[FFmpeg-devel] [PATCH v2 13/16] avcodec/dca: add core decoder
foo86
foobaz86 at gmail.com
Thu Jan 21 19:49:02 CET 2016
---
libavcodec/dca_core.c | 2602 +++++++++++++++++++++++++++++++++++++++++++++++++
libavcodec/dca_core.h | 206 ++++
2 files changed, 2808 insertions(+)
create mode 100644 libavcodec/dca_core.c
create mode 100644 libavcodec/dca_core.h
diff --git a/libavcodec/dca_core.c b/libavcodec/dca_core.c
new file mode 100644
index 0000000..61f7ff3
--- /dev/null
+++ b/libavcodec/dca_core.c
@@ -0,0 +1,2602 @@
+/*
+ * Copyright (C) 2016 foo86
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "dcadec.h"
+#include "dcadata.h"
+#include "dcahuff.h"
+#include "dcamath.h"
+#include "dca_syncwords.h"
+
+#if ARCH_ARM
+#include "arm/dca.h"
+#endif
+
+enum HeaderType {
+ HEADER_CORE,
+ HEADER_XCH,
+ HEADER_XXCH
+};
+
+enum AudioMode {
+ AMODE_MONO, // Mode 0: A (mono)
+ AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
+ AMODE_STEREO, // Mode 2: L + R (stereo)
+ AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
+ AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
+ AMODE_3F, // Mode 5: C + L + R
+ AMODE_2F1R, // Mode 6: L + R + S
+ AMODE_3F1R, // Mode 7: C + L + R + S
+ AMODE_2F2R, // Mode 8: L + R + SL + SR
+ AMODE_3F2R, // Mode 9: C + L + R + SL + SR
+
+ AMODE_COUNT
+};
+
+enum ExtAudioType {
+ EXT_AUDIO_XCH = 0,
+ EXT_AUDIO_X96 = 2,
+ EXT_AUDIO_XXCH = 6
+};
+
+enum LFEFlag {
+ LFE_FLAG_NONE,
+ LFE_FLAG_128,
+ LFE_FLAG_64,
+ LFE_FLAG_INVALID
+};
+
+static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
+ { DCA_SPEAKER_C, -1, -1, -1, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
+ { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , -1, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs, -1, -1 },
+ { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs, -1 },
+ { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs, -1 },
+ { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
+};
+
+static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = {
+ DCA_SPEAKER_LAYOUT_MONO,
+ DCA_SPEAKER_LAYOUT_STEREO,
+ DCA_SPEAKER_LAYOUT_STEREO,
+ DCA_SPEAKER_LAYOUT_STEREO,
+ DCA_SPEAKER_LAYOUT_STEREO,
+ DCA_SPEAKER_LAYOUT_3_0,
+ DCA_SPEAKER_LAYOUT_2_1,
+ DCA_SPEAKER_LAYOUT_3_1,
+ DCA_SPEAKER_LAYOUT_2_2,
+ DCA_SPEAKER_LAYOUT_5POINT0
+};
+
+static const uint8_t block_code_nbits[7] = {
+ 7, 10, 12, 13, 15, 17, 19
+};
+
+static const uint8_t quant_index_sel_nbits[DCA_CODE_BOOKS] = {
+ 1, 2, 2, 2, 2, 3, 3, 3, 3, 3
+};
+
+static const uint8_t quant_index_group_size[DCA_CODE_BOOKS] = {
+ 1, 3, 3, 3, 3, 7, 7, 7, 7, 7
+};
+
+typedef struct DCAVLC {
+ int offset; ///< Code values offset
+ int max_depth; ///< Parameter for get_vlc2()
+ VLC vlc[7]; ///< Actual codes
+} DCAVLC;
+
+static DCAVLC vlc_bit_allocation;
+static DCAVLC vlc_transition_mode;
+static DCAVLC vlc_scale_factor;
+static DCAVLC vlc_quant_index[DCA_CODE_BOOKS];
+
+static av_cold void dca_init_vlcs(void)
+{
+ static VLC_TYPE dca_table[23622][2];
+ static int vlcs_initialized = 0;
+ int i, j, k;
+
+ if (vlcs_initialized)
+ return;
+
+#define DCA_INIT_VLC(vlc, a, b, c, d) \
+ do { \
+ vlc.table = &dca_table[ff_dca_vlc_offs[k]]; \
+ vlc.table_allocated = ff_dca_vlc_offs[k + 1] - ff_dca_vlc_offs[k]; \
+ init_vlc(&vlc, a, b, c, 1, 1, d, 2, 2, INIT_VLC_USE_NEW_STATIC); \
+ } while (0)
+
+ vlc_bit_allocation.offset = 1;
+ vlc_bit_allocation.max_depth = 2;
+ for (i = 0, k = 0; i < 5; i++, k++)
+ DCA_INIT_VLC(vlc_bit_allocation.vlc[i], bitalloc_12_vlc_bits[i], 12,
+ bitalloc_12_bits[i], bitalloc_12_codes[i]);
+
+ vlc_scale_factor.offset = -64;
+ vlc_scale_factor.max_depth = 2;
+ for (i = 0; i < 5; i++, k++)
+ DCA_INIT_VLC(vlc_scale_factor.vlc[i], SCALES_VLC_BITS, 129,
+ scales_bits[i], scales_codes[i]);
+
+ vlc_transition_mode.offset = 0;
+ vlc_transition_mode.max_depth = 1;
+ for (i = 0; i < 4; i++, k++)
+ DCA_INIT_VLC(vlc_transition_mode.vlc[i], tmode_vlc_bits[i], 4,
+ tmode_bits[i], tmode_codes[i]);
+
+ for (i = 0; i < DCA_CODE_BOOKS; i++) {
+ vlc_quant_index[i].offset = bitalloc_offsets[i];
+ vlc_quant_index[i].max_depth = 1 + (i > 4);
+ for (j = 0; j < quant_index_group_size[i]; j++, k++)
+ DCA_INIT_VLC(vlc_quant_index[i].vlc[j], bitalloc_maxbits[i][j],
+ bitalloc_sizes[i], bitalloc_bits[i][j], bitalloc_codes[i][j]);
+ }
+
+ vlcs_initialized = 1;
+}
+
+static int get_vlc(GetBitContext *s, DCAVLC *v, int i)
+{
+ return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
+}
+
+static void get_array(GetBitContext *s, int32_t *array, int size, int n)
+{
+ int i;
+
+ for (i = 0; i < size; i++)
+ array[i] = get_sbits(s, n);
+}
+
+// 5.3.1 - Bit stream header
+static int parse_frame_header(DCACoreDecoder *s)
+{
+ int normal_frame, pcmr_index;
+
+ // Frame type
+ normal_frame = get_bits1(&s->gb);
+
+ // Deficit sample count
+ if (get_bits(&s->gb, 5) != DCA_PCMBLOCK_SAMPLES - 1) {
+ av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n");
+ return normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
+ }
+
+ // CRC present flag
+ s->crc_present = get_bits1(&s->gb);
+
+ // Number of PCM sample blocks
+ s->npcmblocks = get_bits(&s->gb, 7) + 1;
+ if (s->npcmblocks & (DCA_SUBBAND_SAMPLES - 1)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", s->npcmblocks);
+ return (s->npcmblocks < 6 || normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
+ }
+
+ // Primary frame byte size
+ s->frame_size = get_bits(&s->gb, 14) + 1;
+ if (s->frame_size < 96) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", s->frame_size);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Audio channel arrangement
+ s->audio_mode = get_bits(&s->gb, 6);
+ if (s->audio_mode >= AMODE_COUNT) {
+ av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Core audio sampling frequency
+ s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
+ if (!s->sample_rate) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Transmission bit rate
+ s->bit_rate = ff_dca_bit_rates[get_bits(&s->gb, 5)];
+
+ // Reserved field
+ skip_bits1(&s->gb);
+
+ // Embedded dynamic range flag
+ s->drc_present = get_bits1(&s->gb);
+
+ // Embedded time stamp flag
+ s->ts_present = get_bits1(&s->gb);
+
+ // Auxiliary data flag
+ s->aux_present = get_bits1(&s->gb);
+
+ // HDCD mastering flag
+ skip_bits1(&s->gb);
+
+ // Extension audio descriptor flag
+ s->ext_audio_type = get_bits(&s->gb, 3);
+
+ // Extended coding flag
+ s->ext_audio_present = get_bits1(&s->gb);
+
+ // Audio sync word insertion flag
+ s->sync_ssf = get_bits1(&s->gb);
+
+ // Low frequency effects flag
+ s->lfe_present = get_bits(&s->gb, 2);
+ if (s->lfe_present == LFE_FLAG_INVALID) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Predictor history flag switch
+ s->predictor_history = get_bits1(&s->gb);
+
+ // Header CRC check bytes
+ if (s->crc_present)
+ skip_bits(&s->gb, 16);
+
+ // Multirate interpolator switch
+ s->filter_perfect = get_bits1(&s->gb);
+
+ // Encoder software revision
+ skip_bits(&s->gb, 4);
+
+ // Copy history
+ skip_bits(&s->gb, 2);
+
+ // Source PCM resolution
+ s->source_pcm_res = ff_dca_bits_per_sample[pcmr_index = get_bits(&s->gb, 3)];
+ if (!s->source_pcm_res) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->es_format = pcmr_index & 1;
+
+ // Front sum/difference flag
+ s->sumdiff_front = get_bits1(&s->gb);
+
+ // Surround sum/difference flag
+ s->sumdiff_surround = get_bits1(&s->gb);
+
+ // Dialog normalization / unspecified
+ skip_bits(&s->gb, 4);
+
+ return 0;
+}
+
+// 5.3.2 - Primary audio coding header
+static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base)
+{
+ int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb);
+ unsigned int mask, index;
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ switch (header) {
+ case HEADER_CORE:
+ // Number of subframes
+ s->nsubframes = get_bits(&s->gb, 4) + 1;
+
+ // Number of primary audio channels
+ s->nchannels = get_bits(&s->gb, 3) + 1;
+ if (s->nchannels != ff_dca_channels[s->audio_mode]) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode);
+ return AVERROR_INVALIDDATA;
+ }
+ av_assert1(s->nchannels <= DCA_CHANNELS - 2);
+
+ s->ch_mask = audio_mode_ch_mask[s->audio_mode];
+
+ // Add LFE channel if present
+ if (s->lfe_present)
+ s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
+ break;
+
+ case HEADER_XCH:
+ s->nchannels = ff_dca_channels[s->audio_mode] + 1;
+ av_assert1(s->nchannels <= DCA_CHANNELS - 1);
+ s->ch_mask |= DCA_SPEAKER_MASK_Cs;
+ break;
+
+ case HEADER_XXCH:
+ // Channel set header length
+ header_size = get_bits(&s->gb, 7) + 1;
+
+ // Check CRC
+ if (s->xxch_crc_present
+ && (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of channels in a channel set
+ nchannels = get_bits(&s->gb, 3) + 1;
+ if (nchannels > DCA_XXCH_CHANNELS_MAX) {
+ avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels);
+ return AVERROR_PATCHWELCOME;
+ }
+ s->nchannels = ff_dca_channels[s->audio_mode] + nchannels;
+ av_assert1(s->nchannels <= DCA_CHANNELS);
+
+ // Loudspeaker layout mask
+ mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs);
+ s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs;
+
+ if (av_popcount(s->xxch_spkr_mask) != nchannels) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask);
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (s->xxch_core_mask & s->xxch_spkr_mask) {
+ av_log(s->avctx, AV_LOG_ERROR, "XXCH speaker layout mask (%#x) overlaps with core (%#x)\n", s->xxch_spkr_mask, s->xxch_core_mask);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Combine core and XXCH masks together
+ s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask;
+
+ // Downmix coefficients present in stream
+ if (get_bits1(&s->gb)) {
+ int *coeff_ptr = s->xxch_dmix_coeff;
+
+ // Downmix already performed by encoder
+ s->xxch_dmix_embedded = get_bits1(&s->gb);
+
+ // Downmix scale factor
+ index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3;
+ if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index);
+ return AVERROR_INVALIDDATA;
+ }
+ s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index];
+
+ // Downmix channel mapping mask
+ for (ch = 0; ch < nchannels; ch++) {
+ mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
