[FFmpeg-cvslog] avcodec/dca: add new decoder based on libdcadec
foo86
git at videolan.org
Sun Jan 31 17:14:14 CET 2016
ffmpeg | branch: master | foo86 <foobaz86 at gmail.com> | Sat Jan 16 11:54:38 2016 +0300| [ae5b2c52501d5009fe712334428138a9b758849b] | committer: Hendrik Leppkes
avcodec/dca: add new decoder based on libdcadec
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ae5b2c52501d5009fe712334428138a9b758849b
---
Changelog | 1 +
configure | 1 +
libavcodec/Makefile | 3 +
libavcodec/aarch64/Makefile | 4 +-
libavcodec/allcodecs.c | 2 +-
libavcodec/arm/Makefile | 6 +-
libavcodec/dca_core.c | 2603 +++++++++++++++++++++++++++++++++++++++++++
libavcodec/dca_core.h | 206 ++++
libavcodec/dca_exss.c | 514 +++++++++
libavcodec/dca_exss.h | 92 ++
libavcodec/dca_xll.c | 1499 +++++++++++++++++++++++++
libavcodec/dca_xll.h | 149 +++
libavcodec/dcadec.c | 417 +++++++
libavcodec/dcadec.h | 80 ++
libavcodec/dcadsp.c | 413 +++++++
libavcodec/dcadsp.h | 91 ++
libavcodec/version.h | 2 +-
libavcodec/x86/Makefile | 4 +-
tests/checkasm/Makefile | 2 +-
tests/checkasm/checkasm.c | 4 +-
tests/fate/acodec.mak | 4 +-
tests/fate/audio.mak | 4 +-
22 files changed, 6085 insertions(+), 16 deletions(-)
diff --git a/Changelog b/Changelog
index 2d7ad06..5b06ff1 100644
--- a/Changelog
+++ b/Changelog
@@ -61,6 +61,7 @@ version <next>:
- support for dvaudio in wav and avi
- libaacplus and libvo-aacenc support removed
- Cineform HD decoder
+- new DCA decoder with full support for DTS-HD extensions
version 2.8:
diff --git a/configure b/configure
index 66e1139..d7029c3 100755
--- a/configure
+++ b/configure
@@ -2271,6 +2271,7 @@ comfortnoise_encoder_select="lpc"
cook_decoder_select="audiodsp mdct sinewin"
cscd_decoder_select="lzo"
cscd_decoder_suggest="zlib"
+dca_decoder_select="mdct"
dds_decoder_select="texturedsp"
dirac_decoder_select="dirac_parse dwt golomb videodsp mpegvideoenc"
dnxhd_decoder_select="blockdsp idctdsp"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 1ad2e93..a89fb11 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -222,6 +222,9 @@ OBJS-$(CONFIG_COMFORTNOISE_ENCODER) += cngenc.o
OBJS-$(CONFIG_CPIA_DECODER) += cpia.o
OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
+OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o \
+ dca_core.o dca_exss.o dca_xll.o \
+ dcadsp.o dcadct.o synth_filter.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o
OBJS-$(CONFIG_DDS_DECODER) += dds.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o \
diff --git a/libavcodec/aarch64/Makefile b/libavcodec/aarch64/Makefile
index 803f55b..fd89035 100644
--- a/libavcodec/aarch64/Makefile
+++ b/libavcodec/aarch64/Makefile
@@ -1,4 +1,4 @@
-#OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_init.o
+OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_init.o
OBJS-$(CONFIG_FFT) += aarch64/fft_init_aarch64.o
OBJS-$(CONFIG_FMTCONVERT) += aarch64/fmtconvert_init.o
OBJS-$(CONFIG_H264CHROMA) += aarch64/h264chroma_init_aarch64.o
@@ -17,7 +17,7 @@ OBJS-$(CONFIG_VORBIS_DECODER) += aarch64/vorbisdsp_init.o
ARMV8-OBJS-$(CONFIG_VIDEODSP) += aarch64/videodsp.o
-#NEON-OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_neon.o
+NEON-OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_neon.o
NEON-OBJS-$(CONFIG_FFT) += aarch64/fft_neon.o
NEON-OBJS-$(CONFIG_FMTCONVERT) += aarch64/fmtconvert_neon.o
NEON-OBJS-$(CONFIG_H264CHROMA) += aarch64/h264cmc_neon.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index b174729..c7c1af5 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -391,7 +391,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER(BMV_AUDIO, bmv_audio);
REGISTER_DECODER(COOK, cook);
- REGISTER_ENCODER(DCA, dca);
+ REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER(DSD_LSBF, dsd_lsbf);
REGISTER_DECODER(DSD_MSBF, dsd_msbf);
REGISTER_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar);
diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile
index b2f5a5a..179c403 100644
--- a/libavcodec/arm/Makefile
+++ b/libavcodec/arm/Makefile
@@ -36,7 +36,7 @@ OBJS-$(CONFIG_VP8DSP) += arm/vp8dsp_init_arm.o
# decoders/encoders
OBJS-$(CONFIG_AAC_DECODER) += arm/aacpsdsp_init_arm.o \
arm/sbrdsp_init_arm.o
-#OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_init_arm.o
+OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_init_arm.o
OBJS-$(CONFIG_HEVC_DECODER) += arm/hevcdsp_init_arm.o
OBJS-$(CONFIG_MLP_DECODER) += arm/mlpdsp_init_arm.o
OBJS-$(CONFIG_RV40_DECODER) += arm/rv40dsp_init_arm.o
@@ -87,7 +87,7 @@ VFP-OBJS-$(CONFIG_FMTCONVERT) += arm/fmtconvert_vfp.o
VFP-OBJS-$(CONFIG_MDCT) += arm/mdct_vfp.o
# decoders/encoders
-#VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_vfp.o
+VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_vfp.o
# NEON optimizations
@@ -126,7 +126,7 @@ NEON-OBJS-$(CONFIG_VP8DSP) += arm/vp8dsp_init_neon.o \
NEON-OBJS-$(CONFIG_AAC_DECODER) += arm/aacpsdsp_neon.o \
arm/sbrdsp_neon.o
NEON-OBJS-$(CONFIG_LLAUDDSP) += arm/lossless_audiodsp_neon.o
-#NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_neon.o
+NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_neon.o
NEON-OBJS-$(CONFIG_HEVC_DECODER) += arm/hevcdsp_init_neon.o \
arm/hevcdsp_deblock_neon.o \
arm/hevcdsp_idct_neon.o \
diff --git a/libavcodec/dca_core.c b/libavcodec/dca_core.c
new file mode 100644
index 0000000..94f0f3d
--- /dev/null
+++ b/libavcodec/dca_core.c
@@ -0,0 +1,2603 @@
+/*
+ * 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");
+ if (s->avctx->err_recognition & AV_EF_EXPLODE)
+ 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
diff --git a/libavcodec/dca_exss.c b/libavcodec/dca_exss.c
new file mode 100644
index 0000000..4579f23
--- /dev/null
+++ b/libavcodec/dca_exss.c
@@ -0,0 +1,514 @@
+/*
+ * 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"
+
+static int count_chs_for_mask(int mask)
+{
+ return av_popcount(mask) + av_popcount(mask & 0xae66);
+}
+
+static void parse_xll_parameters(DCAExssParser *s, DCAExssAsset *asset)
+{
+ // Size of XLL data in extension substream
+ asset->xll_size = get_bits(&s->gb, s->exss_size_nbits) + 1;
+
+ // XLL sync word present flag
+ if (asset->xll_sync_present = get_bits1(&s->gb)) {
+ int xll_delay_nbits;
+
+ // Peak bit rate smoothing buffer size
+ skip_bits(&s->gb, 4);
+
+ // Number of bits for XLL decoding delay
+ xll_delay_nbits = get_bits(&s->gb, 5) + 1;
+
+ // Initial XLL decoding delay in frames
+ asset->xll_delay_nframes = get_bits_long(&s->gb, xll_delay_nbits);
+
+ // Number of bytes offset to XLL sync
+ asset->xll_sync_offset = get_bits(&s->gb, s->exss_size_nbits);
+ } else {
+ asset->xll_delay_nframes = 0;
+ asset->xll_sync_offset = 0;
+ }
+}
+
+static void parse_lbr_parameters(DCAExssParser *s, DCAExssAsset *asset)
+{
+ // Size of LBR component in extension substream
+ asset->lbr_size = get_bits(&s->gb, 14) + 1;
+
+ // LBR sync word present flag
+ if (get_bits1(&s->gb))
+ // LBR sync distance
+ skip_bits(&s->gb, 2);
+}
+
+static int parse_descriptor(DCAExssParser *s, DCAExssAsset *asset)
+{
+ int i, j, drc_present, descr_size, descr_pos = get_bits_count(&s->gb);
+
+ // Size of audio asset descriptor in bytes
+ descr_size = get_bits(&s->gb, 9) + 1;
+
+ // Audio asset identifier
+ asset->asset_index = get_bits(&s->gb, 3);
+
+ //
+ // Per stream static metadata
+ //
+
+ if (s->static_fields_present) {
+ // Asset type descriptor presence
+ if (get_bits1(&s->gb))
+ // Asset type descriptor
+ skip_bits(&s->gb, 4);
+
+ // Language descriptor presence
+ if (get_bits1(&s->gb))
+ // Language descriptor
+ skip_bits(&s->gb, 24);
+
+ // Additional textual information presence
+ if (get_bits1(&s->gb)) {
+ // Byte size of additional text info
+ int text_size = get_bits(&s->gb, 10) + 1;
+
+ // Sanity check available size
+ if (get_bits_left(&s->gb) < text_size * 8)
+ return AVERROR_INVALIDDATA;
+
+ // Additional textual information string
+ skip_bits_long(&s->gb, text_size * 8);
+ }
+
+ // PCM bit resolution
+ asset->pcm_bit_res = get_bits(&s->gb, 5) + 1;
+
+ // Maximum sample rate
+ asset->max_sample_rate = ff_dca_sampling_freqs[get_bits(&s->gb, 4)];
+
+ // Total number of channels
+ asset->nchannels_total = get_bits(&s->gb, 8) + 1;
+
+ // One to one map channel to speakers
+ if (asset->one_to_one_map_ch_to_spkr = get_bits1(&s->gb)) {
+ int spkr_mask_nbits = 0;
+ int spkr_remap_nsets;
+ int nspeakers[8];
+
+ // Embedded stereo flag
+ if (asset->nchannels_total > 2)
+ asset->embedded_stereo = get_bits1(&s->gb);
+
+ // Embedded 6 channels flag
+ if (asset->nchannels_total > 6)
+ asset->embedded_6ch = get_bits1(&s->gb);
+
+ // Speaker mask enabled flag
+ if (asset->spkr_mask_enabled = get_bits1(&s->gb)) {
+ // Number of bits for speaker activity mask
+ spkr_mask_nbits = (get_bits(&s->gb, 2) + 1) << 2;
+
+ // Loudspeaker activity mask
+ asset->spkr_mask = get_bits(&s->gb, spkr_mask_nbits);
+ }
+
+ // Number of speaker remapping sets
+ if ((spkr_remap_nsets = get_bits(&s->gb, 3)) && !spkr_mask_nbits) {
+ av_log(s->avctx, AV_LOG_ERROR, "Speaker mask disabled yet there are remapping sets\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Standard loudspeaker layout mask
+ for (i = 0; i < spkr_remap_nsets; i++)
+ nspeakers[i] = count_chs_for_mask(get_bits(&s->gb, spkr_mask_nbits));
+
+ for (i = 0; i < spkr_remap_nsets; i++) {
+ // Number of channels to be decoded for speaker remapping
+ int nch_for_remaps = get_bits(&s->gb, 5) + 1;
+
+ for (j = 0; j < nspeakers[i]; j++) {
+ // Decoded channels to output speaker mapping mask
+ int remap_ch_mask = get_bits_long(&s->gb, nch_for_remaps);
+
+ // Loudspeaker remapping codes
+ skip_bits_long(&s->gb, av_popcount(remap_ch_mask) * 5);
+ }
+ }
+ } else {
+ asset->embedded_stereo = 0;
+ asset->embedded_6ch = 0;
