_core_channels + !!s->lfe >
- FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
- avpriv_request_sample(s->avctx, "Downmixing %d channels",
- s->audio_header.prim_channels + !!s->lfe);
- return AVERROR_PATCHWELCOME;
- }
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
- s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
- }
- }
- ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
- s->downmix_coef[i][0]);
- ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
- s->downmix_coef[i][1]);
- }
- ff_dlog(s->avctx, "\n");
- }
-
- if (s->ext_coding)
- s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr];
- else
- s->core_ext_mask = 0;
-
- ret = scan_for_extensions(avctx);
- avctx->profile = s->profile;
+ ff_dca_core_close(&s->core);
+ ff_dca_xll_close(&s->xll);
- full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
+ av_freep(&s->buffer);
+ s->buffer_size = 0;
- ret = set_channel_layout(avctx, &channels, num_core_channels);
- if (ret < 0)
- return ret;
-
- /* get output buffer */
- frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
- int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
- /* Check for invalid/unsupported conditions first */
- if (s->xll_residual_channels > channels) {
- av_log(s->avctx, AV_LOG_WARNING,
- "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
- s->xll_residual_channels, channels);
- s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
- } else if (xll_nb_samples != frame->nb_samples &&
- 2 * frame->nb_samples != xll_nb_samples) {
- av_log(s->avctx, AV_LOG_WARNING,
- "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
- xll_nb_samples, frame->nb_samples);
- s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
- } else {
- if (2 * frame->nb_samples == xll_nb_samples) {
- av_log(s->avctx, AV_LOG_INFO,
- "XLL: upsampling core channels by a factor of 2\n");
- upsample = 1;
-
- frame->nb_samples = xll_nb_samples;
- // FIXME: Is it good enough to copy from the first channel set?
- avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
- }
- /* If downmixing to stereo, don't decode additional channels.
- * FIXME: Using the xch_disable flag for this doesn't seem right. */
- if (!s->xch_disable)
- channels = s->xll_channels;
- }
- }
-
- if (avctx->channels != channels) {
- if (avctx->channels)
- av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels);
- avctx->channels = channels;
- }
-
- /* FIXME: This is an ugly hack, to just revert to the default
- * layout if we have additional channels. Need to convert the XLL
- * channel masks to ffmpeg channel_layout mask. */
- if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
- avctx->channel_layout = 0;
-
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
- return ret;
- samples_flt = (float **) frame->extended_data;
-
- /* allocate buffer for extra channels if downmixing */
- if (avctx->channels < full_channels) {
- ret = av_samples_get_buffer_size(NULL, full_channels - channels,
- frame->nb_samples,
- avctx->sample_fmt, 0);
- if (ret < 0)
- return ret;
-
- av_fast_malloc(&s->extra_channels_buffer,
- &s->extra_channels_buffer_size, ret);
- if (!s->extra_channels_buffer)
- return AVERROR(ENOMEM);
-
- ret = av_samples_fill_arrays((uint8_t **) s->extra_channels, NULL,
- s->extra_channels_buffer,
- full_channels - channels,
- frame->nb_samples, avctx->sample_fmt, 0);
- if (ret < 0)
- return ret;
- }
-
- /* filter to get final output */
- for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); i++) {
- int ch;
- unsigned block = upsample ? 512 : 256;
- for (ch = 0; ch < channels; ch++)
- s->samples_chanptr[ch] = samples_flt[ch] + i * block;
- for (; ch < full_channels; ch++)
- s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
-
- dca_filter_channels(s, i, upsample);
-
- /* If this was marked as a DTS-ES stream we need to subtract back- */
- /* channel from SL & SR to remove matrixed back-channel signal */
- if ((s->source_pcm_res & 1) && s->xch_present) {
- float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]];
- float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]];
- float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]];
- s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256);
- s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256);
- }
-
- /* If stream contains XXCH, we might need to undo an embedded downmix */
- if (s->xxch_dmix_embedded) {
- /* Loop over channel sets in turn */
- ch = num_core_channels;
- for (chset = 0; chset < s->xxch_chset; chset++) {
- endch = ch + s->xxch_chset_nch[chset];
- mask = s->xxch_dmix_embedded;
-
- /* undo downmix */
- for (j = ch; j < endch; j++) {
- if (mask & (1 << j)) { /* this channel has been mixed-out */
- src_chan = s->samples_chanptr[s->channel_order_tab[j]];
- for (k = 0; k < endch; k++) {