+ if ((mask & s->xxch_core_mask) != mask) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask);
+ return AVERROR_INVALIDDATA;
+ }
+ s->xxch_dmix_mask[ch] = mask;
+ }
+
+ // Downmix coefficients
+ for (ch = 0; ch < nchannels; ch++) {
+ for (n = 0; n < s->xxch_mask_nbits; n++) {
+ if (s->xxch_dmix_mask[ch] & (1U << n)) {
+ int code = get_bits(&s->gb, 7);
+ int sign = (code >> 6) - 1;
+ if (code &= 63) {
+ index = code * 4 - 3;
+ if (index >= FF_DCA_DMIXTABLE_SIZE) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index);
+ return AVERROR_INVALIDDATA;
+ }
+ *coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign;
+ } else {
+ *coeff_ptr++ = 0;
+ }
+ }
+ }
+ }
+ } else {
+ s->xxch_dmix_embedded = 0;
+ }
+
+ break;
+ }
+
+ // Subband activity count
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ s->nsubbands[ch] = get_bits(&s->gb, 5) + 2;
+ if (s->nsubbands[ch] > DCA_SUBBANDS) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // High frequency VQ start subband
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1;
+
+ // Joint intensity coding index
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH)
+ n += xch_base - 1;
+ if (n > s->nchannels) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->joint_intensity_index[ch] = n;
+ }
+
+ // Transient mode code book
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ s->transition_mode_sel[ch] = get_bits(&s->gb, 2);
+
+ // Scale factor code book
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
+ if (s->scale_factor_sel[ch] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Bit allocation quantizer select
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
+ if (s->bit_allocation_sel[ch] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Quantization index codebook select
+ for (n = 0; n < DCA_CODE_BOOKS; n++)
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
+
+ // Scale factor adjustment index
+ for (n = 0; n < DCA_CODE_BOOKS; n++)
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ if (s->quant_index_sel[ch][n] < quant_index_group_size[n])
+ s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];
+
+ if (header == HEADER_XXCH) {
+ // Reserved
+ // Byte align
+ // CRC16 of channel set header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n");
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+ // Audio header CRC check word
+ if (s->crc_present)
+ skip_bits(&s->gb, 16);
+ }
+
+ return 0;
+}
+
+static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel)
+{
+ const uint32_t *scale_table;
+ unsigned int scale_size;
+
+ // Select the root square table
+ if (sel > 5) {
+ scale_table = ff_dca_scale_factor_quant7;
+ scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
+ } else {
+ scale_table = ff_dca_scale_factor_quant6;
+ scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
+ }
+
+ // If Huffman code was used, the difference of scales was encoded
+ if (sel < 5)
+ *scale_index += get_vlc(&s->gb, &vlc_scale_factor, sel);
+ else
+ *scale_index = get_bits(&s->gb, sel + 1);
+
+ // Look up scale factor from the root square table
+ if ((unsigned int)*scale_index >= scale_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return scale_table[*scale_index];
+}
+
+static inline int parse_joint_scale(DCACoreDecoder *s, int sel)
+{
+ int scale_index;
+
+ // Absolute value was encoded even when Huffman code was used
+ if (sel < 5)
+ scale_index = get_vlc(&s->gb, &vlc_scale_factor, sel);
+ else
+ scale_index = get_bits(&s->gb, sel + 1);
+
+ // Bias by 64
+ scale_index += 64;
+
+ // Look up joint scale factor
+ if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return ff_dca_joint_scale_factors[scale_index];
+}
+
+// 5.4.1 - Primary audio coding side information
+static int parse_subframe_header(DCACoreDecoder *s, int sf,
+ enum HeaderType header, int xch_base)
+{
+ int ch, band, ret;
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (header == HEADER_CORE) {
+ // Subsubframe count
+ s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1;
+
+ // Partial subsubframe sample count
+ skip_bits(&s->gb, 3);
+ }
+
+ // Prediction mode
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ for (band = 0; band < s->nsubbands[ch]; band++)
+ s->prediction_mode[ch][band] = get_bits1(&s->gb);
+
+ // Prediction coefficients VQ address
+ for (ch = xch_base; ch < s->nchannels; ch++)
+ for (band = 0; band < s->nsubbands[ch]; band++)
+ if (s->prediction_mode[ch][band])
+ s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
+
+ // Bit allocation index
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int sel = s->bit_allocation_sel[ch];
+
+ for (band = 0; band < s->subband_vq_start[ch]; band++) {
+ int abits;
+
+ if (sel < 5)
+ abits = get_vlc(&s->gb, &vlc_bit_allocation, sel);
+ else
+ abits = get_bits(&s->gb, sel - 1);
+
+ if (abits > DCA_ABITS_MAX) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ s->bit_allocation[ch][band] = abits;
+ }
+ }
+
+ // Transition mode
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ // Clear transition mode for all subbands
+ memset(s->transition_mode[sf][ch], 0, sizeof(s->transition_mode[0][0]));
+
+ // Transient possible only if more than one subsubframe
+ if (s->nsubsubframes[sf] > 1) {
+ int sel = s->transition_mode_sel[ch];
+ for (band = 0; band < s->subband_vq_start[ch]; band++)
+ if (s->bit_allocation[ch][band])
+ s->transition_mode[sf][ch][band] = get_vlc(&s->gb, &vlc_transition_mode, sel);
+ }
+ }
+
+ // Scale factors
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int sel = s->scale_factor_sel[ch];
+ int scale_index = 0;
+
+ // Extract scales for subbands up to VQ
+ for (band = 0; band < s->subband_vq_start[ch]; band++) {
+ if (s->bit_allocation[ch][band]) {
+ if ((ret = parse_scale(s, &scale_index, sel)) < 0)
+ return ret;
+ s->scale_factors[ch][band][0] = ret;
+ if (s->transition_mode[sf][ch][band]) {
+ if ((ret = parse_scale(s, &scale_index, sel)) < 0)
+ return ret;
+ s->scale_factors[ch][band][1] = ret;
+ }
+ } else {
+ s->scale_factors[ch][band][0] = 0;
+ }
+ }
+
+ // High frequency VQ subbands
+ for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) {
+ if ((ret = parse_scale(s, &scale_index, sel)) < 0)
+ return ret;
+ s->scale_factors[ch][band][0] = ret;
+ }
+ }
+
+ // Joint subband codebook select
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ if (s->joint_intensity_index[ch]) {
+ s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
+ if (s->joint_scale_sel[ch] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor code book\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ }
+
+ // Scale factors for joint subband coding
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int src_ch = s->joint_intensity_index[ch] - 1;
+ if (src_ch >= 0) {
+ int sel = s->joint_scale_sel[ch];
+ for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
+ if ((ret = parse_joint_scale(s, sel)) < 0)
+ return ret;
+ s->joint_scale_factors[ch][band] = ret;
+ }
+ }
+ }
+
+ // Dynamic range coefficient
+ if (s->drc_present && header == HEADER_CORE)
+ skip_bits(&s->gb, 8);
+
+ // Side information CRC check word
+ if (s->crc_present)
+ skip_bits(&s->gb, 16);
+
+ return 0;
+}
+
+#ifndef decode_blockcodes
+static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio)
+{
+ int offset = (levels - 1) / 2;
+ int n, div;
+
+ for (n = 0; n < DCA_SUBBAND_SAMPLES / 2; n++) {
+ div = FASTDIV(code1, levels);
+ audio[n] = code1 - div * levels - offset;
+ code1 = div;
+ }
+ for (; n < DCA_SUBBAND_SAMPLES; n++) {
+ div = FASTDIV(code2, levels);
+ audio[n] = code2 - div * levels - offset;
+ code2 = div;
+ }
+
+ return code1 | code2;
+}
+#endif
+
+static inline int parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits)
+{
+ // Extract block code indices from the bit stream
+ int code1 = get_bits(&s->gb, block_code_nbits[abits - 1]);
+ int code2 = get_bits(&s->gb, block_code_nbits[abits - 1]);
+ int levels = ff_dca_quant_levels[abits];
+
+ // Look up samples from the block code book
+ if (decode_blockcodes(code1, code2, levels, audio)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Failed to decode block code(s)\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static inline int parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel)
+{
+ int i;
+
+ // Extract Huffman codes from the bit stream
+ for (i = 0; i < DCA_SUBBAND_SAMPLES; i++)
+ audio[i] = get_vlc(&s->gb, &vlc_quant_index[abits - 1], sel);
+
+ return 1;
+}
+
+static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch)
+{
+ av_assert1(abits >= 0 && abits <= DCA_ABITS_MAX);
+
+ if (abits == 0) {
+ // No bits allocated
+ memset(audio, 0, DCA_SUBBAND_SAMPLES * sizeof(*audio));
+ return 0;
+ }
+
+ if (abits <= DCA_CODE_BOOKS) {
+ int sel = s->quant_index_sel[ch][abits - 1];
+ if (sel < quant_index_group_size[abits - 1]) {
+ // Huffman codes
+ return parse_huffman_codes(s, audio, abits, sel);
+ }
+ if (abits <= 7) {
+ // Block codes
+ return parse_block_codes(s, audio, abits);
+ }
+ }
+
+ // No further encoding
+ get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
+ return 0;
+}
+
+static inline void dequantize(int32_t *output, const int32_t *input,
+ int32_t step_size, int32_t scale, int residual)
+{
+ // Account for quantizer step size
+ int64_t step_scale = (int64_t)step_size * scale;
+ int n, shift = 0;
+
+ // Limit scale factor resolution to 22 bits
+ if (step_scale > (1 << 23)) {
+ shift = av_log2(step_scale >> 23) + 1;
+ step_scale >>= shift;
+ }
+
+ // Scale the samples
+ if (residual) {
+ for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
+ output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
+ } else {
+ for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
+ output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
+ }
+}
+
+static inline void inverse_adpcm(int32_t **subband_samples,
+ const int16_t *vq_index,
+ const int8_t *prediction_mode,
+ int sb_start, int sb_end,
+ int ofs, int len)
+{
+ int i, j, k;
+
+ for (i = sb_start; i < sb_end; i++) {
+ if (prediction_mode[i]) {
+ const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
+ int32_t *ptr = subband_samples[i] + ofs;
+ for (j = 0; j < len; j++) {
+ int64_t err = 0;
+ for (k = 0; k < DCA_ADPCM_COEFFS; k++)
+ err += (int64_t)ptr[j - k - 1] * coeff[k];
+ ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
+ }
+ }
+ }
+}
+
+// 5.5 - Primary audio data arrays
+static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType header,
+ int xch_base, int *sub_pos, int *lfe_pos)
+{
+ int32_t audio[16], scale;
+ int n, ssf, ofs, ch, band;
+
+ // Check number of subband samples in this subframe
+ int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
+ if (*sub_pos + nsamples > s->npcmblocks) {
+ av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // VQ encoded subbands
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int32_t vq_index[DCA_SUBBANDS];
+
+ for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++)
+ // Extract the VQ address from the bit stream
+ vq_index[band] = get_bits(&s->gb, 10);
+
+ if (s->subband_vq_start[ch] < s->nsubbands[ch]) {
+ s->dcadsp->decode_hf(s->subband_samples[ch], vq_index,
+ ff_dca_high_freq_vq, s->scale_factors[ch],
+ s->subband_vq_start[ch], s->nsubbands[ch],
+ *sub_pos, nsamples);
+ }
+ }
+
+ // Low frequency effect data
+ if (s->lfe_present && header == HEADER_CORE) {
+ unsigned int index;
+
+ // Determine number of LFE samples in this subframe
+ int nlfesamples = 2 * s->lfe_present * s->nsubsubframes[sf];
+ av_assert1((unsigned int)nlfesamples <= FF_ARRAY_ELEMS(audio));
+
+ // Extract LFE samples from the bit stream
+ get_array(&s->gb, audio, nlfesamples, 8);
+
+ // Extract scale factor index from the bit stream
+ index = get_bits(&s->gb, 8);
+ if (index >= FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Look up the 7-bit root square quantization table
+ scale = ff_dca_scale_factor_quant7[index];
+
+ // Account for quantizer step size which is 0.035
+ scale = mul23(4697620 /* 0.035 * (1 << 27) */, scale);
+
+ // Scale and take the LFE samples
+ for (n = 0, ofs = *lfe_pos; n < nlfesamples; n++, ofs++)
+ s->lfe_samples[ofs] = clip23(audio[n] * scale >> 4);
+
+ // Advance LFE sample pointer for the next subframe
+ *lfe_pos = ofs;
+ }
+
+ // Audio data
+ for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // Not high frequency VQ subbands
+ for (band = 0; band < s->subband_vq_start[ch]; band++) {
+ int ret, trans_ssf, abits = s->bit_allocation[ch][band];
+ int32_t step_size;
+
+ // Extract bits from the bit stream
+ if ((ret = extract_audio(s, audio, abits, ch)) < 0)
+ return ret;
+
+ // Select quantization step size table and look up
+ // quantization step size
+ if (s->bit_rate == 3)
+ step_size = ff_dca_lossless_quant[abits];
+ else
+ step_size = ff_dca_lossy_quant[abits];
+
+ // Identify transient location
+ trans_ssf = s->transition_mode[sf][ch][band];
+
+ // Determine proper scale factor
+ if (trans_ssf == 0 || ssf < trans_ssf)
+ scale = s->scale_factors[ch][band][0];
+ else
+ scale = s->scale_factors[ch][band][1];
+
+ // Adjust scale factor when SEL indicates Huffman code
+ if (ret > 0) {
+ int64_t adj = s->scale_factor_adj[ch][abits - 1];
+ scale = clip23(adj * scale >> 22);
+ }
+
+ dequantize(s->subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 0);
+ }
+ }
+
+ // DSYNC
+ if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
+ av_log(s->avctx, AV_LOG_ERROR, "DSYNC check failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ ofs += DCA_SUBBAND_SAMPLES;
+ }
+
+ // Inverse ADPCM
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ inverse_adpcm(s->subband_samples[ch], s->prediction_vq_index[ch],
+ s->prediction_mode[ch], 0, s->nsubbands[ch],
+ *sub_pos, nsamples);
+ }
+
+ // Joint subband coding
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int src_ch = s->joint_intensity_index[ch] - 1;
+ if (src_ch >= 0) {
+ s->dcadsp->decode_joint(s->subband_samples[ch], s->subband_samples[src_ch],
+ s->joint_scale_factors[ch], s->nsubbands[ch],
+ s->nsubbands[src_ch], *sub_pos, nsamples);
+ }
+ }
+
+ // Advance subband sample pointer for the next subframe
+ *sub_pos = ofs;
+ return 0;
+}
+
+static void erase_adpcm_history(DCACoreDecoder *s)
+{
+ int ch, band;
+
+ // Erase ADPCM history from previous frame if
+ // predictor history switch was disabled
+ for (ch = 0; ch < DCA_CHANNELS; ch++)
+ for (band = 0; band < DCA_SUBBANDS; band++)
+ AV_ZERO128(s->subband_samples[ch][band] - DCA_ADPCM_COEFFS);
+}
+
+static int alloc_sample_buffer(DCACoreDecoder *s)
+{
+ int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
+ int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS;
+ int nlfesamples = DCA_LFE_HISTORY + s->npcmblocks / 2;
+ unsigned int size = s->subband_size;
+ int ch, band;
+
+ // Reallocate subband sample buffer
+ av_fast_mallocz(&s->subband_buffer, &s->subband_size,
+ (nframesamples + nlfesamples) * sizeof(int32_t));
+ if (!s->subband_buffer)
+ return AVERROR(ENOMEM);
+
+ if (size != s->subband_size) {
+ for (ch = 0; ch < DCA_CHANNELS; ch++)
+ for (band = 0; band < DCA_SUBBANDS; band++)
+ s->subband_samples[ch][band] = s->subband_buffer +
+ (ch * DCA_SUBBANDS + band) * nchsamples + DCA_ADPCM_COEFFS;
+ s->lfe_samples = s->subband_buffer + nframesamples;
+ }
+
+ if (!s->predictor_history)
+ erase_adpcm_history(s);
+
+ return 0;
+}
+
+static int parse_frame_data(DCACoreDecoder *s, enum HeaderType header, int xch_base)
+{
+ int sf, ch, ret, band, sub_pos, lfe_pos;
+
+ if ((ret = parse_coding_header(s, header, xch_base)) < 0)
+ return ret;
+
+ for (sf = 0, sub_pos = 0, lfe_pos = DCA_LFE_HISTORY; sf < s->nsubframes; sf++) {
+ if ((ret = parse_subframe_header(s, sf, header, xch_base)) < 0)
+ return ret;
+ if ((ret = parse_subframe_audio(s, sf, header, xch_base, &sub_pos, &lfe_pos)) < 0)
+ return ret;
+ }
+
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ // Determine number of active subbands for this channel
+ int nsubbands = s->nsubbands[ch];
+ if (s->joint_intensity_index[ch])
+ nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
+
+ // Update history for ADPCM
+ for (band = 0; band < nsubbands; band++) {
+ int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
+ AV_COPY128(samples, samples + s->npcmblocks);
+ }
+
+ // Clear inactive subbands
+ for (; band < DCA_SUBBANDS; band++) {
+ int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
+ memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
+ }
+ }
+
+ return 0;
+}
+
+static int parse_xch_frame(DCACoreDecoder *s)
+{
+ int ret;
+
+ if (s->ch_mask & DCA_SPEAKER_MASK_Cs) {
+ av_log(s->avctx, AV_LOG_ERROR, "XCH with Cs speaker already present\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if ((ret = parse_frame_data(s, HEADER_XCH, s->nchannels)) < 0)
+ return ret;
+
+ // Seek to the end of core frame, don't trust XCH frame size
+ if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XCH frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int parse_xxch_frame(DCACoreDecoder *s)
+{
+ int xxch_nchsets, xxch_frame_size;
+ int ret, mask, header_size, header_pos = get_bits_count(&s->gb);
+
+ // XXCH sync word
+ if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XXCH) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH sync word\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // XXCH frame header length
+ header_size = get_bits(&s->gb, 6) + 1;
+
+ // Check XXCH frame header CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH frame header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // CRC presence flag for channel set header
+ s->xxch_crc_present = get_bits1(&s->gb);
+
+ // Number of bits for loudspeaker mask
+ s->xxch_mask_nbits = get_bits(&s->gb, 5) + 1;
+ if (s->xxch_mask_nbits <= DCA_SPEAKER_Cs) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XXCH speaker mask (%d)\n", s->xxch_mask_nbits);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of channel sets
+ xxch_nchsets = get_bits(&s->gb, 2) + 1;
+ if (xxch_nchsets > 1) {
+ avpriv_request_sample(s->avctx, "%d XXCH channel sets", xxch_nchsets);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Channel set 0 data byte size
+ xxch_frame_size = get_bits(&s->gb, 14) + 1;
+
+ // Core loudspeaker activity mask
+ s->xxch_core_mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
+
+ // Validate the core mask
+ mask = s->ch_mask;
+
+ if ((mask & DCA_SPEAKER_MASK_Ls) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
+ mask = (mask & ~DCA_SPEAKER_MASK_Ls) | DCA_SPEAKER_MASK_Lss;
+
+ if ((mask & DCA_SPEAKER_MASK_Rs) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
+ mask = (mask & ~DCA_SPEAKER_MASK_Rs) | DCA_SPEAKER_MASK_Rss;
+
+ if (mask != s->xxch_core_mask) {
+ av_log(s->avctx, AV_LOG_ERROR, "XXCH core speaker activity mask (%#x) disagrees with core (%#x)\n", s->xxch_core_mask, mask);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Reserved
+ // Byte align
+ // CRC16 of XXCH frame header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH frame header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Parse XXCH channel set 0
+ if ((ret = parse_frame_data(s, HEADER_XXCH, s->nchannels)) < 0)
+ return ret;
+
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8 + xxch_frame_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchannels,
+ int *xbr_nsubbands, int xbr_transition_mode, int sf, int *sub_pos)
+{
+ int xbr_nabits[DCA_CHANNELS];
+ int xbr_bit_allocation[DCA_CHANNELS][DCA_SUBBANDS];
+ int xbr_scale_nbits[DCA_CHANNELS];
+ int32_t xbr_scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2];
+ int ssf, ch, band, ofs;
+
+ // Check number of subband samples in this subframe
+ if (*sub_pos + s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES > s->npcmblocks) {
+ av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // Number of bits for XBR bit allocation index
+ for (ch = xbr_base_ch; ch < xbr_nchannels; ch++)
+ xbr_nabits[ch] = get_bits(&s->gb, 2) + 2;
+
+ // XBR bit allocation index
+ for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
+ for (band = 0; band < xbr_nsubbands[ch]; band++) {
+ xbr_bit_allocation[ch][band] = get_bits(&s->gb, xbr_nabits[ch]);
+ if (xbr_bit_allocation[ch][band] > DCA_ABITS_MAX) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR bit allocation index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ }
+
+ // Number of bits for scale indices
+ for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
+ xbr_scale_nbits[ch] = get_bits(&s->gb, 3);