+ asset->spkr_mask_enabled = 0;
+ asset->spkr_mask = 0;
+
+ // Representation type
+ asset->representation_type = get_bits(&s->gb, 3);
+ }
+ }
+
+ //
+ // DRC, DNC and mixing metadata
+ //
+
+ // Dynamic range coefficient presence flag
+ drc_present = get_bits1(&s->gb);
+
+ // Code for dynamic range coefficient
+ if (drc_present)
+ skip_bits(&s->gb, 8);
+
+ // Dialog normalization presence flag
+ if (get_bits1(&s->gb))
+ // Dialog normalization code
+ skip_bits(&s->gb, 5);
+
+ // DRC for stereo downmix
+ if (drc_present && asset->embedded_stereo)
+ skip_bits(&s->gb, 8);
+
+ // Mixing metadata presence flag
+ if (s->mix_metadata_enabled && get_bits1(&s->gb)) {
+ int nchannels_dmix;
+
+ // External mixing flag
+ skip_bits1(&s->gb);
+
+ // Post mixing / replacement gain adjustment
+ skip_bits(&s->gb, 6);
+
+ // DRC prior to mixing
+ if (get_bits(&s->gb, 2) == 3)
+ // Custom code for mixing DRC
+ skip_bits(&s->gb, 8);
+ else
+ // Limit for mixing DRC
+ skip_bits(&s->gb, 3);
+
+ // Scaling type for channels of main audio
+ // Scaling parameters of main audio
+ if (get_bits1(&s->gb))
+ for (i = 0; i < s->nmixoutconfigs; i++)
+ skip_bits_long(&s->gb, 6 * s->nmixoutchs[i]);
+ else
+ skip_bits_long(&s->gb, 6 * s->nmixoutconfigs);
+
+ nchannels_dmix = asset->nchannels_total;
+ if (asset->embedded_6ch)
+ nchannels_dmix += 6;
+ if (asset->embedded_stereo)
+ nchannels_dmix += 2;
+
+ for (i = 0; i < s->nmixoutconfigs; i++) {
+ if (!s->nmixoutchs[i]) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid speaker layout mask for mixing configuration\n");
+ return AVERROR_INVALIDDATA;
+ }
+ for (j = 0; j < nchannels_dmix; j++) {
+ // Mix output mask
+ int mix_map_mask = get_bits(&s->gb, s->nmixoutchs[i]);
+
+ // Mixing coefficients
+ skip_bits_long(&s->gb, av_popcount(mix_map_mask) * 6);
+ }
+ }
+ }
+
+ //
+ // Decoder navigation data
+ //
+
+ // Coding mode for the asset
+ asset->coding_mode = get_bits(&s->gb, 2);
+
+ // Coding components used in asset
+ switch (asset->coding_mode) {
+ case 0: // Coding mode that may contain multiple coding components
+ asset->extension_mask = get_bits(&s->gb, 12);
+
+ if (asset->extension_mask & DCA_EXSS_CORE) {
+ // Size of core component in extension substream
+ asset->core_size = get_bits(&s->gb, 14) + 1;
+ // Core sync word present flag
+ if (get_bits1(&s->gb))
+ // Core sync distance
+ skip_bits(&s->gb, 2);
+ }
+
+ if (asset->extension_mask & DCA_EXSS_XBR)
+ // Size of XBR extension in extension substream
+ asset->xbr_size = get_bits(&s->gb, 14) + 1;
+
+ if (asset->extension_mask & DCA_EXSS_XXCH)
+ // Size of XXCH extension in extension substream
+ asset->xxch_size = get_bits(&s->gb, 14) + 1;
+
+ if (asset->extension_mask & DCA_EXSS_X96)
+ // Size of X96 extension in extension substream
+ asset->x96_size = get_bits(&s->gb, 12) + 1;
+
+ if (asset->extension_mask & DCA_EXSS_LBR)
+ parse_lbr_parameters(s, asset);
+
+ if (asset->extension_mask & DCA_EXSS_XLL)
+ parse_xll_parameters(s, asset);
+
+ if (asset->extension_mask & DCA_EXSS_RSV1)
+ skip_bits(&s->gb, 16);
+
+ if (asset->extension_mask & DCA_EXSS_RSV2)
+ skip_bits(&s->gb, 16);
+ break;
+
+ case 1: // Loss-less coding mode without CBR component
+ asset->extension_mask = DCA_EXSS_XLL;
+ parse_xll_parameters(s, asset);
+ break;
+
+ case 2: // Low bit rate mode
+ asset->extension_mask = DCA_EXSS_LBR;
+ parse_lbr_parameters(s, asset);
+ break;
+
+ case 3: // Auxiliary coding mode
+ asset->extension_mask = 0;
+
+ // Size of auxiliary coded data
+ skip_bits(&s->gb, 14);
+
+ // Auxiliary codec identification
+ skip_bits(&s->gb, 8);
+
+ // Aux sync word present flag
+ if (get_bits1(&s->gb))
+ // Aux sync distance
+ skip_bits(&s->gb, 3);
+ break;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_XLL)
+ // DTS-HD stream ID
+ asset->hd_stream_id = get_bits(&s->gb, 3);
+
+ // One to one mixing flag
+ // Per channel main audio scaling flag
+ // Main audio scaling codes
+ // Decode asset in secondary decoder flag
+ // Revision 2 DRC metadata
+ // Reserved
+ // Zero pad
+ if (ff_dca_seek_bits(&s->gb, descr_pos + descr_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of EXSS asset descriptor\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int set_exss_offsets(DCAExssAsset *asset)
+{
+ int offs = asset->asset_offset;
+ int size = asset->asset_size;
+
+ if (asset->extension_mask & DCA_EXSS_CORE) {
+ asset->core_offset = offs;
+ if (asset->core_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->core_size;
+ size -= asset->core_size;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_XBR) {
+ asset->xbr_offset = offs;
+ if (asset->xbr_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->xbr_size;
+ size -= asset->xbr_size;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_XXCH) {
+ asset->xxch_offset = offs;
+ if (asset->xxch_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->xxch_size;
+ size -= asset->xxch_size;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_X96) {
+ asset->x96_offset = offs;
+ if (asset->x96_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->x96_size;
+ size -= asset->x96_size;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_LBR) {
+ asset->lbr_offset = offs;
+ if (asset->lbr_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->lbr_size;
+ size -= asset->lbr_size;
+ }
+
+ if (asset->extension_mask & DCA_EXSS_XLL) {
+ asset->xll_offset = offs;
+ if (asset->xll_size > size)
+ return AVERROR_INVALIDDATA;
+ offs += asset->xll_size;
+ size -= asset->xll_size;
+ }
+
+ return 0;
+}
+
+int ff_dca_exss_parse(DCAExssParser *s, uint8_t *data, int size)
+{
+ int i, ret, offset, wide_hdr, header_size;
+
+ if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
+ return ret;
+
+ // Extension substream sync word
+ skip_bits_long(&s->gb, 32);
+
+ // User defined bits
+ skip_bits(&s->gb, 8);
+
+ // Extension substream index
+ s->exss_index = get_bits(&s->gb, 2);
+
+ // Flag indicating short or long header size
+ wide_hdr = get_bits1(&s->gb);
+
+ // Extension substream header length
+ header_size = get_bits(&s->gb, 8 + 4 * wide_hdr) + 1;
+
+ // Check CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, 32 + 8, header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid EXSS header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ s->exss_size_nbits = 16 + 4 * wide_hdr;
+
+ // Number of bytes of extension substream
+ s->exss_size = get_bits(&s->gb, s->exss_size_nbits) + 1;
+ if (s->exss_size > size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Packet too short for EXSS frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Per stream static fields presence flag
+ if (s->static_fields_present = get_bits1(&s->gb)) {
+ int active_exss_mask[8];
+
+ // Reference clock code
+ skip_bits(&s->gb, 2);
+
+ // Extension substream frame duration
+ skip_bits(&s->gb, 3);
+
+ // Timecode presence flag
+ if (get_bits1(&s->gb))
+ // Timecode data
+ skip_bits_long(&s->gb, 36);
+
+ // Number of defined audio presentations
+ s->npresents = get_bits(&s->gb, 3) + 1;
+ if (s->npresents > 1) {
+ avpriv_request_sample(s->avctx, "%d audio presentations", s->npresents);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Number of audio assets in extension substream
+ s->nassets = get_bits(&s->gb, 3) + 1;
+ if (s->nassets > 1) {
+ avpriv_request_sample(s->avctx, "%d audio assets", s->nassets);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Active extension substream mask for audio presentation
+ for (i = 0; i < s->npresents; i++)
+ active_exss_mask[i] = get_bits(&s->gb, s->exss_index + 1);
+
+ // Active audio asset mask
+ for (i = 0; i < s->npresents; i++)
+ skip_bits_long(&s->gb, av_popcount(active_exss_mask[i]) * 8);
+
+ // Mixing metadata enable flag
+ if (s->mix_metadata_enabled = get_bits1(&s->gb)) {
+ int spkr_mask_nbits;
+
+ // Mixing metadata adjustment level
+ skip_bits(&s->gb, 2);
+
+ // Number of bits for mixer output speaker activity mask
+ spkr_mask_nbits = (get_bits(&s->gb, 2) + 1) << 2;
+
+ // Number of mixing configurations
+ s->nmixoutconfigs = get_bits(&s->gb, 2) + 1;
+
+ // Speaker layout mask for mixer output channels
+ for (i = 0; i < s->nmixoutconfigs; i++)
+ s->nmixoutchs[i] = count_chs_for_mask(get_bits(&s->gb, spkr_mask_nbits));
+ }
+ } else {
+ s->npresents = 1;
+ s->nassets = 1;
+ }
+
+ // Size of encoded asset data in bytes
+ offset = header_size;
+ for (i = 0; i < s->nassets; i++) {
+ s->assets[i].asset_offset = offset;
+ s->assets[i].asset_size = get_bits(&s->gb, s->exss_size_nbits) + 1;
+ offset += s->assets[i].asset_size;
+ if (offset > s->exss_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "EXSS asset out of bounds\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Audio asset descriptor
+ for (i = 0; i < s->nassets; i++) {
+ if ((ret = parse_descriptor(s, &s->assets[i])) < 0)
+ return ret;
+ if ((ret = set_exss_offsets(&s->assets[i])) < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid extension size in EXSS asset descriptor\n");
+ return ret;
+ }
+ }
+
+ // Backward compatible core present
+ // Backward compatible core substream index
+ // Backward compatible core asset index
+ // Reserved
+ // Byte align
+ // CRC16 of extension substream header
+ if (ff_dca_seek_bits(&s->gb, header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of EXSS header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
diff --git a/libavcodec/dca_exss.h b/libavcodec/dca_exss.h
new file mode 100644
index 0000000..323063a
--- /dev/null
+++ b/libavcodec/dca_exss.h
@@ -0,0 +1,92 @@
+/*
+ * 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_EXSS_H
+#define AVCODEC_DCA_EXSS_H
+
+#include "libavutil/common.h"
+
+#include "avcodec.h"
+#include "get_bits.h"
+
+typedef struct DCAExssAsset {