- achan = s->channel_order_tab[k];
- scale = s->xxch_dmix_coeff[j][k];
- if (scale != 0.0) {
- dst_chan = s->samples_chanptr[achan];
- s->fdsp->vector_fmac_scalar(dst_chan, src_chan,
- -scale, 256);
- }
- }
- }
- }
-
- /* if a downmix has been embedded then undo the pre-scaling */
- if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) {
- scale = s->xxch_dmix_sf[chset];
-
- for (j = 0; j < ch; j++) {
- src_chan = s->samples_chanptr[s->channel_order_tab[j]];
- for (k = 0; k < 256; k++)
- src_chan[k] *= scale;
- }
-
- /* LFE channel is always part of core, scale if it exists */
- if (s->lfe) {
- src_chan = s->samples_chanptr[s->lfe_index];
- for (k = 0; k < 256; k++)
- src_chan[k] *= scale;
- }
- }
-
- ch = endch;
- }
-
- }
- }
-
- /* update lfe history */
- lfe_samples = 2 * s->lfe * (s->sample_blocks / SAMPLES_PER_SUBBAND);
- for (i = 0; i < 2 * s->lfe * 4; i++)
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
-
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
- ret = ff_dca_xll_decode_audio(s, frame);
- if (ret < 0)
- return ret;
- }
- /* AVMatrixEncoding
- *
- * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
- ret = ff_side_data_update_matrix_encoding(frame,
- (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
- AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
- if (ret < 0)
- return ret;
-
- if ( avctx->profile != FF_PROFILE_DTS_HD_MA
- && avctx->profile != FF_PROFILE_DTS_HD_HRA)
- avctx->bit_rate = s->bit_rate;
- *got_frame_ptr = 1;
-
- return buf_size;
+ return 0;
}
-/**
- * DCA initialization
- *
- * @param avctx pointer to the AVCodecContext
- */
-
-static av_cold int dca_decode_init(AVCodecContext *avctx)
+static av_cold int dcadec_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
s->avctx = avctx;
- dca_init_vlcs();
+ s->core.avctx = avctx;
+ s->exss.avctx = avctx;
+ s->xll.avctx = avctx;
- s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
- if (!s->fdsp)
+ if (ff_dca_core_init(&s->core) < 0)
return AVERROR(ENOMEM);
- ff_mdct_init(&s->imdct, 6, 1, 1.0);
- ff_synth_filter_init(&s->synth);
ff_dcadsp_init(&s->dcadsp);
- ff_fmt_convert_init(&s->fmt_conv, avctx);
+ s->core.dcadsp = &s->dcadsp;
+ s->xll.dcadsp = &s->dcadsp;
- avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ switch (avctx->request_channel_layout & ~AV_CH_LAYOUT_NATIVE) {
+ case 0:
+ s->request_channel_layout = 0;
+ break;
+ case AV_CH_LAYOUT_STEREO:
+ case AV_CH_LAYOUT_STEREO_DOWNMIX:
+ s->request_channel_layout = DCA_SPEAKER_LAYOUT_STEREO;
+ break;
+ case AV_CH_LAYOUT_5POINT0:
+ s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT0;
+ break;
+ case AV_CH_LAYOUT_5POINT1:
+ s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT1;
+ break;
+ default:
+ av_log(avctx, AV_LOG_WARNING, "Invalid request_channel_layout\n");
+ break;
+ }
- /* allow downmixing to stereo */
- if (avctx->channels > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
- avctx->channels = 2;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ avctx->bits_per_raw_sample = 24;
return 0;
}
-static av_cold int dca_decode_end(AVCodecContext *avctx)
-{
- DCAContext *s = avctx->priv_data;
- ff_mdct_end(&s->imdct);
- av_freep(&s->extra_channels_buffer);
- av_freep(&s->fdsp);
- av_freep(&s->xll_sample_buf);
- av_freep(&s->qmf64_table);
- return 0;
-}
+#define OFFSET(x) offsetof(DCAContext, x)
+#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
-static const AVOption options[] = {
- { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
- { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
- { NULL },
+static const AVOption dcadec_options[] = {
+ { "core_only", "Decode core only without extensions", OFFSET(core_only), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, PARAM },
+ { NULL }
};
-static const AVClass dca_decoder_class = {
+static const AVClass dcadec_class = {
.class_name = "DCA decoder",
.item_name = av_default_item_name,
- .option = options,
+ .option = dcadec_options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DECODER,
};
AVCodec ff_dca_decoder = {
- .name = "dca",
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAContext),
- .init = dca_decode_init,
- .decode = dca_decode_frame,
- .close = dca_decode_end,
- .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
- .priv_class = &dca_decoder_class,
+ .name = "dca",
+ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_DTS,
+ .priv_data_size = sizeof(DCAContext),
+ .init = dcadec_init,
+ .decode = dcadec_decode_frame,
+ .close = dcadec_close,
+ .flush = dcadec_flush,
+ .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
+ .priv_class = &dcadec_class,
+ .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
+ .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};
projects / ffmpeg / blobdiff