+ if (!xbr_scale_nbits[ch]) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XBR scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // XBR scale factors
+ for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
+ const uint32_t *scale_table;
+ int scale_size;
+
+ // Select the root square table
+ if (s->scale_factor_sel[ch] > 5) {
+ scale_table = ff_dca_scale_factor_quant7;
+ scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
+ } else {
+ scale_table = ff_dca_scale_factor_quant6;
+ scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
+ }
+
+ // Parse scale factor indices and look up scale factors from the root
+ // square table
+ for (band = 0; band < xbr_nsubbands[ch]; band++) {
+ if (xbr_bit_allocation[ch][band]) {
+ int scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
+ if (scale_index >= scale_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ xbr_scale_factors[ch][band][0] = scale_table[scale_index];
+ if (xbr_transition_mode && s->transition_mode[sf][ch][band]) {
+ scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
+ if (scale_index >= scale_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ xbr_scale_factors[ch][band][1] = scale_table[scale_index];
+ }
+ }
+ }
+ }
+
+ // Audio data
+ for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
+ for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ for (band = 0; band < xbr_nsubbands[ch]; band++) {
+ int ret, trans_ssf, abits = xbr_bit_allocation[ch][band];
+ int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
+
+ // Extract bits from the bit stream
+ if (abits > 7) {
+ // No further encoding
+ get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
+ } else if (abits > 0) {
+ // Block codes
+ if ((ret = parse_block_codes(s, audio, abits)) < 0)
+ return ret;
+ } else {
+ // No bits allocated
+ continue;
+ }
+
+ // Look up quantization step size
+ step_size = ff_dca_lossless_quant[abits];
+
+ // Identify transient location
+ if (xbr_transition_mode)
+ trans_ssf = s->transition_mode[sf][ch][band];
+ else
+ trans_ssf = 0;
+
+ // Determine proper scale factor
+ if (trans_ssf == 0 || ssf < trans_ssf)
+ scale = xbr_scale_factors[ch][band][0];
+ else
+ scale = xbr_scale_factors[ch][band][1];
+
+ dequantize(s->subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 1);
+ }
+ }
+
+ // DSYNC
+ if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
+ av_log(s->avctx, AV_LOG_ERROR, "XBR-DSYNC check failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ ofs += DCA_SUBBAND_SAMPLES;
+ }
+
+ // Advance subband sample pointer for the next subframe
+ *sub_pos = ofs;
+ return 0;
+}
+
+static int parse_xbr_frame(DCACoreDecoder *s)
+{
+ int xbr_frame_size[DCA_EXSS_CHSETS_MAX];
+ int xbr_nchannels[DCA_EXSS_CHSETS_MAX];
+ int xbr_nsubbands[DCA_EXSS_CHSETS_MAX * DCA_EXSS_CHANNELS_MAX];
+ int xbr_nchsets, xbr_transition_mode, xbr_band_nbits, xbr_base_ch;
+ int i, ch1, ch2, ret, header_size, header_pos = get_bits_count(&s->gb);
+
+ // XBR sync word
+ if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XBR) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR sync word\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // XBR frame header length
+ header_size = get_bits(&s->gb, 6) + 1;
+
+ // Check XBR frame header CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR frame header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of channel sets
+ xbr_nchsets = get_bits(&s->gb, 2) + 1;
+
+ // Channel set data byte size
+ for (i = 0; i < xbr_nchsets; i++)
+ xbr_frame_size[i] = get_bits(&s->gb, 14) + 1;
+
+ // Transition mode flag
+ xbr_transition_mode = get_bits1(&s->gb);
+
+ // Channel set headers
+ for (i = 0, ch2 = 0; i < xbr_nchsets; i++) {
+ xbr_nchannels[i] = get_bits(&s->gb, 3) + 1;
+ xbr_band_nbits = get_bits(&s->gb, 2) + 5;
+ for (ch1 = 0; ch1 < xbr_nchannels[i]; ch1++, ch2++) {
+ xbr_nsubbands[ch2] = get_bits(&s->gb, xbr_band_nbits) + 1;
+ if (xbr_nsubbands[ch2] > DCA_SUBBANDS) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid number of active XBR subbands (%d)\n", xbr_nsubbands[ch2]);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ }
+
+ // Reserved
+ // Byte align
+ // CRC16 of XBR frame header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR frame header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Channel set data
+ for (i = 0, xbr_base_ch = 0; i < xbr_nchsets; i++) {
+ header_pos = get_bits_count(&s->gb);
+
+ if (xbr_base_ch + xbr_nchannels[i] <= s->nchannels) {
+ int sf, sub_pos;
+
+ for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
+ if ((ret = parse_xbr_subframe(s, xbr_base_ch,
+ xbr_base_ch + xbr_nchannels[i],
+ xbr_nsubbands, xbr_transition_mode,
+ sf, &sub_pos)) < 0)
+ return ret;
+ }
+ }
+
+ xbr_base_ch += xbr_nchannels[i];
+
+ if (ff_dca_seek_bits(&s->gb, header_pos + xbr_frame_size[i] * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR channel set\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ return 0;
+}
+
+// Modified ISO/IEC 9899 linear congruential generator
+// Returns pseudorandom integer in range [-2^30, 2^30 - 1]
+static int rand_x96(DCACoreDecoder *s)
+{
+ s->x96_rand = 1103515245U * s->x96_rand + 12345U;
+ return (s->x96_rand & 0x7fffffff) - 0x40000000;
+}
+
+static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int *sub_pos)
+{
+ int n, ssf, ch, band, ofs;
+
+ // Check number of subband samples in this subframe
+ int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
+ if (*sub_pos + nsamples > s->npcmblocks) {
+ av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // VQ encoded or unallocated subbands
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
+ // Get the sample pointer and scale factor
+ int32_t *samples = s->x96_subband_samples[ch][band] + *sub_pos;
+ int32_t scale = s->scale_factors[ch][band >> 1][band & 1];
+
+ switch (s->bit_allocation[ch][band]) {
+ case 0: // No bits allocated for subband
+ if (scale <= 1)
+ memset(samples, 0, nsamples * sizeof(int32_t));
+ else for (n = 0; n < nsamples; n++)
+ // Generate scaled random samples
+ samples[n] = mul31(rand_x96(s), scale);
+ break;
+
+ case 1: // VQ encoded subband
+ for (ssf = 0; ssf < (s->nsubsubframes[sf] + 1) / 2; ssf++) {
+ // Extract the VQ address from the bit stream and look up
+ // the VQ code book for up to 16 subband samples
+ const int8_t *vq_samples = ff_dca_high_freq_vq[get_bits(&s->gb, 10)];
+ // Scale and take the samples
+ for (n = 0; n < FFMIN(nsamples - ssf * 16, 16); n++)
+ *samples++ = clip23(vq_samples[n] * scale + (1 << 3) >> 4);
+ }
+ break;
+ }
+ }
+ }
+
+ // Audio data
+ for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
+ int ret, abits = s->bit_allocation[ch][band] - 1;
+ int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
+
+ // Not VQ encoded or unallocated subbands
+ if (abits < 1)
+ continue;
+
+ // Extract bits from the bit stream
+ if ((ret = extract_audio(s, audio, abits, ch)) < 0)
+ return ret;
+
+ // Select quantization step size table and look up quantization
+ // step size
+ if (s->bit_rate == 3)
+ step_size = ff_dca_lossless_quant[abits];
+ else
+ step_size = ff_dca_lossy_quant[abits];
+
+ // Get the scale factor
+ scale = s->scale_factors[ch][band >> 1][band & 1];
+
+ dequantize(s->x96_subband_samples[ch][band] + ofs,
+ audio, step_size, scale, 0);
+ }
+ }
+
+ // DSYNC
+ if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
+ av_log(s->avctx, AV_LOG_ERROR, "X96-DSYNC check failed\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ ofs += DCA_SUBBAND_SAMPLES;
+ }
+
+ // Inverse ADPCM
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ inverse_adpcm(s->x96_subband_samples[ch], s->prediction_vq_index[ch],
+ s->prediction_mode[ch], s->x96_subband_start, s->nsubbands[ch],
+ *sub_pos, nsamples);
+ }
+
+ // Joint subband coding
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ int src_ch = s->joint_intensity_index[ch] - 1;
+ if (src_ch >= 0) {
+ s->dcadsp->decode_joint(s->x96_subband_samples[ch], s->x96_subband_samples[src_ch],
+ s->joint_scale_factors[ch], s->nsubbands[ch],
+ s->nsubbands[src_ch], *sub_pos, nsamples);
+ }
+ }
+
+ // Advance subband sample pointer for the next subframe
+ *sub_pos = ofs;
+ return 0;
+}
+
+static void erase_x96_adpcm_history(DCACoreDecoder *s)
+{
+ int ch, band;
+
+ // Erase ADPCM history from previous frame if
+ // predictor history switch was disabled
+ for (ch = 0; ch < DCA_CHANNELS; ch++)
+ for (band = 0; band < DCA_SUBBANDS_X96; band++)
+ AV_ZERO128(s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS);
+}
+
+static int alloc_x96_sample_buffer(DCACoreDecoder *s)
+{
+ int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
+ int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS_X96;
+ unsigned int size = s->x96_subband_size;
+ int ch, band;
+
+ // Reallocate subband sample buffer
+ av_fast_mallocz(&s->x96_subband_buffer, &s->x96_subband_size,
+ nframesamples * sizeof(int32_t));
+ if (!s->x96_subband_buffer)
+ return AVERROR(ENOMEM);
+
+ if (size != s->x96_subband_size) {
+ for (ch = 0; ch < DCA_CHANNELS; ch++)
+ for (band = 0; band < DCA_SUBBANDS_X96; band++)
+ s->x96_subband_samples[ch][band] = s->x96_subband_buffer +
+ (ch * DCA_SUBBANDS_X96 + band) * nchsamples + DCA_ADPCM_COEFFS;
+ }
+
+ if (!s->predictor_history)
+ erase_x96_adpcm_history(s);
+
+ return 0;
+}
+
+static int parse_x96_subframe_header(DCACoreDecoder *s, int xch_base)
+{
+ int ch, band, ret;
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // Prediction mode
+ for (ch = xch_base; ch < s->x96_nchannels; ch++)
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
+ s->prediction_mode[ch][band] = get_bits1(&s->gb);
+
+ // Prediction coefficients VQ address
+ for (ch = xch_base; ch < s->x96_nchannels; ch++)
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
+ if (s->prediction_mode[ch][band])
+ s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
+
+ // Bit allocation index
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ int sel = s->bit_allocation_sel[ch];
+ int abits = 0;
+
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
+ // If Huffman code was used, the difference of abits was encoded
+ if (sel < 7)
+ abits += get_vlc(&s->gb, &vlc_quant_index[5 + 2 * s->x96_high_res], sel);