+ int asset_offset; ///< Offset to asset data from start of substream
+ int asset_size; ///< Size of encoded asset data
+ int asset_index; ///< Audio asset identifier
+
+ int pcm_bit_res; ///< PCM bit resolution
+ int max_sample_rate; ///< Maximum sample rate
+ int nchannels_total; ///< Total number of channels
+ int one_to_one_map_ch_to_spkr; ///< One to one channel to speaker mapping flag
+ int embedded_stereo; ///< Embedded stereo flag
+ int embedded_6ch; ///< Embedded 6 channels flag
+ int spkr_mask_enabled; ///< Speaker mask enabled flag
+ int spkr_mask; ///< Loudspeaker activity mask
+ int representation_type; ///< Representation type
+
+ int coding_mode; ///< Coding mode for the asset
+ int extension_mask; ///< Coding components used in asset
+
+ int core_offset; ///< Offset to core component from start of substream
+ int core_size; ///< Size of core component in extension substream
+
+ int xbr_offset; ///< Offset to XBR extension from start of substream
+ int xbr_size; ///< Size of XBR extension in extension substream
+
+ int xxch_offset; ///< Offset to XXCH extension from start of substream
+ int xxch_size; ///< Size of XXCH extension in extension substream
+
+ int x96_offset; ///< Offset to X96 extension from start of substream
+ int x96_size; ///< Size of X96 extension in extension substream
+
+ int lbr_offset; ///< Offset to LBR component from start of substream
+ int lbr_size; ///< Size of LBR component in extension substream
+
+ int xll_offset; ///< Offset to XLL data from start of substream
+ int xll_size; ///< Size of XLL data in extension substream
+ int xll_sync_present; ///< XLL sync word present flag
+ int xll_delay_nframes; ///< Initial XLL decoding delay in frames
+ int xll_sync_offset; ///< Number of bytes offset to XLL sync
+
+ int hd_stream_id; ///< DTS-HD stream ID
+} DCAExssAsset;
+
+typedef struct DCAExssParser {
+ AVCodecContext *avctx;
+ GetBitContext gb;
+
+ int exss_index; ///< Extension substream index
+ int exss_size_nbits; ///< Number of bits for extension substream size
+ int exss_size; ///< Number of bytes of extension substream
+
+ int static_fields_present; ///< Per stream static fields presence flag
+ int npresents; ///< Number of defined audio presentations
+ int nassets; ///< Number of audio assets in extension substream
+
+ int mix_metadata_enabled; ///< Mixing metadata enable flag
+ int nmixoutconfigs; ///< Number of mixing configurations
+ int nmixoutchs[4]; ///< Speaker layout mask for mixer output channels
+
+ DCAExssAsset assets[1]; ///< Audio asset descriptors
+} DCAExssParser;
+
+int ff_dca_exss_parse(DCAExssParser *s, uint8_t *data, int size);
+
+#endif
diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
new file mode 100644
index 0000000..cd1af81
--- /dev/null
+++ b/libavcodec/dca_xll.c
@@ -0,0 +1,1499 @@
+/*
+ * 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 "dcamath.h"
+#include "dca_syncwords.h"
+#include "unary.h"
+
+static int get_linear(GetBitContext *gb, int n)
+{
+ unsigned int v = get_bits_long(gb, n);
+ return (v >> 1) ^ -(v & 1);
+}
+
+static int get_rice_un(GetBitContext *gb, int k)
+{
+ unsigned int v = get_unary(gb, 1, 128);
+ return (v << k) | get_bits_long(gb, k);
+}
+
+static int get_rice(GetBitContext *gb, int k)
+{
+ unsigned int v = get_rice_un(gb, k);
+ return (v >> 1) ^ -(v & 1);
+}
+
+static void get_array(GetBitContext *gb, int32_t *array, int size, int n)
+{
+ int i;
+
+ for (i = 0; i < size; i++)
+ array[i] = get_bits(gb, n);
+}
+
+static void get_linear_array(GetBitContext *gb, int32_t *array, int size, int n)
+{
+ int i;
+
+ if (n == 0)
+ memset(array, 0, sizeof(*array) * size);
+ else for (i = 0; i < size; i++)
+ array[i] = get_linear(gb, n);
+}
+
+static void get_rice_array(GetBitContext *gb, int32_t *array, int size, int k)
+{
+ int i;
+
+ for (i = 0; i < size; i++)
+ array[i] = get_rice(gb, k);
+}
+
+static int parse_dmix_coeffs(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ // Size of downmix coefficient matrix
+ int m = c->primary_chset ? ff_dca_dmix_primary_nch[c->dmix_type] : c->hier_ofs;
+ int i, j, *coeff_ptr = c->dmix_coeff;
+
+ for (i = 0; i < m; i++) {
+ int code, sign, coeff, scale, scale_inv = 0;
+ unsigned int index;
+
+ // Downmix scale (only for non-primary channel sets)
+ if (!c->primary_chset) {
+ code = get_bits(&s->gb, 9);
+ sign = (code >> 8) - 1;
+ index = (code & 0xff) - FF_DCA_DMIXTABLE_OFFSET;
+ if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL downmix scale index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ scale = ff_dca_dmixtable[index + FF_DCA_DMIXTABLE_OFFSET];
+ scale_inv = ff_dca_inv_dmixtable[index];
+ c->dmix_scale[i] = (scale ^ sign) - sign;
+ c->dmix_scale_inv[i] = (scale_inv ^ sign) - sign;
+ }
+
+ // Downmix coefficients
+ for (j = 0; j < c->nchannels; j++) {
+ code = get_bits(&s->gb, 9);
+ sign = (code >> 8) - 1;
+ index = code & 0xff;
+ if (index >= FF_DCA_DMIXTABLE_SIZE) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL downmix coefficient index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ coeff = ff_dca_dmixtable[index];
+ if (!c->primary_chset)
+ // Multiply by |InvDmixScale| to get |UndoDmixScale|
+ coeff = mul16(scale_inv, coeff);
+ *coeff_ptr++ = (coeff ^ sign) - sign;
+ }
+ }
+
+ return 0;
+}
+
+static int chs_parse_header(DCAXllDecoder *s, DCAXllChSet *c, DCAExssAsset *asset)
+{
+ int i, j, k, ret, band, header_size, header_pos = get_bits_count(&s->gb);
+ DCAXllChSet *p = &s->chset[0];
+ DCAXllBand *b;
+
+ // Size of channel set sub-header
+ header_size = get_bits(&s->gb, 10) + 1;
+
+ // Check CRC
+ if ((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 XLL sub-header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of channels in the channel set
+ c->nchannels = get_bits(&s->gb, 4) + 1;
+ if (c->nchannels > DCA_XLL_CHANNELS_MAX) {
+ avpriv_request_sample(s->avctx, "%d XLL channels", c->nchannels);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Residual type
+ c->residual_encode = get_bits(&s->gb, c->nchannels);
+
+ // PCM bit resolution
+ c->pcm_bit_res = get_bits(&s->gb, 5) + 1;
+
+ // Storage unit width
+ c->storage_bit_res = get_bits(&s->gb, 5) + 1;
+ if (c->storage_bit_res != 16 && c->storage_bit_res != 24) {
+ avpriv_request_sample(s->avctx, "%d-bit XLL storage resolution", c->storage_bit_res);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (c->pcm_bit_res > c->storage_bit_res) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid PCM bit resolution for XLL channel set (%d > %d)\n", c->pcm_bit_res, c->storage_bit_res);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Original sampling frequency
+ c->freq = ff_dca_sampling_freqs[get_bits(&s->gb, 4)];
+ if (c->freq > 192000) {
+ avpriv_request_sample(s->avctx, "%d Hz XLL sampling frequency", c->freq);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Sampling frequency modifier
+ if (get_bits(&s->gb, 2)) {
+ avpriv_request_sample(s->avctx, "XLL sampling frequency modifier");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Which replacement set this channel set is member of
+ if (get_bits(&s->gb, 2)) {
+ avpriv_request_sample(s->avctx, "XLL replacement set");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (asset->one_to_one_map_ch_to_spkr) {
+ // Primary channel set flag
+ c->primary_chset = get_bits1(&s->gb);
+ if (c->primary_chset != (c == p)) {
+ av_log(s->avctx, AV_LOG_ERROR, "The first (and only) XLL channel set must be primary\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Downmix coefficients present in stream
+ c->dmix_coeffs_present = get_bits1(&s->gb);
+
+ // Downmix already performed by encoder
+ c->dmix_embedded = c->dmix_coeffs_present && get_bits1(&s->gb);
+
+ // Downmix type
+ if (c->dmix_coeffs_present && c->primary_chset) {
+ c->dmix_type = get_bits(&s->gb, 3);
+ if (c->dmix_type >= DCA_DMIX_TYPE_COUNT) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL primary channel set downmix type\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Whether the channel set is part of a hierarchy
+ c->hier_chset = get_bits1(&s->gb);
+ if (!c->hier_chset && s->nchsets != 1) {
+ avpriv_request_sample(s->avctx, "XLL channel set outside of hierarchy");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Downmix coefficients
+ if (c->dmix_coeffs_present && (ret = parse_dmix_coeffs(s, c)) < 0)
+ return ret;
+
+ // Channel mask enabled
+ if (!get_bits1(&s->gb)) {
+ avpriv_request_sample(s->avctx, "Disabled XLL channel mask");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Channel mask for set
+ c->ch_mask = get_bits_long(&s->gb, s->ch_mask_nbits);
+ if (av_popcount(c->ch_mask) != c->nchannels) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL channel mask\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Build the channel to speaker map
+ for (i = 0, j = 0; i < s->ch_mask_nbits; i++)
+ if (c->ch_mask & (1U << i))
+ c->ch_remap[j++] = i;
+ } else {
+ // Mapping coeffs present flag
+ if (c->nchannels != 2 || s->nchsets != 1 || get_bits1(&s->gb)) {
+ avpriv_request_sample(s->avctx, "Custom XLL channel to speaker mapping");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Setup for LtRt decoding
+ c->primary_chset = 1;
+ c->dmix_coeffs_present = 0;
+ c->dmix_embedded = 0;
+ c->hier_chset = 0;
+ c->ch_mask = DCA_SPEAKER_LAYOUT_STEREO;
+ c->ch_remap[0] = DCA_SPEAKER_L;
+ c->ch_remap[1] = DCA_SPEAKER_R;
+ }
+
+ if (c->freq > 96000) {
+ // Extra frequency bands flag
+ if (get_bits1(&s->gb)) {
+ avpriv_request_sample(s->avctx, "Extra XLL frequency bands");
+ return AVERROR_PATCHWELCOME;
+ }
+ c->nfreqbands = 2;
+ } else {
+ c->nfreqbands = 1;
+ }
+
+ // Set the sampling frequency to that of the first frequency band.
+ // Frequency will be doubled again after bands assembly.
+ c->freq >>= c->nfreqbands - 1;
+
+ // Verify that all channel sets have the same audio characteristics
+ if (c != p && (c->nfreqbands != p->nfreqbands || c->freq != p->freq
+ || c->pcm_bit_res != p->pcm_bit_res
+ || c->storage_bit_res != p->storage_bit_res)) {
+ avpriv_request_sample(s->avctx, "Different XLL audio characteristics");
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Determine number of bits to read bit allocation coding parameter
+ if (c->storage_bit_res > 16)
+ c->nabits = 5;
+ else if (c->storage_bit_res > 8)
+ c->nabits = 4;
+ else
+ c->nabits = 3;
+
+ // Account for embedded downmix and decimator saturation
+ if ((s->nchsets > 1 || c->nfreqbands > 1) && c->nabits < 5)
+ c->nabits++;
+
+ for (band = 0, b = c->bands; band < c->nfreqbands; band++, b++) {
+ // Pairwise channel decorrelation
+ if ((b->decor_enabled = get_bits1(&s->gb)) && c->nchannels > 1) {
+ int ch_nbits = av_ceil_log2(c->nchannels);
+
+ // Original channel order
+ for (i = 0; i < c->nchannels; i++) {
+ b->orig_order[i] = get_bits(&s->gb, ch_nbits);
+ if (b->orig_order[i] >= c->nchannels) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL original channel order\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ // Pairwise channel coefficients
+ for (i = 0; i < c->nchannels / 2; i++)
+ b->decor_coeff[i] = get_bits1(&s->gb) ? get_linear(&s->gb, 7) : 0;
+ } else {
+ for (i = 0; i < c->nchannels; i++)
+ b->orig_order[i] = i;
+ for (i = 0; i < c->nchannels / 2; i++)
+ b->decor_coeff[i] = 0;
+ }
+
+ // Adaptive predictor order
+ b->highest_pred_order = 0;
+ for (i = 0; i < c->nchannels; i++) {
+ b->adapt_pred_order[i] = get_bits(&s->gb, 4);
+ if (b->adapt_pred_order[i] > b->highest_pred_order)
+ b->highest_pred_order = b->adapt_pred_order[i];
+ }
+ if (b->highest_pred_order > s->nsegsamples) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL adaptive predicition order\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Fixed predictor order
+ for (i = 0; i < c->nchannels; i++)
+ b->fixed_pred_order[i] = b->adapt_pred_order[i] ? 0 : get_bits(&s->gb, 2);
+
+ // Adaptive predictor quantized reflection coefficients
+ for (i = 0; i < c->nchannels; i++) {
+ for (j = 0; j < b->adapt_pred_order[i]; j++) {
+ k = get_linear(&s->gb, 8);
+ if (k == -128) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL reflection coefficient index\n");
+ return AVERROR_INVALIDDATA;
+ }
+ if (k < 0)
+ b->adapt_refl_coeff[i][j] = -(int)ff_dca_xll_refl_coeff[-k];
+ else
+ b->adapt_refl_coeff[i][j] = (int)ff_dca_xll_refl_coeff[ k];
+ }
+ }
+
+ // Downmix performed by encoder in extension frequency band
+ b->dmix_embedded = c->dmix_embedded && (band == 0 || get_bits1(&s->gb));
+
+ // MSB/LSB split flag in extension frequency band
+ if ((band == 0 && s->scalable_lsbs) || (band != 0 && get_bits1(&s->gb))) {
+ // Size of LSB section in any segment
+ b->lsb_section_size = get_bits_long(&s->gb, s->seg_size_nbits);
+ if (b->lsb_section_size < 0 || b->lsb_section_size > s->frame_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid LSB section size\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Account for optional CRC bytes after LSB section
+ if (b->lsb_section_size && (s->band_crc_present > 2 ||
+ (band == 0 && s->band_crc_present > 1)))
+ b->lsb_section_size += 2;
+
+ // Number of bits to represent the samples in LSB part
+ for (i = 0; i < c->nchannels; i++) {
+ b->nscalablelsbs[i] = get_bits(&s->gb, 4);
+ if (b->nscalablelsbs[i] && !b->lsb_section_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "LSB section missing with non-zero LSB width\n");
+ return AVERROR_INVALIDDATA;
+ }
+ }
+ } else {
+ b->lsb_section_size = 0;
+ for (i = 0; i < c->nchannels; i++)
+ b->nscalablelsbs[i] = 0;
+ }
+
+ // Scalable resolution flag in extension frequency band
+ if ((band == 0 && s->scalable_lsbs) || (band != 0 && get_bits1(&s->gb))) {
+ // Number of bits discarded by authoring
+ for (i = 0; i < c->nchannels; i++)
+ b->bit_width_adjust[i] = get_bits(&s->gb, 4);
+ } else {
+ for (i = 0; i < c->nchannels; i++)
+ b->bit_width_adjust[i] = 0;
+ }
+ }
+
+ // Reserved
+ // Byte align
+ // CRC16 of channel set sub-header
+ if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL sub-header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int chs_alloc_msb_band_data(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ int ndecisamples = c->nfreqbands > 1 ? DCA_XLL_DECI_HISTORY_MAX : 0;
+ int nchsamples = s->nframesamples + ndecisamples;
+ int i, j, nsamples = nchsamples * c->nchannels * c->nfreqbands;
+ int32_t *ptr;
+
+ // Reallocate MSB sample buffer
+ av_fast_malloc(&c->sample_buffer[0], &c->sample_size[0], nsamples * sizeof(int32_t));
+ if (!c->sample_buffer[0])
+ return AVERROR(ENOMEM);
+
+ ptr = c->sample_buffer[0] + ndecisamples;
+ for (i = 0; i < c->nfreqbands; i++) {
+ for (j = 0; j < c->nchannels; j++) {
+ c->bands[i].msb_sample_buffer[j] = ptr;
+ ptr += nchsamples;
+ }
+ }
+
+ return 0;
+}
+
+static int chs_alloc_lsb_band_data(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ int i, j, nsamples = 0;
+ int32_t *ptr;
+
+ // Determine number of frequency bands that have MSB/LSB split
+ for (i = 0; i < c->nfreqbands; i++)
+ if (c->bands[i].lsb_section_size)
+ nsamples += s->nframesamples * c->nchannels;
+ if (!nsamples)
+ return 0;
+
+ // Reallocate LSB sample buffer
+ av_fast_malloc(&c->sample_buffer[1], &c->sample_size[1], nsamples * sizeof(int32_t));
+ if (!c->sample_buffer[1])
+ return AVERROR(ENOMEM);
+
+ ptr = c->sample_buffer[1];
+ for (i = 0; i < c->nfreqbands; i++) {
+ if (c->bands[i].lsb_section_size) {
+ for (j = 0; j < c->nchannels; j++) {
+ c->bands[i].lsb_sample_buffer[j] = ptr;
+ ptr += s->nframesamples;
+ }
+ } else {
+ for (j = 0; j < c->nchannels; j++)
+ c->bands[i].lsb_sample_buffer[j] = NULL;
+ }
+ }
+
+ return 0;
+}
+
+static int chs_parse_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg, int band_data_end)
+{
+ DCAXllBand *b = &c->bands[band];
+ int i, j, k;
+
+ // Start unpacking MSB portion of the segment
+ if (!(seg && get_bits1(&s->gb))) {
+ // Unpack segment type
+ // 0 - distinct coding parameters for each channel
+ // 1 - common coding parameters for all channels
+ c->seg_common = get_bits1(&s->gb);
+
+ // Determine number of coding parameters encoded in segment
+ k = c->seg_common ? 1 : c->nchannels;
+
+ // Unpack Rice coding parameters
+ for (i = 0; i < k; i++) {
+ // Unpack Rice coding flag
+ // 0 - linear code, 1 - Rice code
+ c->rice_code_flag[i] = get_bits1(&s->gb);
+ if (!c->seg_common && c->rice_code_flag[i]) {
+ // Unpack Hybrid Rice coding flag
+ // 0 - Rice code, 1 - Hybrid Rice code
+ if (get_bits1(&s->gb))
+ // Unpack binary code length for isolated samples
+ c->bitalloc_hybrid_linear[i] = get_bits(&s->gb, c->nabits) + 1;
+ else
+ // 0 indicates no Hybrid Rice coding
+ c->bitalloc_hybrid_linear[i] = 0;
+ } else {
+ // 0 indicates no Hybrid Rice coding
+ c->bitalloc_hybrid_linear[i] = 0;
+ }
+ }
+
+ // Unpack coding parameters
+ for (i = 0; i < k; i++) {
+ if (seg == 0) {
+ // Unpack coding parameter for part A of segment 0
+ c->bitalloc_part_a[i] = get_bits(&s->gb, c->nabits);
+
+ // Adjust for the linear code
+ if (!c->rice_code_flag[i] && c->bitalloc_part_a[i])
+ c->bitalloc_part_a[i]++;
+
+ if (!c->seg_common)
+ c->nsamples_part_a[i] = b->adapt_pred_order[i];
+ else
+ c->nsamples_part_a[i] = b->highest_pred_order;
+ } else {
+ c->bitalloc_part_a[i] = 0;
+ c->nsamples_part_a[i] = 0;
+ }
+
+ // Unpack coding parameter for part B of segment
+ c->bitalloc_part_b[i] = get_bits(&s->gb, c->nabits);
+
+ // Adjust for the linear code
+ if (!c->rice_code_flag[i] && c->bitalloc_part_b[i])
+ c->bitalloc_part_b[i]++;
+ }
+ }
+
+ // Unpack entropy codes
+ for (i = 0; i < c->nchannels; i++) {
+ int32_t *part_a, *part_b;
+ int nsamples_part_b;
+
+ // Select index of coding parameters
+ k = c->seg_common ? 0 : i;
+
+ // Slice the segment into parts A and B
+ part_a = b->msb_sample_buffer[i] + seg * s->nsegsamples;
+ part_b = part_a + c->nsamples_part_a[k];
+ nsamples_part_b = s->nsegsamples - c->nsamples_part_a[k];
+
+ if (get_bits_left(&s->gb) < 0)
+ return AVERROR_INVALIDDATA;
+
+ if (!c->rice_code_flag[k]) {
+ // Linear codes
+ // Unpack all residuals of part A of segment 0