+ else
+ abits = get_bits(&s->gb, 3 + s->x96_high_res);
+
+ if (abits < 0 || abits > 7 + 8 * s->x96_high_res) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 bit allocation index\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ s->bit_allocation[ch][band] = abits;
+ }
+ }
+
+ // Scale factors
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ int sel = s->scale_factor_sel[ch];
+ int scale_index = 0;
+
+ // Extract scales for subbands which are transmitted even for
+ // unallocated subbands
+ for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
+ if ((ret = parse_scale(s, &scale_index, sel)) < 0)
+ return ret;
+ s->scale_factors[ch][band >> 1][band & 1] = ret;
+ }
+ }
+
+ // Joint subband codebook select
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ if (s->joint_intensity_index[ch]) {
+ s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
+ if (s->joint_scale_sel[ch] == 7) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint scale factor code book\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ }
+
+ // Scale factors for joint subband coding
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ int src_ch = s->joint_intensity_index[ch] - 1;
+ if (src_ch >= 0) {
+ int sel = s->joint_scale_sel[ch];
+ for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
+ if ((ret = parse_joint_scale(s, sel)) < 0)
+ return ret;
+ s->joint_scale_factors[ch][band] = ret;
+ }
+ }
+ }
+
+ // Side information CRC check word
+ if (s->crc_present)
+ skip_bits(&s->gb, 16);
+
+ return 0;
+}
+
+static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base)
+{
+ int n, ch, header_size = 0, header_pos = get_bits_count(&s->gb);
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (exss) {
+ // Channel set header length
+ header_size = get_bits(&s->gb, 7) + 1;
+
+ // Check CRC
+ if (s->x96_crc_present
+ && (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 channel set header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // High resolution flag
+ s->x96_high_res = get_bits1(&s->gb);
+
+ // First encoded subband
+ if (s->x96_rev_no < 8) {
+ s->x96_subband_start = get_bits(&s->gb, 5);
+ if (s->x96_subband_start > 27) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband start index (%d)\n", s->x96_subband_start);
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+ s->x96_subband_start = DCA_SUBBANDS;
+ }
+
+ // Subband activity count
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ s->nsubbands[ch] = get_bits(&s->gb, 6) + 1;
+ if (s->nsubbands[ch] < DCA_SUBBANDS) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband activity count (%d)\n", s->nsubbands[ch]);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Joint intensity coding index
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ if ((n = get_bits(&s->gb, 3)) && xch_base)
+ n += xch_base - 1;
+ if (n > s->x96_nchannels) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint intensity coding index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->joint_intensity_index[ch] = n;
+ }
+
+ // Scale factor code book
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
+ if (s->scale_factor_sel[ch] >= 6) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 scale factor code book\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Bit allocation quantizer select
+ for (ch = xch_base; ch < s->x96_nchannels; ch++)
+ s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
+
+ // Quantization index codebook select
+ for (n = 0; n < 6 + 4 * s->x96_high_res; n++)
+ for (ch = xch_base; ch < s->x96_nchannels; ch++)
+ s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
+
+ if (exss) {
+ // Reserved
+ // Byte align
+ // CRC16 of channel set header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set header\n");
+ return AVERROR_INVALIDDATA;
+ }
+ } else {
+ if (s->crc_present)
+ skip_bits(&s->gb, 16);
+ }
+
+ return 0;
+}
+
+static int parse_x96_frame_data(DCACoreDecoder *s, int exss, int xch_base)
+{
+ int sf, ch, ret, band, sub_pos;
+
+ if ((ret = parse_x96_coding_header(s, exss, xch_base)) < 0)
+ return ret;
+
+ for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
+ if ((ret = parse_x96_subframe_header(s, xch_base)) < 0)
+ return ret;
+ if ((ret = parse_x96_subframe_audio(s, sf, xch_base, &sub_pos)) < 0)
+ return ret;
+ }
+
+ for (ch = xch_base; ch < s->x96_nchannels; ch++) {
+ // Determine number of active subbands for this channel
+ int nsubbands = s->nsubbands[ch];
+ if (s->joint_intensity_index[ch])
+ nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
+
+ // Update history for ADPCM and clear inactive subbands
+ for (band = 0; band < DCA_SUBBANDS_X96; band++) {
+ int32_t *samples = s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS;
+ if (band >= s->x96_subband_start && band < nsubbands)
+ AV_COPY128(samples, samples + s->npcmblocks);
+ else
+ memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
+ }
+ }
+
+ return 0;
+}
+
+static int parse_x96_frame(DCACoreDecoder *s)
+{
+ int ret;
+
+ // Revision number
+ s->x96_rev_no = get_bits(&s->gb, 4);
+ if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
+ return AVERROR_INVALIDDATA;
+ }
+
+ s->x96_crc_present = 0;
+ s->x96_nchannels = s->nchannels;
+
+ if ((ret = alloc_x96_sample_buffer(s)) < 0)
+ return ret;
+
+ if ((ret = parse_x96_frame_data(s, 0, 0)) < 0)
+ return ret;
+
+ // Seek to the end of core frame
+ if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int parse_x96_frame_exss(DCACoreDecoder *s)
+{
+ int x96_frame_size[DCA_EXSS_CHSETS_MAX];
+ int x96_nchannels[DCA_EXSS_CHSETS_MAX];
+ int x96_nchsets, x96_base_ch;
+ int i, ret, header_size, header_pos = get_bits_count(&s->gb);
+
+ // X96 sync word
+ if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_X96) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 sync word\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // X96 frame header length
+ header_size = get_bits(&s->gb, 6) + 1;
+
+ // Check X96 frame header CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 frame header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Revision number
+ s->x96_rev_no = get_bits(&s->gb, 4);
+ if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // CRC presence flag for channel set header
+ s->x96_crc_present = get_bits1(&s->gb);
+
+ // Number of channel sets
+ x96_nchsets = get_bits(&s->gb, 2) + 1;
+
+ // Channel set data byte size
+ for (i = 0; i < x96_nchsets; i++)
+ x96_frame_size[i] = get_bits(&s->gb, 12) + 1;
+
+ // Number of channels in channel set
+ for (i = 0; i < x96_nchsets; i++)
+ x96_nchannels[i] = get_bits(&s->gb, 3) + 1;
+
+ // Reserved
+ // Byte align
+ // CRC16 of X96 frame header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if ((ret = alloc_x96_sample_buffer(s)) < 0)
+ return ret;
+
+ // Channel set data
+ for (i = 0, x96_base_ch = 0; i < x96_nchsets; i++) {
+ header_pos = get_bits_count(&s->gb);
+
+ if (x96_base_ch + x96_nchannels[i] <= s->nchannels) {
+ s->x96_nchannels = x96_base_ch + x96_nchannels[i];
+ if ((ret = parse_x96_frame_data(s, 1, x96_base_ch)) < 0)
+ return ret;
+ }
+
+ x96_base_ch += x96_nchannels[i];
+
+ if (ff_dca_seek_bits(&s->gb, header_pos + x96_frame_size[i] * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ return 0;
+}
+
+static int parse_aux_data(DCACoreDecoder *s)
+{
+ int aux_pos;
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ // Auxiliary data byte count (can't be trusted)
+ skip_bits(&s->gb, 6);
+
+ // 4-byte align
+ skip_bits_long(&s->gb, -get_bits_count(&s->gb) & 31);
+
+ // Auxiliary data sync word
+ if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_REV1AUX) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data sync word\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ aux_pos = get_bits_count(&s->gb);
+
+ // Auxiliary decode time stamp flag
+ if (get_bits1(&s->gb))
+ skip_bits_long(&s->gb, 47);
+
+ // Auxiliary dynamic downmix flag
+ if (s->prim_dmix_embedded = get_bits1(&s->gb)) {
+ int i, m, n;
+
+ // Auxiliary primary channel downmix type
+ s->prim_dmix_type = get_bits(&s->gb, 3);
+ if (s->prim_dmix_type >= DCA_DMIX_TYPE_COUNT) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid primary channel set downmix type\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Size of downmix coefficients matrix
+ m = ff_dca_dmix_primary_nch[s->prim_dmix_type];
+ n = ff_dca_channels[s->audio_mode] + !!s->lfe_present;
+
+ // Dynamic downmix code coefficients
+ for (i = 0; i < m * n; i++) {
+ int code = get_bits(&s->gb, 9);
+ int sign = (code >> 8) - 1;
+ unsigned int index = code & 0xff;
+ if (index >= FF_DCA_DMIXTABLE_SIZE) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->prim_dmix_coeff[i] = (ff_dca_dmixtable[index] ^ sign) - sign;
+ }
+ }
+
+ // Byte align
+ skip_bits(&s->gb, -get_bits_count(&s->gb) & 7);
+
+ // CRC16 of auxiliary data
+ skip_bits(&s->gb, 16);
+
+ // Check CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, aux_pos, get_bits_count(&s->gb))) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int parse_optional_info(DCACoreDecoder *s)
+{
+ DCAContext *dca = s->avctx->priv_data;
+ int ret = -1;
+
+ // Time code stamp
+ if (s->ts_present)
+ skip_bits_long(&s->gb, 32);
+
+ // Auxiliary data
+ if (s->aux_present && (ret = parse_aux_data(s)) < 0
+ && (s->avctx->err_recognition & AV_EF_EXPLODE))
+ return ret;
+
+ if (ret < 0)
+ s->prim_dmix_embedded = 0;
+
+ // Core extensions
+ if (s->ext_audio_present && !dca->core_only) {
+ int sync_pos = FFMIN(s->frame_size / 4, s->gb.size_in_bits / 32) - 1;
+ int last_pos = get_bits_count(&s->gb) / 32;
+ int size, dist;
+
+ // Search for extension sync words aligned on 4-byte boundary. Search
+ // must be done backwards from the end of core frame to work around
+ // sync word aliasing issues.
+ switch (s->ext_audio_type) {
+ case EXT_AUDIO_XCH:
+ if (dca->request_channel_layout)
+ break;
+
+ // The distance between XCH sync word and end of the core frame
+ // must be equal to XCH frame size. Off by one error is allowed for
+ // compatibility with legacy bitstreams. Minimum XCH frame size is
+ // 96 bytes. AMODE and PCHS are further checked to reduce
+ // probability of alias sync detection.