+ get_linear_array(&s->gb, part_a, c->nsamples_part_a[k],
+ c->bitalloc_part_a[k]);
+
+ // Unpack all residuals of part B of segment 0 and others
+ get_linear_array(&s->gb, part_b, nsamples_part_b,
+ c->bitalloc_part_b[k]);
+ } else {
+ // Rice codes
+ // Unpack all residuals of part A of segment 0
+ get_rice_array(&s->gb, part_a, c->nsamples_part_a[k],
+ c->bitalloc_part_a[k]);
+
+ if (c->bitalloc_hybrid_linear[k]) {
+ // Hybrid Rice codes
+ // Unpack the number of isolated samples
+ int nisosamples = get_bits(&s->gb, s->nsegsamples_log2);
+
+ // Set all locations to 0
+ memset(part_b, 0, sizeof(*part_b) * nsamples_part_b);
+
+ // Extract the locations of isolated samples and flag by -1
+ for (j = 0; j < nisosamples; j++) {
+ int loc = get_bits(&s->gb, s->nsegsamples_log2);
+ if (loc >= nsamples_part_b) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid isolated sample location\n");
+ return AVERROR_INVALIDDATA;
+ }
+ part_b[loc] = -1;
+ }
+
+ // Unpack all residuals of part B of segment 0 and others
+ for (j = 0; j < nsamples_part_b; j++) {
+ if (part_b[j])
+ part_b[j] = get_linear(&s->gb, c->bitalloc_hybrid_linear[k]);
+ else
+ part_b[j] = get_rice(&s->gb, c->bitalloc_part_b[k]);
+ }
+ } else {
+ // Rice codes
+ // Unpack all residuals of part B of segment 0 and others
+ get_rice_array(&s->gb, part_b, nsamples_part_b, c->bitalloc_part_b[k]);
+ }
+ }
+ }
+
+ // Unpack decimator history for frequency band 1
+ if (seg == 0 && band == 1) {
+ int nbits = get_bits(&s->gb, 5) + 1;
+ for (i = 0; i < c->nchannels; i++)
+ for (j = 1; j < DCA_XLL_DECI_HISTORY_MAX; j++)
+ c->deci_history[i][j] = get_sbits_long(&s->gb, nbits);
+ }
+
+ // Start unpacking LSB portion of the segment
+ if (b->lsb_section_size) {
+ // Skip to the start of LSB portion
+ if (ff_dca_seek_bits(&s->gb, band_data_end - b->lsb_section_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL band data\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Unpack all LSB parts of residuals of this segment
+ for (i = 0; i < c->nchannels; i++) {
+ if (b->nscalablelsbs[i]) {
+ get_array(&s->gb,
+ b->lsb_sample_buffer[i] + seg * s->nsegsamples,
+ s->nsegsamples, b->nscalablelsbs[i]);
+ }
+ }
+ }
+
+ // Skip to the end of band data
+ if (ff_dca_seek_bits(&s->gb, band_data_end)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL band data\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static void av_cold chs_clear_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg)
+{
+ DCAXllBand *b = &c->bands[band];
+ int i, offset, nsamples;
+
+ if (seg < 0) {
+ offset = 0;
+ nsamples = s->nframesamples;
+ } else {
+ offset = seg * s->nsegsamples;
+ nsamples = s->nsegsamples;
+ }
+
+ for (i = 0; i < c->nchannels; i++) {
+ memset(b->msb_sample_buffer[i] + offset, 0, nsamples * sizeof(int32_t));
+ if (b->lsb_section_size)
+ memset(b->lsb_sample_buffer[i] + offset, 0, nsamples * sizeof(int32_t));
+ }
+
+ if (seg <= 0 && band)
+ memset(c->deci_history, 0, sizeof(c->deci_history));
+
+ if (seg < 0) {
+ memset(b->nscalablelsbs, 0, sizeof(b->nscalablelsbs));
+ memset(b->bit_width_adjust, 0, sizeof(b->bit_width_adjust));
+ }
+}
+
+static void chs_filter_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band)
+{
+ DCAXllBand *b = &c->bands[band];
+ int nsamples = s->nframesamples;
+ int i, j, k;
+
+ // Inverse adaptive or fixed prediction
+ for (i = 0; i < c->nchannels; i++) {
+ int32_t *buf = b->msb_sample_buffer[i];
+ int order = b->adapt_pred_order[i];
+ if (order > 0) {
+ int coeff[DCA_XLL_ADAPT_PRED_ORDER_MAX];
+ // Conversion from reflection coefficients to direct form coefficients
+ for (j = 0; j < order; j++) {
+ int rc = b->adapt_refl_coeff[i][j];
+ for (k = 0; k < (j + 1) / 2; k++) {
+ int tmp1 = coeff[ k ];
+ int tmp2 = coeff[j - k - 1];
+ coeff[ k ] = tmp1 + mul16(rc, tmp2);
+ coeff[j - k - 1] = tmp2 + mul16(rc, tmp1);
+ }
+ coeff[j] = rc;
+ }
+ // Inverse adaptive prediction
+ for (j = 0; j < nsamples - order; j++) {
+ int64_t err = 0;
+ for (k = 0; k < order; k++)
+ err += (int64_t)buf[j + k] * coeff[order - k - 1];
+ buf[j + k] -= clip23(norm16(err));
+ }
+ } else {
+ // Inverse fixed coefficient prediction
+ for (j = 0; j < b->fixed_pred_order[i]; j++)
+ for (k = 1; k < nsamples; k++)
+ buf[k] += buf[k - 1];
+ }
+ }
+
+ // Inverse pairwise channel decorrellation
+ if (b->decor_enabled) {
+ int32_t *tmp[DCA_XLL_CHANNELS_MAX];
+
+ for (i = 0; i < c->nchannels / 2; i++) {
+ int coeff = b->decor_coeff[i];
+ if (coeff) {
+ s->dcadsp->decor(b->msb_sample_buffer[i * 2 + 1],
+ b->msb_sample_buffer[i * 2 ],
+ coeff, nsamples);
+ }
+ }
+
+ // Reorder channel pointers to the original order
+ for (i = 0; i < c->nchannels; i++)
+ tmp[i] = b->msb_sample_buffer[i];
+
+ for (i = 0; i < c->nchannels; i++)
+ b->msb_sample_buffer[b->orig_order[i]] = tmp[i];
+ }
+
+ // Map output channel pointers for frequency band 0
+ if (c->nfreqbands == 1)
+ for (i = 0; i < c->nchannels; i++)
+ s->output_samples[c->ch_remap[i]] = b->msb_sample_buffer[i];
+}
+
+static int chs_get_lsb_width(DCAXllDecoder *s, DCAXllChSet *c, int band, int ch)
+{
+ int adj = c->bands[band].bit_width_adjust[ch];
+ int shift = c->bands[band].nscalablelsbs[ch];
+
+ if (s->fixed_lsb_width)
+ shift = s->fixed_lsb_width;
+ else if (shift && adj)
+ shift += adj - 1;
+ else
+ shift += adj;
+
+ return shift;
+}
+
+static void chs_assemble_msbs_lsbs(DCAXllDecoder *s, DCAXllChSet *c, int band)
+{
+ DCAXllBand *b = &c->bands[band];
+ int n, ch, nsamples = s->nframesamples;
+
+ for (ch = 0; ch < c->nchannels; ch++) {
+ int shift = chs_get_lsb_width(s, c, band, ch);
+ if (shift) {
+ int32_t *msb = b->msb_sample_buffer[ch];
+ if (b->nscalablelsbs[ch]) {
+ int32_t *lsb = b->lsb_sample_buffer[ch];
+ int adj = b->bit_width_adjust[ch];
+ for (n = 0; n < nsamples; n++)
+ msb[n] = msb[n] * (1 << shift) + (lsb[n] << adj);
+ } else {
+ for (n = 0; n < nsamples; n++)
+ msb[n] = msb[n] * (1 << shift);
+ }
+ }
+ }
+}
+
+static int chs_assemble_freq_bands(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ int ch, nsamples = s->nframesamples;
+ int32_t *ptr;
+
+ av_assert1(c->nfreqbands > 1);
+
+ // Reallocate frequency band assembly buffer
+ av_fast_malloc(&c->sample_buffer[2], &c->sample_size[2],
+ 2 * nsamples * c->nchannels * sizeof(int32_t));
+ if (!c->sample_buffer[2])
+ return AVERROR(ENOMEM);
+
+ // Assemble frequency bands 0 and 1
+ ptr = c->sample_buffer[2];
+ for (ch = 0; ch < c->nchannels; ch++) {
+ int32_t *band0 = c->bands[0].msb_sample_buffer[ch];
+ int32_t *band1 = c->bands[1].msb_sample_buffer[ch];
+
+ // Copy decimator history
+ memcpy(band0 - DCA_XLL_DECI_HISTORY_MAX,
+ c->deci_history[ch], sizeof(c->deci_history[0]));
+
+ // Filter
+ s->dcadsp->assemble_freq_bands(ptr, band0, band1,
+ ff_dca_xll_band_coeff,
+ nsamples);
+
+ // Remap output channel pointer to assembly buffer
+ s->output_samples[c->ch_remap[ch]] = ptr;
+ ptr += nsamples * 2;
+ }
+
+ return 0;
+}
+
+static int parse_common_header(DCAXllDecoder *s)
+{
+ int stream_ver, header_size, frame_size_nbits, nframesegs_log2;
+
+ // XLL extension sync word
+ if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XLL) {
+ av_log(s->avctx, AV_LOG_VERBOSE, "Invalid XLL sync word\n");
+ return AVERROR(EAGAIN);
+ }
+
+ // Version number
+ stream_ver = get_bits(&s->gb, 4) + 1;
+ if (stream_ver > 1) {
+ avpriv_request_sample(s->avctx, "XLL stream version %d", stream_ver);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Lossless frame header length
+ header_size = get_bits(&s->gb, 8) + 1;
+
+ // Check CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, 32, header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL common header checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of bits used to read frame size
+ frame_size_nbits = get_bits(&s->gb, 5) + 1;
+
+ // Number of bytes in a lossless frame
+ s->frame_size = get_bits_long(&s->gb, frame_size_nbits);
+ if (s->frame_size < 0 || s->frame_size >= DCA_XLL_PBR_BUFFER_MAX) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL frame size (%d bytes)\n", s->frame_size);
+ return AVERROR_INVALIDDATA;
+ }
+ s->frame_size++;
+
+ // Number of channels sets per frame
+ s->nchsets = get_bits(&s->gb, 4) + 1;
+ if (s->nchsets > DCA_XLL_CHSETS_MAX) {
+ avpriv_request_sample(s->avctx, "%d XLL channel sets", s->nchsets);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ // Number of segments per frame
+ nframesegs_log2 = get_bits(&s->gb, 4);
+ s->nframesegs = 1 << nframesegs_log2;
+ if (s->nframesegs > 1024) {
+ av_log(s->avctx, AV_LOG_ERROR, "Too many segments per XLL frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Samples in segment per one frequency band for the first channel set
+ // Maximum value is 256 for sampling frequencies <= 48 kHz
+ // Maximum value is 512 for sampling frequencies > 48 kHz
+ s->nsegsamples_log2 = get_bits(&s->gb, 4);
+ if (!s->nsegsamples_log2) {
+ av_log(s->avctx, AV_LOG_ERROR, "Too few samples per XLL segment\n");
+ return AVERROR_INVALIDDATA;
+ }
+ s->nsegsamples = 1 << s->nsegsamples_log2;
+ if (s->nsegsamples > 512) {
+ av_log(s->avctx, AV_LOG_ERROR, "Too many samples per XLL segment\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Samples in frame per one frequency band for the first channel set
+ s->nframesamples_log2 = s->nsegsamples_log2 + nframesegs_log2;
+ s->nframesamples = 1 << s->nframesamples_log2;
+ if (s->nframesamples > 65536) {
+ av_log(s->avctx, AV_LOG_ERROR, "Too many samples per XLL frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Number of bits used to read segment size
+ s->seg_size_nbits = get_bits(&s->gb, 5) + 1;
+
+ // Presence of CRC16 within each frequency band
+ // 0 - No CRC16 within band
+ // 1 - CRC16 placed at the end of MSB0
+ // 2 - CRC16 placed at the end of MSB0 and LSB0
+ // 3 - CRC16 placed at the end of MSB0 and LSB0 and other frequency bands
+ s->band_crc_present = get_bits(&s->gb, 2);
+
+ // MSB/LSB split flag
+ s->scalable_lsbs = get_bits1(&s->gb);
+
+ // Channel position mask
+ s->ch_mask_nbits = get_bits(&s->gb, 5) + 1;
+
+ // Fixed LSB width
+ if (s->scalable_lsbs)
+ s->fixed_lsb_width = get_bits(&s->gb, 4);
+ else
+ s->fixed_lsb_width = 0;
+
+ // Reserved
+ // Byte align
+ // Header CRC16 protection
+ if (ff_dca_seek_bits(&s->gb, header_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL common header\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int is_hier_dmix_chset(DCAXllChSet *c)
+{
+ return !c->primary_chset && c->dmix_embedded && c->hier_chset;
+}
+
+static DCAXllChSet *find_next_hier_dmix_chset(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ if (c->hier_chset)
+ while (++c < &s->chset[s->nchsets])
+ if (is_hier_dmix_chset(c))
+ return c;
+
+ return NULL;
+}
+
+static void prescale_down_mix(DCAXllChSet *c, DCAXllChSet *o)
+{
+ int i, j, *coeff_ptr = c->dmix_coeff;
+
+ for (i = 0; i < c->hier_ofs; i++) {
+ int scale = o->dmix_scale[i];
+ int scale_inv = o->dmix_scale_inv[i];
+ c->dmix_scale[i] = mul15(c->dmix_scale[i], scale);
+ c->dmix_scale_inv[i] = mul16(c->dmix_scale_inv[i], scale_inv);
+ for (j = 0; j < c->nchannels; j++) {
+ int coeff = mul16(*coeff_ptr, scale_inv);
+ *coeff_ptr++ = mul15(coeff, o->dmix_scale[c->hier_ofs + j]);
+ }
+ }
+}
+
+static int parse_sub_headers(DCAXllDecoder *s, DCAExssAsset *asset)
+{
+ DCAContext *dca = s->avctx->priv_data;
+ DCAXllChSet *c;
+ int i, ret;
+
+ // Parse channel set headers
+ s->nfreqbands = 0;
+ s->nchannels = 0;
+ s->nreschsets = 0;
+ for (i = 0, c = s->chset; i < s->nchsets; i++, c++) {
+ c->hier_ofs = s->nchannels;
+ if ((ret = chs_parse_header(s, c, asset)) < 0)
+ return ret;
+ if (c->nfreqbands > s->nfreqbands)
+ s->nfreqbands = c->nfreqbands;
+ if (c->hier_chset)
+ s->nchannels += c->nchannels;
+ if (c->residual_encode != (1 << c->nchannels) - 1)
+ s->nreschsets++;
+ }
+
+ // Pre-scale downmixing coefficients for all non-primary channel sets
+ for (i = s->nchsets - 1, c = &s->chset[i]; i > 0; i--, c--) {
+ if (is_hier_dmix_chset(c)) {
+ DCAXllChSet *o = find_next_hier_dmix_chset(s, c);
+ if (o)
+ prescale_down_mix(c, o);
+ }
+ }
+
+ // Determine number of active channel sets to decode
+ switch (dca->request_channel_layout) {
+ case DCA_SPEAKER_LAYOUT_STEREO:
+ s->nactivechsets = 1;
+ break;
+ case DCA_SPEAKER_LAYOUT_5POINT0:
+ case DCA_SPEAKER_LAYOUT_5POINT1:
+ s->nactivechsets = (s->chset[0].nchannels < 5 && s->nchsets > 1) ? 2 : 1;
+ break;
+ default:
+ s->nactivechsets = s->nchsets;
+ break;
+ }
+
+ return 0;
+}
+
+static int parse_navi_table(DCAXllDecoder *s)
+{
+ int chs, seg, band, navi_nb, navi_pos, *navi_ptr;
+ DCAXllChSet *c;
+
+ // Determine size of NAVI table
+ navi_nb = s->nfreqbands * s->nframesegs * s->nchsets;
+ if (navi_nb > 1024) {
+ av_log(s->avctx, AV_LOG_ERROR, "Too many NAVI entries (%d)\n", navi_nb);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Reallocate NAVI table
+ av_fast_malloc(&s->navi, &s->navi_size, navi_nb * sizeof(*s->navi));
+ if (!s->navi)
+ return AVERROR(ENOMEM);
+
+ // Parse NAVI
+ navi_pos = get_bits_count(&s->gb);
+ navi_ptr = s->navi;
+ for (band = 0; band < s->nfreqbands; band++) {
+ for (seg = 0; seg < s->nframesegs; seg++) {
+ for (chs = 0, c = s->chset; chs < s->nchsets; chs++, c++) {
+ int size = 0;
+ if (c->nfreqbands > band) {
+ size = get_bits_long(&s->gb, s->seg_size_nbits);
+ if (size < 0 || size >= s->frame_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI segment size (%d bytes)\n", size);
+ return AVERROR_INVALIDDATA;
+ }
+ size++;
+ }
+ *navi_ptr++ = size;
+ }
+ }
+ }
+
+ // Byte align
+ // CRC16
+ skip_bits(&s->gb, -get_bits_count(&s->gb) & 7);
+ skip_bits(&s->gb, 16);
+
+ // Check CRC
+ if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
+ && ff_dca_check_crc(&s->gb, navi_pos, get_bits_count(&s->gb))) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI checksum\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ return 0;
+}
+
+static int parse_band_data(DCAXllDecoder *s)
+{
+ int ret, chs, seg, band, navi_pos, *navi_ptr;
+ DCAXllChSet *c;
+
+ for (chs = 0, c = s->chset; chs < s->nactivechsets; chs++, c++) {
+ if ((ret = chs_alloc_msb_band_data(s, c)) < 0)
+ return ret;
+ if ((ret = chs_alloc_lsb_band_data(s, c)) < 0)
+ return ret;
+ }
+
+ navi_pos = get_bits_count(&s->gb);
+ navi_ptr = s->navi;
+ for (band = 0; band < s->nfreqbands; band++) {
+ for (seg = 0; seg < s->nframesegs; seg++) {
+ for (chs = 0, c = s->chset; chs < s->nchsets; chs++, c++) {
+ if (c->nfreqbands > band) {
+ navi_pos += *navi_ptr * 8;
+ if (navi_pos > s->gb.size_in_bits) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI position\n");
+ return AVERROR_INVALIDDATA;
+ }
+ if (chs < s->nactivechsets &&
+ (ret = chs_parse_band_data(s, c, band, seg, navi_pos)) < 0) {
+ if (s->avctx->err_recognition & AV_EF_EXPLODE)
+ return ret;
+ chs_clear_band_data(s, c, band, seg);
+ }
+ s->gb.index = navi_pos;
+ }
+ navi_ptr++;
+ }
+ }
+ }
+
+ return 0;
+}
+
+static int parse_frame(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset)
+{
+ int ret;
+
+ if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
+ return ret;
+ if ((ret = parse_common_header(s)) < 0)
+ return ret;
+ if ((ret = parse_sub_headers(s, asset)) < 0)
+ return ret;
+ if ((ret = parse_navi_table(s)) < 0)
+ return ret;
+ if ((ret = parse_band_data(s)) < 0)
+ return ret;
+ if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL frame\n");
+ return AVERROR_INVALIDDATA;
+ }
+ return ret;
+}
+
+static void clear_pbr(DCAXllDecoder *s)
+{
+ s->pbr_length = 0;
+ s->pbr_delay = 0;
+}
+
+static int copy_to_pbr(DCAXllDecoder *s, uint8_t *data, int size, int delay)
+{
+ if (size > DCA_XLL_PBR_BUFFER_MAX)
+ return AVERROR(ENOSPC);
+
+ if (!s->pbr_buffer && !(s->pbr_buffer = av_malloc(DCA_XLL_PBR_BUFFER_MAX + DCA_BUFFER_PADDING_SIZE)))
+ return AVERROR(ENOMEM);
+
+ memcpy(s->pbr_buffer, data, size);
+ s->pbr_length = size;
+ s->pbr_delay = delay;
+ return 0;
+}
+
+static int parse_frame_no_pbr(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset)
+{
+ int ret = parse_frame(s, data, size, asset);
+
+ // If XLL packet data didn't start with a sync word, we must have jumped
+ // right into the middle of PBR smoothing period
+ if (ret == AVERROR(EAGAIN) && asset->xll_sync_present && asset->xll_sync_offset < size) {
+ // Skip to the next sync word in this packet
+ data += asset->xll_sync_offset;
+ size -= asset->xll_sync_offset;
+
+ // If decoding delay is set, put the frame into PBR buffer and return
+ // failure code. Higher level decoder is expected to switch to lossy
+ // core decoding or mute its output until decoding delay expires.