+ for (; sync_pos >= last_pos; sync_pos--) {
+ if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XCH) {
+ s->gb.index = (sync_pos + 1) * 32;
+ size = get_bits(&s->gb, 10) + 1;
+ dist = s->frame_size - sync_pos * 4;
+ if (size >= 96
+ && (size == dist || size - 1 == dist)
+ && get_bits(&s->gb, 7) == 0x08) {
+ s->xch_pos = get_bits_count(&s->gb);
+ break;
+ }
+ }
+ }
+
+ if (s->avctx->err_recognition & AV_EF_EXPLODE) {
+ av_log(s->avctx, AV_LOG_ERROR, "XCH sync word not found\n");
+ return AVERROR_INVALIDDATA;
+ }
+ break;
+
+ case EXT_AUDIO_X96:
+ // The distance between X96 sync word and end of the core frame
+ // must be equal to X96 frame size. Minimum X96 frame size is 96
+ // bytes.
+ for (; sync_pos >= last_pos; sync_pos--) {
+ if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_X96) {
+ s->gb.index = (sync_pos + 1) * 32;
+ size = get_bits(&s->gb, 12) + 1;
+ dist = s->frame_size - sync_pos * 4;
+ if (size >= 96 && size == dist) {
+ s->x96_pos = get_bits_count(&s->gb);
+ break;
+ }
+ }
+ }
+
+ if (s->avctx->err_recognition & AV_EF_EXPLODE) {
+ av_log(s->avctx, AV_LOG_ERROR, "X96 sync word not found\n");
+ return AVERROR_INVALIDDATA;
+ }
+ break;
+
+ case EXT_AUDIO_XXCH:
+ if (dca->request_channel_layout)
+ break;
+
+ // XXCH frame header CRC must be valid. Minimum XXCH frame header
+ // size is 11 bytes.
+ for (; sync_pos >= last_pos; sync_pos--) {
+ if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XXCH) {
+ s->gb.index = (sync_pos + 1) * 32;
+ size = get_bits(&s->gb, 6) + 1;
+ if (size >= 11 &&
+ !ff_dca_check_crc(&s->gb, (sync_pos + 1) * 32,
+ sync_pos * 32 + size * 8)) {
+ s->xxch_pos = sync_pos * 32;
+ break;
+ }
+ }
+ }
+
+ if (s->avctx->err_recognition & AV_EF_EXPLODE) {
+ av_log(s->avctx, AV_LOG_ERROR, "XXCH sync word not found\n");
+ return AVERROR_INVALIDDATA;
+ }
+ break;
+ }
+ }
+
+ return 0;
+}
+
+int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size)
+{
+ int ret;
+
+ s->ext_audio_mask = 0;
+ s->xch_pos = s->xxch_pos = s->x96_pos = 0;
+
+ if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
+ return ret;
+
+ skip_bits_long(&s->gb, 32);
+ if ((ret = parse_frame_header(s)) < 0)
+ return ret;
+ if ((ret = alloc_sample_buffer(s)) < 0)
+ return ret;
+ if ((ret = parse_frame_data(s, HEADER_CORE, 0)) < 0)
+ return ret;
+ if ((ret = parse_optional_info(s)) < 0)
+ return ret;
+
+ // Workaround for DTS in WAV
+ if (s->frame_size > size && s->frame_size < size + 4) {
+ av_log(s->avctx, AV_LOG_DEBUG, "Working around excessive core frame size (%d > %d)\n", s->frame_size, size);
+ s->frame_size = size;
+ }
+
+ if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of core frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset)
+{
+ AVCodecContext *avctx = s->avctx;
+ DCAContext *dca = avctx->priv_data;
+ GetBitContext gb = s->gb;
+ int exss_mask = asset ? asset->extension_mask : 0;
+ int ret = 0, ext = 0;
+
+ // Parse (X)XCH unless downmixing
+ if (!dca->request_channel_layout) {
+ if (exss_mask & DCA_EXSS_XXCH) {
+ if ((ret = init_get_bits8(&s->gb, data + asset->xxch_offset, asset->xxch_size)) < 0)
+ return ret;
+ ret = parse_xxch_frame(s);
+ ext = DCA_EXSS_XXCH;
+ } else if (s->xxch_pos) {
+ s->gb.index = s->xxch_pos;
+ ret = parse_xxch_frame(s);
+ ext = DCA_CSS_XXCH;
+ } else if (s->xch_pos) {
+ s->gb.index = s->xch_pos;
+ ret = parse_xch_frame(s);
+ ext = DCA_CSS_XCH;
+ }
+
+ // Revert to primary channel set in case (X)XCH parsing fails
+ if (ret < 0) {
+ if (avctx->err_recognition & AV_EF_EXPLODE)
+ return ret;
+ s->nchannels = ff_dca_channels[s->audio_mode];
+ s->ch_mask = audio_mode_ch_mask[s->audio_mode];
+ if (s->lfe_present)
+ s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
+ } else {
+ s->ext_audio_mask |= ext;
+ }
+ }
+
+ // Parse XBR
+ if (exss_mask & DCA_EXSS_XBR) {
+ if ((ret = init_get_bits8(&s->gb, data + asset->xbr_offset, asset->xbr_size)) < 0)
+ return ret;
+ if ((ret = parse_xbr_frame(s)) < 0) {
+ if (avctx->err_recognition & AV_EF_EXPLODE)
+ return ret;
+ } else {
+ s->ext_audio_mask |= DCA_EXSS_XBR;
+ }
+ }
+
+ // Parse X96 unless decoding XLL
+ if (!(dca->packet & DCA_PACKET_XLL)) {
+ if (exss_mask & DCA_EXSS_X96) {
+ if ((ret = init_get_bits8(&s->gb, data + asset->x96_offset, asset->x96_size)) < 0)
+ return ret;
+ if ((ret = parse_x96_frame_exss(s)) < 0) {
+ if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
+ return ret;
+ } else {
+ s->ext_audio_mask |= DCA_EXSS_X96;
+ }
+ } else if (s->x96_pos) {
+ s->gb = gb;
+ s->gb.index = s->x96_pos;
+ if ((ret = parse_x96_frame(s)) < 0) {
+ if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
+ return ret;
+ } else {
+ s->ext_audio_mask |= DCA_CSS_X96;
+ }
+ }
+ }
+
+ return 0;
+}
+
+static int map_prm_ch_to_spkr(DCACoreDecoder *s, int ch)
+{
+ int pos, spkr;
+
+ // Try to map this channel to core first
+ pos = ff_dca_channels[s->audio_mode];
+ if (ch < pos) {
+ spkr = prm_ch_to_spkr_map[s->audio_mode][ch];
+ if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
+ if (s->xxch_core_mask & (1U << spkr))
+ return spkr;
+ if (spkr == DCA_SPEAKER_Ls && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
+ return DCA_SPEAKER_Lss;
+ if (spkr == DCA_SPEAKER_Rs && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
+ return DCA_SPEAKER_Rss;
+ return -1;
+ }
+ return spkr;
+ }
+
+ // Then XCH
+ if ((s->ext_audio_mask & DCA_CSS_XCH) && ch == pos)
+ return DCA_SPEAKER_Cs;
+
+ // Then XXCH
+ if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
+ for (spkr = DCA_SPEAKER_Cs; spkr < s->xxch_mask_nbits; spkr++)
+ if (s->xxch_spkr_mask & (1U << spkr))
+ if (pos++ == ch)
+ return spkr;
+ }
+
+ // No mapping
+ return -1;
+}
+
+static void erase_dsp_history(DCACoreDecoder *s)
+{
+ memset(s->dcadsp_data, 0, sizeof(s->dcadsp_data));
+ s->output_history_lfe_fixed = 0;
+ s->output_history_lfe_float = 0;
+}
+
+static void set_filter_mode(DCACoreDecoder *s, int mode)
+{
+ if (s->filter_mode != mode) {
+ erase_dsp_history(s);
+ s->filter_mode = mode;
+ }
+}
+
+int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
+{
+ int n, ch, spkr, nsamples, x96_nchannels = 0;
+ const int32_t *filter_coeff;
+ int32_t *ptr;
+
+ // Externally set x96_synth flag implies that X96 synthesis should be
+ // enabled, yet actual X96 subband data should be discarded. This is a
+ // special case for lossless residual decoder that ignores X96 data if
+ // present.