+ if (asset->xll_delay_nframes > 0) {
+ if ((ret = copy_to_pbr(s, data, size, asset->xll_delay_nframes)) < 0)
+ return ret;
+ return AVERROR(EAGAIN);
+ }
+
+ // No decoding delay, just parse the frame in place
+ ret = parse_frame(s, data, size, asset);
+ }
+
+ if (ret < 0)
+ return ret;
+
+ if (s->frame_size > size)
+ return AVERROR(EINVAL);
+
+ // If the XLL decoder didn't consume full packet, start PBR smoothing period
+ if (s->frame_size < size)
+ if ((ret = copy_to_pbr(s, data + s->frame_size, size - s->frame_size, 0)) < 0)
+ return ret;
+
+ return 0;
+}
+
+static int parse_frame_pbr(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset)
+{
+ int ret;
+
+ if (size > DCA_XLL_PBR_BUFFER_MAX - s->pbr_length) {
+ ret = AVERROR(ENOSPC);
+ goto fail;
+ }
+
+ memcpy(s->pbr_buffer + s->pbr_length, data, size);
+ s->pbr_length += size;
+
+ // Respect decoding delay after synchronization error
+ if (s->pbr_delay > 0 && --s->pbr_delay)
+ return AVERROR(EAGAIN);
+
+ if ((ret = parse_frame(s, s->pbr_buffer, s->pbr_length, asset)) < 0)
+ goto fail;
+
+ if (s->frame_size > s->pbr_length) {
+ ret = AVERROR(EINVAL);
+ goto fail;
+ }
+
+ if (s->frame_size == s->pbr_length) {
+ // End of PBR smoothing period
+ clear_pbr(s);
+ } else {
+ s->pbr_length -= s->frame_size;
+ memmove(s->pbr_buffer, s->pbr_buffer + s->frame_size, s->pbr_length);
+ }
+
+ return 0;
+
+fail:
+ // For now, throw out all PBR state on failure.
+ // Perhaps we can be smarter and try to resync somehow.
+ clear_pbr(s);
+ return ret;
+}
+
+int ff_dca_xll_parse(DCAXllDecoder *s, uint8_t *data, DCAExssAsset *asset)
+{
+ int ret;
+
+ if (s->hd_stream_id != asset->hd_stream_id) {
+ clear_pbr(s);
+ s->hd_stream_id = asset->hd_stream_id;
+ }
+
+ if (s->pbr_length)
+ ret = parse_frame_pbr(s, data + asset->xll_offset, asset->xll_size, asset);
+ else
+ ret = parse_frame_no_pbr(s, data + asset->xll_offset, asset->xll_size, asset);
+
+ return ret;
+}
+
+static void undo_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band)
+{
+ int i, j, k, nchannels = 0, *coeff_ptr = o->dmix_coeff;
+ DCAXllChSet *c;
+
+ for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) {
+ if (!c->hier_chset)
+ continue;
+
+ av_assert1(band < c->nfreqbands);
+ for (j = 0; j < c->nchannels; j++) {
+ for (k = 0; k < o->nchannels; k++) {
+ int coeff = *coeff_ptr++;
+ if (coeff) {
+ s->dcadsp->dmix_sub(c->bands[band].msb_sample_buffer[j],
+ o->bands[band].msb_sample_buffer[k],
+ coeff, s->nframesamples);
+ if (band)
+ s->dcadsp->dmix_sub(c->deci_history[j],
+ o->deci_history[k],
+ coeff, DCA_XLL_DECI_HISTORY_MAX);
+ }
+ }
+ }
+
+ nchannels += c->nchannels;
+ if (nchannels >= o->hier_ofs)
+ break;
+ }
+}
+
+static void scale_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band)
+{
+ int i, j, nchannels = 0;
+ DCAXllChSet *c;
+
+ for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) {
+ if (!c->hier_chset)
+ continue;
+
+ av_assert1(band < c->nfreqbands);
+ for (j = 0; j < c->nchannels; j++) {
+ int scale = o->dmix_scale[nchannels++];
+ if (scale != (1 << 15)) {
+ s->dcadsp->dmix_scale(c->bands[band].msb_sample_buffer[j],
+ scale, s->nframesamples);
+ if (band)
+ s->dcadsp->dmix_scale(c->deci_history[j],
+ scale, DCA_XLL_DECI_HISTORY_MAX);
+ }
+ }
+
+ if (nchannels >= o->hier_ofs)
+ break;
+ }
+}
+
+// Clear all band data and replace non-residual encoded channels with lossy
+// counterparts
+static void av_cold force_lossy_output(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ DCAContext *dca = s->avctx->priv_data;
+ int band, ch;
+
+ for (band = 0; band < c->nfreqbands; band++)
+ chs_clear_band_data(s, c, band, -1);
+
+ for (ch = 0; ch < c->nchannels; ch++) {
+ if (!(c->residual_encode & (1 << ch)))
+ continue;
+ if (ff_dca_core_map_spkr(&dca->core, c->ch_remap[ch]) < 0)
+ continue;
+ c->residual_encode &= ~(1 << ch);
+ }
+}
+
+static int combine_residual_frame(DCAXllDecoder *s, DCAXllChSet *c)
+{
+ DCAContext *dca = s->avctx->priv_data;
+ int ch, nsamples = s->nframesamples;
+ DCAXllChSet *o;
+
+ // Verify that core is compatible
+ if (!(dca->packet & DCA_PACKET_CORE)) {
+ av_log(s->avctx, AV_LOG_ERROR, "Residual encoded channels are present without core\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (c->freq != dca->core.output_rate) {
+ av_log(s->avctx, AV_LOG_WARNING, "Sample rate mismatch between core (%d Hz) and XLL (%d Hz)\n", dca->core.output_rate, c->freq);
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (nsamples != dca->core.npcmsamples) {
+ av_log(s->avctx, AV_LOG_WARNING, "Number of samples per frame mismatch between core (%d) and XLL (%d)\n", dca->core.npcmsamples, nsamples);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // See if this channel set is downmixed and find the next channel set in
+ // hierarchy. If downmixed, undo core pre-scaling before combining with
+ // residual (residual is not scaled).