+ if (!x96_synth && (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96))) {
+ x96_nchannels = s->x96_nchannels;
+ x96_synth = 1;
+ }
+ if (x96_synth < 0)
+ x96_synth = 0;
+
+ s->output_rate = s->sample_rate << x96_synth;
+ s->npcmsamples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
+
+ // Reallocate PCM output buffer
+ av_fast_malloc(&s->output_buffer, &s->output_size,
+ nsamples * av_popcount(s->ch_mask) * sizeof(int32_t));
+ if (!s->output_buffer)
+ return AVERROR(ENOMEM);
+
+ ptr = (int32_t *)s->output_buffer;
+ for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
+ if (s->ch_mask & (1U << spkr)) {
+ s->output_samples[spkr] = ptr;
+ ptr += nsamples;
+ } else {
+ s->output_samples[spkr] = NULL;
+ }
+ }
+
+ // Handle change of filtering mode
+ set_filter_mode(s, x96_synth | DCA_FILTER_MODE_FIXED);
+
+ // Select filter
+ if (x96_synth)
+ filter_coeff = ff_dca_fir_64bands_fixed;
+ else if (s->filter_perfect)
+ filter_coeff = ff_dca_fir_32bands_perfect_fixed;
+ else
+ filter_coeff = ff_dca_fir_32bands_nonperfect_fixed;
+
+ // Filter primary channels
+ for (ch = 0; ch < s->nchannels; ch++) {
+ // Map this primary channel to speaker
+ spkr = map_prm_ch_to_spkr(s, ch);
+ if (spkr < 0)
+ return AVERROR(EINVAL);
+
+ // Filter bank reconstruction
+ s->dcadsp->sub_qmf_fixed[x96_synth](
+ &s->synth,
+ &s->dcadct,
+ s->output_samples[spkr],
+ s->subband_samples[ch],
+ ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
+ s->dcadsp_data[ch].u.fix.hist1,
+ &s->dcadsp_data[ch].offset,
+ s->dcadsp_data[ch].u.fix.hist2,
+ filter_coeff,
+ s->npcmblocks);
+ }
+
+ // Filter LFE channel
+ if (s->lfe_present) {
+ int32_t *samples = s->output_samples[DCA_SPEAKER_LFE1];
+ int nlfesamples = s->npcmblocks >> 1;
+
+ // Check LFF
+ if (s->lfe_present == LFE_FLAG_128) {
+ av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n");
+ return AVERROR(EINVAL);
+ }
+
+ // Offset intermediate buffer for X96
+ if (x96_synth)
+ samples += nsamples / 2;
+
+ // Interpolate LFE channel
+ s->dcadsp->lfe_fir_fixed(samples, s->lfe_samples + DCA_LFE_HISTORY,
+ ff_dca_lfe_fir_64_fixed, s->npcmblocks);
+
+ if (x96_synth) {
+ // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
+ // (47.6 - 48.0 kHz) components of interpolation image
+ s->dcadsp->lfe_x96_fixed(s->output_samples[DCA_SPEAKER_LFE1],
+ samples, &s->output_history_lfe_fixed,
+ nsamples / 2);
+
+ }
+
+ // Update LFE history
+ for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
+ s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
+ }
+
+ return 0;
+}
+
+static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
+{
+ AVCodecContext *avctx = s->avctx;
+ DCAContext *dca = avctx->priv_data;
+ int i, n, ch, ret, spkr, nsamples;
+
+ // Don't filter twice when falling back from XLL
+ if (!(dca->packet & DCA_PACKET_XLL) && (ret = ff_dca_core_filter_fixed(s, 0)) < 0)
+ return ret;
+
+ avctx->sample_rate = s->output_rate;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ avctx->bits_per_raw_sample = 24;
+
+ frame->nb_samples = nsamples = s->npcmsamples;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+ return ret;
+
+ // Undo embedded XCH downmix
+ if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
+ && s->audio_mode >= AMODE_2F2R) {
+ s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls],
+ s->output_samples[DCA_SPEAKER_Rs],
+ s->output_samples[DCA_SPEAKER_Cs],
+ nsamples);
+
+ }
+
+ // Undo embedded XXCH downmix
+ if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
+ && s->xxch_dmix_embedded) {
+ int scale_inv = s->xxch_dmix_scale_inv;
+ int *coeff_ptr = s->xxch_dmix_coeff;
+ int xch_base = ff_dca_channels[s->audio_mode];
+ av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
+
+ // Undo embedded core downmix pre-scaling
+ for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
+ if (s->xxch_core_mask & (1U << spkr)) {
+ s->dcadsp->dmix_scale_inv(s->output_samples[spkr],
+ scale_inv, nsamples);
+ }
+ }
+
+ // Undo downmix
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int src_spkr = map_prm_ch_to_spkr(s, ch);
+ if (src_spkr < 0)
+ return AVERROR(EINVAL);
+ for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
+ if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
+ int coeff = mul16(*coeff_ptr++, scale_inv);
+ if (coeff) {
+ s->dcadsp->dmix_sub(s->output_samples[spkr ],
+ s->output_samples[src_spkr],
+ coeff, nsamples);
+ }
+ }
+ }
+ }
+ }
+
+ if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
+ // Front sum/difference decoding
+ if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
+ || s->audio_mode == AMODE_STEREO_SUMDIFF) {
+ s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L],
+ s->output_samples[DCA_SPEAKER_R],
+ nsamples);
+ }
+
+ // Surround sum/difference decoding
+ if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
+ s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls],
+ s->output_samples[DCA_SPEAKER_Rs],
+ nsamples);
+ }
+ }
+
+ // Downmix primary channel set to stereo
+ if (s->request_mask != s->ch_mask) {
+ ff_dca_downmix_to_stereo_fixed(s->dcadsp,
+ s->output_samples,
+ s->prim_dmix_coeff,
+ nsamples, s->ch_mask);
+ }
+
+ for (i = 0; i < avctx->channels; i++) {
+ int32_t *samples = s->output_samples[s->ch_remap[i]];
+ int32_t *plane = (int32_t *)frame->extended_data[i];
+ for (n = 0; n < nsamples; n++)
+ plane[n] = clip23(samples[n]) * (1 << 8);
+ }
+
+ return 0;
+}
+
+static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
+{
+ AVCodecContext *avctx = s->avctx;
+ int x96_nchannels = 0, x96_synth = 0;
+ int i, n, ch, ret, spkr, nsamples, nchannels;
+ float *output_samples[DCA_SPEAKER_COUNT] = { NULL }, *ptr;
+ const float *filter_coeff;
+
+ if (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96)) {
+ x96_nchannels = s->x96_nchannels;
+ x96_synth = 1;
+ }
+
+ avctx->sample_rate = s->sample_rate << x96_synth;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ avctx->bits_per_raw_sample = 0;
+
+ frame->nb_samples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+ return ret;
+
+ // Build reverse speaker to channel mapping
+ for (i = 0; i < avctx->channels; i++)
+ output_samples[s->ch_remap[i]] = (float *)frame->extended_data[i];
+
+ // Allocate space for extra channels
+ nchannels = av_popcount(s->ch_mask) - avctx->channels;
+ if (nchannels > 0) {
+ av_fast_malloc(&s->output_buffer, &s->output_size,
+ nsamples * nchannels * sizeof(float));
+ if (!s->output_buffer)
+ return AVERROR(ENOMEM);
+
+ ptr = (float *)s->output_buffer;
+ for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
+ if (!(s->ch_mask & (1U << spkr)))
+ continue;
+ if (output_samples[spkr])
+ continue;
+ output_samples[spkr] = ptr;
+ ptr += nsamples;
+ }
+ }
+
+ // Handle change of filtering mode
+ set_filter_mode(s, x96_synth);
+
+ // Select filter
+ if (x96_synth)
+ filter_coeff = ff_dca_fir_64bands;
+ else if (s->filter_perfect)
+ filter_coeff = ff_dca_fir_32bands_perfect;
+ else
+ filter_coeff = ff_dca_fir_32bands_nonperfect;
+
+ // Filter primary channels
+ for (ch = 0; ch < s->nchannels; ch++) {
+ // Map this primary channel to speaker
+ spkr = map_prm_ch_to_spkr(s, ch);
+ if (spkr < 0)
+ return AVERROR(EINVAL);
+
+ // Filter bank reconstruction
+ s->dcadsp->sub_qmf_float[x96_synth](
+ &s->synth,
+ &s->imdct[x96_synth],
+ output_samples[spkr],
+ s->subband_samples[ch],
+ ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
+ s->dcadsp_data[ch].u.flt.hist1,
+ &s->dcadsp_data[ch].offset,
+ s->dcadsp_data[ch].u.flt.hist2,
+ filter_coeff,
+ s->npcmblocks,
+ 1.0f / (1 << (17 - x96_synth)));
+ }
+
+ // Filter LFE channel
+ if (s->lfe_present) {
+ int dec_select = (s->lfe_present == LFE_FLAG_128);
+ float *samples = output_samples[DCA_SPEAKER_LFE1];
+ int nlfesamples = s->npcmblocks >> (dec_select + 1);
+
+ // Offset intermediate buffer for X96
+ if (x96_synth)
+ samples += nsamples / 2;
+
+ // Select filter
+ if (dec_select)
+ filter_coeff = ff_dca_lfe_fir_128;
+ else
+ filter_coeff = ff_dca_lfe_fir_64;
+
+ // Interpolate LFE channel
+ s->dcadsp->lfe_fir_float[dec_select](
+ samples, s->lfe_samples + DCA_LFE_HISTORY,
+ filter_coeff, s->npcmblocks);
+
+ if (x96_synth) {
+ // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
+ // (47.6 - 48.0 kHz) components of interpolation image
+ s->dcadsp->lfe_x96_float(output_samples[DCA_SPEAKER_LFE1],
+ samples, &s->output_history_lfe_float,
+ nsamples / 2);
+ }
+
+ // Update LFE history
+ for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
+ s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
+ }
+
+ // Undo embedded XCH downmix
+ if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
+ && s->audio_mode >= AMODE_2F2R) {
+ s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls],
+ output_samples[DCA_SPEAKER_Cs],
+ -M_SQRT1_2, nsamples);
+ s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Rs],
+ output_samples[DCA_SPEAKER_Cs],
+ -M_SQRT1_2, nsamples);
+ }
+
+ // Undo embedded XXCH downmix
+ if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
+ && s->xxch_dmix_embedded) {
+ float scale_inv = s->xxch_dmix_scale_inv * (1.0f / (1 << 16));
+ int *coeff_ptr = s->xxch_dmix_coeff;
+ int xch_base = ff_dca_channels[s->audio_mode];
+ av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
+
+ // Undo downmix
+ for (ch = xch_base; ch < s->nchannels; ch++) {
+ int src_spkr = map_prm_ch_to_spkr(s, ch);
+ if (src_spkr < 0)
+ return AVERROR(EINVAL);
+ for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
+ if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
+ int coeff = *coeff_ptr++;
+ if (coeff) {
+ s->float_dsp->vector_fmac_scalar(output_samples[ spkr],
+ output_samples[src_spkr],
+ coeff * (-1.0f / (1 << 15)),
+ nsamples);
+ }
+ }
+ }
+ }
+
+ // Undo embedded core downmix pre-scaling
+ for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
+ if (s->xxch_core_mask & (1U << spkr)) {
+ s->float_dsp->vector_fmul_scalar(output_samples[spkr],
+ output_samples[spkr],
+ scale_inv, nsamples);
+ }
+ }
+ }
+
+ if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
+ // Front sum/difference decoding
+ if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
+ || s->audio_mode == AMODE_STEREO_SUMDIFF) {
+ s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L],
+ output_samples[DCA_SPEAKER_R],
+ nsamples);
+ }
+
+ // Surround sum/difference decoding
+ if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
+ s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls],
+ output_samples[DCA_SPEAKER_Rs],
+ nsamples);
+ }
+ }
+
+ // Downmix primary channel set to stereo
+ if (s->request_mask != s->ch_mask) {
+ ff_dca_downmix_to_stereo_float(s->float_dsp, output_samples,
+ s->prim_dmix_coeff,
+ nsamples, s->ch_mask);
+ }
+
+ return 0;
+}
+
+int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
+{
+ AVCodecContext *avctx = s->avctx;
+ DCAContext *dca = avctx->priv_data;
+ DCAExssAsset *asset = &dca->exss.assets[0];
+ enum AVMatrixEncoding matrix_encoding;
+ int ret;
+
+ // Handle downmixing to stereo request
+ if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
+ && s->audio_mode > AMODE_MONO && s->prim_dmix_embedded
+ && (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo ||
+ s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
+ s->request_mask = DCA_SPEAKER_LAYOUT_STEREO;
+ else
+ s->request_mask = s->ch_mask;
+ if (!ff_dca_set_channel_layout(avctx, s->ch_remap, s->request_mask))
+ return AVERROR(EINVAL);
+
+ // Force fixed point mode when falling back from XLL
+ if ((avctx->flags & AV_CODEC_FLAG_BITEXACT) || ((dca->packet & DCA_PACKET_EXSS)
+ && (asset->extension_mask & DCA_EXSS_XLL)))
+ ret = filter_frame_fixed(s, frame);
+ else
+ ret = filter_frame_float(s, frame);
+ if (ret < 0)
+ return ret;
+
+ // Set profile, bit rate, etc
+ if (s->ext_audio_mask & DCA_EXSS_MASK)
+ avctx->profile = FF_PROFILE_DTS_HD_HRA;
+ else if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH))
+ avctx->profile = FF_PROFILE_DTS_ES;
+ else if (s->ext_audio_mask & DCA_CSS_X96)
+ avctx->profile = FF_PROFILE_DTS_96_24;
+ else
+ avctx->profile = FF_PROFILE_DTS;
+
+ if (s->bit_rate > 3 && !(s->ext_audio_mask & DCA_EXSS_MASK))
+ avctx->bit_rate = s->bit_rate;
+ else
+ avctx->bit_rate = 0;
+
+ if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
+ s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
+ else
+ matrix_encoding = AV_MATRIX_ENCODING_NONE;
+ if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
+ return ret;
+
+ return 0;
+}
+
+av_cold void ff_dca_core_flush(DCACoreDecoder *s)
+{
+ if (s->subband_buffer) {
+ erase_adpcm_history(s);
+ memset(s->lfe_samples, 0, DCA_LFE_HISTORY * sizeof(int32_t));
+ }
+
+ if (s->x96_subband_buffer)
+ erase_x96_adpcm_history(s);
+
+ erase_dsp_history(s);
+}
+
+av_cold int ff_dca_core_init(DCACoreDecoder *s)
+{
+ dca_init_vlcs();
+
+ if (!(s->float_dsp = avpriv_float_dsp_alloc(0)))
+ return -1;
+ if (!(s->fixed_dsp = avpriv_alloc_fixed_dsp(0)))
+ return -1;
+
+ ff_dcadct_init(&s->dcadct);
+ if (ff_mdct_init(&s->imdct[0], 6, 1, 1.0) < 0)
+ return -1;
+ if (ff_mdct_init(&s->imdct[1], 7, 1, 1.0) < 0)
+ return -1;
+ ff_synth_filter_init(&s->synth);
+
+ s->x96_rand = 1;
+ return 0;
+}
+
+av_cold void ff_dca_core_close(DCACoreDecoder *s)
+{
+ av_freep(&s->float_dsp);
+ av_freep(&s->fixed_dsp);
+
+ ff_mdct_end(&s->imdct[0]);
+ ff_mdct_end(&s->imdct[1]);
+
+ av_freep(&s->subband_buffer);
+ s->subband_size = 0;
+
+ av_freep(&s->x96_subband_buffer);
+ s->x96_subband_size = 0;
+
+ av_freep(&s->output_buffer);
+ s->output_size = 0;
+}
diff --git a/libavcodec/dca_core.h b/libavcodec/dca_core.h
new file mode 100644
index 0000000..112b72b
--- /dev/null
+++ b/libavcodec/dca_core.h
@@ -0,0 +1,206 @@
+/*
+ * Copyright (C) 2016 foo86
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_DCA_CORE_H
+#define AVCODEC_DCA_CORE_H
+
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/fixed_dsp.h"
+#include "libavutil/mem.h"
+
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "dca.h"
+#include "dca_exss.h"
+#include "dcadsp.h"
+#include "dcadct.h"
+#include "fft.h"
+#include "synth_filter.h"
+
+#define DCA_CHANNELS 7
+#define DCA_SUBBANDS 32
+#define DCA_SUBBANDS_X96 64
+#define DCA_SUBFRAMES 16
+#define DCA_SUBBAND_SAMPLES 8
+#define DCA_PCMBLOCK_SAMPLES 32
+#define DCA_ADPCM_COEFFS 4
+#define DCA_LFE_HISTORY 8
+#define DCA_CODE_BOOKS 10
+#define DCA_ABITS_MAX 26
+
+#define DCA_CORE_CHANNELS_MAX 6
+#define DCA_DMIX_CHANNELS_MAX 4
+#define DCA_XXCH_CHANNELS_MAX 2
+#define DCA_EXSS_CHANNELS_MAX 8
+#define DCA_EXSS_CHSETS_MAX 4
+
+#define DCA_FILTER_MODE_X96 0x01
+#define DCA_FILTER_MODE_FIXED 0x02
+
+typedef struct DCADSPData {
+ union {
+ struct {
+ DECLARE_ALIGNED(32, float, hist1)[1024];
+ DECLARE_ALIGNED(32, float, hist2)[64];
+ } flt;
+ struct {
+ DECLARE_ALIGNED(32, int32_t, hist1)[1024];
+ DECLARE_ALIGNED(32, int32_t, hist2)[64];
+ } fix;
+ } u;
+ int offset;
+} DCADSPData;
+
+typedef struct DCACoreDecoder {
+ AVCodecContext *avctx;
+ GetBitContext gb;
+
+ // Bit stream header
+ int crc_present; ///< CRC present flag
+ int npcmblocks; ///< Number of PCM sample blocks
+ int frame_size; ///< Primary frame byte size
+ int audio_mode; ///< Audio channel arrangement
+ int sample_rate; ///< Core audio sampling frequency
+ int bit_rate; ///< Transmission bit rate
+ int drc_present; ///< Embedded dynamic range flag
+ int ts_present; ///< Embedded time stamp flag
+ int aux_present; ///< Auxiliary data flag
+ int ext_audio_type; ///< Extension audio descriptor flag
+ int ext_audio_present; ///< Extended coding flag
+ int sync_ssf; ///< Audio sync word insertion flag
+ int lfe_present; ///< Low frequency effects flag
+ int predictor_history; ///< Predictor history flag switch
+ int filter_perfect; ///< Multirate interpolator switch
+ int source_pcm_res; ///< Source PCM resolution
+ int es_format; ///< Extended surround (ES) mastering flag
+ int sumdiff_front; ///< Front sum/difference flag
+ int sumdiff_surround; ///< Surround sum/difference flag
+
+ // Primary audio coding header
+ int nsubframes; ///< Number of subframes
+ int nchannels; ///< Number of primary audio channels (incl. extension channels)
+ int ch_mask; ///< Speaker layout mask (incl. LFE and extension channels)
+ int8_t nsubbands[DCA_CHANNELS]; ///< Subband activity count
+ int8_t subband_vq_start[DCA_CHANNELS]; ///< High frequency VQ start subband
+ int8_t joint_intensity_index[DCA_CHANNELS]; ///< Joint intensity coding index
+ int8_t transition_mode_sel[DCA_CHANNELS]; ///< Transient mode code book
+ int8_t scale_factor_sel[DCA_CHANNELS]; ///< Scale factor code book
+ int8_t bit_allocation_sel[DCA_CHANNELS]; ///< Bit allocation quantizer select
+ int8_t quant_index_sel[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Quantization index codebook select
+ int32_t scale_factor_adj[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Scale factor adjustment
+
+ // Primary audio coding side information
+ int8_t nsubsubframes[DCA_SUBFRAMES]; ///< Subsubframe count for each subframe
+ int8_t prediction_mode[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction mode
+ int16_t prediction_vq_index[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction coefficients VQ address
+ int8_t bit_allocation[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Bit allocation index
+ int8_t transition_mode[DCA_SUBFRAMES][DCA_CHANNELS][DCA_SUBBANDS]; ///< Transition mode
+ int32_t scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2]; ///< Scale factors (2x for transients and X96)
+ int8_t joint_scale_sel[DCA_CHANNELS]; ///< Joint subband codebook select
+ int32_t joint_scale_factors[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Scale factors for joint subband coding
+
+ // Auxiliary data
+ int prim_dmix_embedded; ///< Auxiliary dynamic downmix flag
+ int prim_dmix_type; ///< Auxiliary primary channel downmix type
+ int prim_dmix_coeff[DCA_DMIX_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Dynamic downmix code coefficients
+
+ // Core extensions
+ int ext_audio_mask; ///< Bit mask of fully decoded core extensions
+
+ // XCH extension data
+ int xch_pos; ///< Bit position of XCH frame in core substream
+
+ // XXCH extension data
+ int xxch_crc_present; ///< CRC presence flag for XXCH channel set header
+ int xxch_mask_nbits; ///< Number of bits for loudspeaker mask
+ int xxch_core_mask; ///< Core loudspeaker activity mask
+ int xxch_spkr_mask; ///< Loudspeaker layout mask
+ int xxch_dmix_embedded; ///< Downmix already performed by encoder
+ int xxch_dmix_scale_inv; ///< Downmix scale factor
+ int xxch_dmix_mask[DCA_XXCH_CHANNELS_MAX]; ///< Downmix channel mapping mask
+ int xxch_dmix_coeff[DCA_XXCH_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Downmix coefficients
+ int xxch_pos; ///< Bit position of XXCH frame in core substream
+
+ // X96 extension data
+ int x96_rev_no; ///< X96 revision number
+ int x96_crc_present; ///< CRC presence flag for X96 channel set header
+ int x96_nchannels; ///< Number of primary channels in X96 extension
+ int x96_high_res; ///< X96 high resolution flag
+ int x96_subband_start; ///< First encoded subband in X96 extension
+ int x96_rand; ///< Random seed for generating samples for unallocated X96 subbands
+ int x96_pos; ///< Bit position of X96 frame in core substream
+
+ // Sample buffers
+ unsigned int x96_subband_size;
+ int32_t *x96_subband_buffer; ///< X96 subband sample buffer base
+ int32_t *x96_subband_samples[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< X96 subband samples
+
+ unsigned int subband_size;
+ int32_t *subband_buffer; ///< Subband sample buffer base
+ int32_t *subband_samples[DCA_CHANNELS][DCA_SUBBANDS]; ///< Subband samples
+ int32_t *lfe_samples; ///< Decimated LFE samples
+
+ // DSP contexts
+ DCADSPData dcadsp_data[DCA_CHANNELS]; ///< FIR history buffers
+ DCADSPContext *dcadsp;
+ DCADCTContext dcadct;
+ FFTContext imdct[2];
+ SynthFilterContext synth;
+ AVFloatDSPContext *float_dsp;
+ AVFixedDSPContext *fixed_dsp;
+
+ // PCM output data
+ unsigned int output_size;
+ void *output_buffer; ///< PCM output buffer base
+ int32_t *output_samples[DCA_SPEAKER_COUNT]; ///< PCM output for fixed point mode
+ int32_t output_history_lfe_fixed; ///< LFE PCM history for X96 filter
+ float output_history_lfe_float; ///< LFE PCM history for X96 filter
+
+ int ch_remap[DCA_SPEAKER_COUNT]; ///< Channel to speaker map
+ int request_mask; ///< Requested channel layout (for stereo downmix)
+
+ int npcmsamples; ///< Number of PCM samples per channel
+ int output_rate; ///< Output sample rate (1x or 2x header rate)
+
+ int filter_mode; ///< Previous filtering mode for detecting changes
+} DCACoreDecoder;
+
+static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
+{
+ if (core->ch_mask & (1U << spkr))
+ return spkr;
+ if (spkr == DCA_SPEAKER_Lss && (core->ch_mask & DCA_SPEAKER_MASK_Ls))
+ return DCA_SPEAKER_Ls;
+ if (spkr == DCA_SPEAKER_Rss && (core->ch_mask & DCA_SPEAKER_MASK_Rs))
+ return DCA_SPEAKER_Rs;
+ return -1;
+}
+
+int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size);
+int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset);
+int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth);
+int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame);
+av_cold void ff_dca_core_flush(DCACoreDecoder *s);
+av_cold int ff_dca_core_init(DCACoreDecoder *s);
+av_cold void ff_dca_core_close(DCACoreDecoder *s);
+
+#endif
--
2.1.4
More information about the ffmpeg-devel
mailing list