+ o = find_next_hier_dmix_chset(s, c);
+
+ // Reduce core bit width and combine with residual
+ for (ch = 0; ch < c->nchannels; ch++) {
+ int n, spkr, shift, round;
+ int32_t *src, *dst;
+
+ if (c->residual_encode & (1 << ch))
+ continue;
+
+ // Map this channel to core speaker
+ spkr = ff_dca_core_map_spkr(&dca->core, c->ch_remap[ch]);
+ if (spkr < 0) {
+ av_log(s->avctx, AV_LOG_WARNING, "Residual encoded channel (%d) references unavailable core channel\n", c->ch_remap[ch]);
+ return AVERROR_INVALIDDATA;
+ }
+
+ // Account for LSB width
+ shift = 24 - c->pcm_bit_res + chs_get_lsb_width(s, c, 0, ch);
+ if (shift > 24) {
+ av_log(s->avctx, AV_LOG_WARNING, "Invalid core shift (%d bits)\n", shift);
+ return AVERROR_INVALIDDATA;
+ }
+
+ round = shift > 0 ? 1 << (shift - 1) : 0;
+
+ src = dca->core.output_samples[spkr];
+ dst = c->bands[0].msb_sample_buffer[ch];
+ if (o) {
+ // Undo embedded core downmix pre-scaling
+ int scale_inv = o->dmix_scale_inv[c->hier_ofs + ch];
+ for (n = 0; n < nsamples; n++)
+ dst[n] += clip23((mul16(src[n], scale_inv) + round) >> shift);
+ } else {
+ // No downmix scaling
+ for (n = 0; n < nsamples; n++)
+ dst[n] += (src[n] + round) >> shift;
+ }
+ }
+
+ return 0;
+}
+
+int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame)
+{
+ AVCodecContext *avctx = s->avctx;
+ DCAContext *dca = avctx->priv_data;
+ DCAExssAsset *asset = &dca->exss.assets[0];
+ DCAXllChSet *p = &s->chset[0], *c;
+ enum AVMatrixEncoding matrix_encoding = AV_MATRIX_ENCODING_NONE;
+ int i, j, k, ret, shift, nsamples, request_mask;
+ int ch_remap[DCA_SPEAKER_COUNT];
+
+ // Force lossy downmixed output during recovery
+ if (dca->packet & DCA_PACKET_RECOVERY) {
+ for (i = 0, c = s->chset; i < s->nchsets; i++, c++) {
+ if (i < s->nactivechsets)
+ force_lossy_output(s, c);
+
+ if (!c->primary_chset)
+ c->dmix_embedded = 0;
+ }
+
+ s->scalable_lsbs = 0;
+ s->fixed_lsb_width = 0;
+ }
+
+ // Filter frequency bands for active channel sets
+ s->output_mask = 0;
+ for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) {
+ chs_filter_band_data(s, c, 0);
+
+ if (c->residual_encode != (1 << c->nchannels) - 1
+ && (ret = combine_residual_frame(s, c)) < 0)
+ return ret;
+
+ if (s->scalable_lsbs)
+ chs_assemble_msbs_lsbs(s, c, 0);
+
+ if (c->nfreqbands > 1) {
+ chs_filter_band_data(s, c, 1);
+ chs_assemble_msbs_lsbs(s, c, 1);
+ }
+
+ s->output_mask |= c->ch_mask;
+ }
+
+ // Undo hierarchial downmix and/or apply scaling
+ for (i = 1, c = &s->chset[1]; i < s->nchsets; i++, c++) {
+ if (!is_hier_dmix_chset(c))
+ continue;
+
+ if (i >= s->nactivechsets) {
+ for (j = 0; j < c->nfreqbands; j++)
+ if (c->bands[j].dmix_embedded)
+ scale_down_mix(s, c, j);
+ break;
+ }
+
+ for (j = 0; j < c->nfreqbands; j++)
+ if (c->bands[j].dmix_embedded)
+ undo_down_mix(s, c, j);
+ }
+
+ // Assemble frequency bands for active channel sets
+ if (s->nfreqbands > 1) {
+ for (i = 0; i < s->nactivechsets; i++)
+ if ((ret = chs_assemble_freq_bands(s, &s->chset[i])) < 0)
+ return ret;
+ }
+
+ // Normalize to regular 5.1 layout if downmixing
+ if (dca->request_channel_layout) {
+ if (s->output_mask & DCA_SPEAKER_MASK_Lss) {
+ s->output_samples[DCA_SPEAKER_Ls] = s->output_samples[DCA_SPEAKER_Lss];
+ s->output_mask = (s->output_mask & ~DCA_SPEAKER_MASK_Lss) | DCA_SPEAKER_MASK_Ls;
+ }
+ if (s->output_mask & DCA_SPEAKER_MASK_Rss) {
+ s->output_samples[DCA_SPEAKER_Rs] = s->output_samples[DCA_SPEAKER_Rss];
+ s->output_mask = (s->output_mask & ~DCA_SPEAKER_MASK_Rss) | DCA_SPEAKER_MASK_Rs;
+ }
+ }
+
+ // Handle downmixing to stereo request
+ if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
+ && DCA_HAS_STEREO(s->output_mask) && p->dmix_embedded
+ && (p->dmix_type == DCA_DMIX_TYPE_LoRo ||
+ p->dmix_type == DCA_DMIX_TYPE_LtRt))
+ request_mask = DCA_SPEAKER_LAYOUT_STEREO;
+ else
+ request_mask = s->output_mask;
+ if (!ff_dca_set_channel_layout(avctx, ch_remap, request_mask))
+ return AVERROR(EINVAL);
+
+ avctx->sample_rate = p->freq << (s->nfreqbands - 1);
+
+ switch (p->storage_bit_res) {
+ case 16:
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ break;
+ case 24:
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ break;
+ default:
+ return AVERROR(EINVAL);
+ }
+
+ avctx->bits_per_raw_sample = p->storage_bit_res;
+ avctx->profile = FF_PROFILE_DTS_HD_MA;
+ avctx->bit_rate = 0;
+
+ frame->nb_samples = nsamples = s->nframesamples << (s->nfreqbands - 1);
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+ return ret;
+
+ // Downmix primary channel set to stereo
+ if (request_mask != s->output_mask) {
+ ff_dca_downmix_to_stereo_fixed(s->dcadsp, s->output_samples,
+ p->dmix_coeff, nsamples,
+ s->output_mask);
+ }
+
+ shift = p->storage_bit_res - p->pcm_bit_res;
+ for (i = 0; i < avctx->channels; i++) {
+ int32_t *samples = s->output_samples[ch_remap[i]];
+ if (frame->format == AV_SAMPLE_FMT_S16P) {
+ int16_t *plane = (int16_t *)frame->extended_data[i];
+ for (k = 0; k < nsamples; k++)
+ plane[k] = av_clip_int16(samples[k] * (1 << shift));
+ } else {
+ int32_t *plane = (int32_t *)frame->extended_data[i];
+ for (k = 0; k < nsamples; k++)
+ plane[k] = clip23(samples[k] * (1 << shift)) * (1 << 8);
+ }
+ }
+
+ if (!asset->one_to_one_map_ch_to_spkr) {
+ if (asset->representation_type == DCA_REPR_TYPE_LtRt)
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
+ else if (asset->representation_type == DCA_REPR_TYPE_LhRh)
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE;
+ } else if (request_mask != s->output_mask && p->dmix_type == DCA_DMIX_TYPE_LtRt) {
+ matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
+ }
+ if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
+ return ret;
+
+ return 0;
+}
+
+av_cold void ff_dca_xll_flush(DCAXllDecoder *s)
+{
+ clear_pbr(s);
+}
+
+av_cold void ff_dca_xll_close(DCAXllDecoder *s)
+{
+ DCAXllChSet *c;
+ int i, j;
+
+ for (i = 0, c = s->chset; i < DCA_XLL_CHSETS_MAX; i++, c++) {
+ for (j = 0; j < DCA_XLL_SAMPLE_BUFFERS_MAX; j++) {
+ av_freep(&c->sample_buffer[j]);
+ c->sample_size[j] = 0;
+ }
+ }
+
+ av_freep(&s->navi);
+ s->navi_size = 0;
+
+ av_freep(&s->pbr_buffer);
+ clear_pbr(s);
+}
diff --git a/libavcodec/dca_xll.h b/libavcodec/dca_xll.h
new file mode 100644
index 0000000..bc0aa65
--- /dev/null
+++ b/libavcodec/dca_xll.h
@@ -0,0 +1,149 @@
+/*
+ * 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_XLL_H
+#define AVCODEC_DCA_XLL_H
+
+#include "libavutil/common.h"
+#include "libavutil/mem.h"
+
+#include "avcodec.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "dca.h"
+#include "dcadsp.h"
+#include "dca_exss.h"
+
+#define DCA_XLL_CHSETS_MAX 3
+#define DCA_XLL_CHANNELS_MAX 8
+#define DCA_XLL_BANDS_MAX 2
+#define DCA_XLL_ADAPT_PRED_ORDER_MAX 16
+#define DCA_XLL_DECI_HISTORY_MAX 8
+#define DCA_XLL_DMIX_SCALES_MAX ((DCA_XLL_CHSETS_MAX - 1) * DCA_XLL_CHANNELS_MAX)
+#define DCA_XLL_DMIX_COEFFS_MAX (DCA_XLL_DMIX_SCALES_MAX * DCA_XLL_CHANNELS_MAX)
+#define DCA_XLL_PBR_BUFFER_MAX (240 << 10)
+#define DCA_XLL_SAMPLE_BUFFERS_MAX 3
+
+typedef struct DCAXllBand {
+ int decor_enabled; ///< Pairwise channel decorrelation flag
+ int orig_order[DCA_XLL_CHANNELS_MAX]; ///< Original channel order
+ int decor_coeff[DCA_XLL_CHANNELS_MAX / 2]; ///< Pairwise channel coefficients
+
+ int adapt_pred_order[DCA_XLL_CHANNELS_MAX]; ///< Adaptive predictor order
+ int highest_pred_order; ///< Highest adaptive predictor order
+ int fixed_pred_order[DCA_XLL_CHANNELS_MAX]; ///< Fixed predictor order
+ int adapt_refl_coeff[DCA_XLL_CHANNELS_MAX][DCA_XLL_ADAPT_PRED_ORDER_MAX]; ///< Adaptive predictor reflection coefficients
+
+ int dmix_embedded; ///< Downmix performed by encoder in frequency band
+
+ int lsb_section_size; ///< Size of LSB section in any segment
+ int nscalablelsbs[DCA_XLL_CHANNELS_MAX]; ///< Number of bits to represent the samples in LSB part
+ int bit_width_adjust[DCA_XLL_CHANNELS_MAX]; ///< Number of bits discarded by authoring
+
+ int32_t *msb_sample_buffer[DCA_XLL_CHANNELS_MAX]; ///< MSB sample buffer pointers
+ int32_t *lsb_sample_buffer[DCA_XLL_CHANNELS_MAX]; ///< LSB sample buffer pointers or NULL
+} DCAXllBand;
+
+typedef struct DCAXllChSet {
+ // Channel set header
+ int nchannels; ///< Number of channels in the channel set (N)
+ int residual_encode; ///< Residual encoding mask (0 - residual, 1 - full channel)
+ int pcm_bit_res; ///< PCM bit resolution (variable)
+ int storage_bit_res; ///< Storage bit resolution (16 or 24)
+ int freq; ///< Original sampling frequency (max. 96000 Hz)
+
+ int primary_chset; ///< Primary channel set flag
+ int dmix_coeffs_present; ///< Downmix coefficients present in stream
+ int dmix_embedded; ///< Downmix already performed by encoder
+ int dmix_type; ///< Primary channel set downmix type
+ int hier_chset; ///< Whether the channel set is part of a hierarchy
+ int hier_ofs; ///< Number of preceding channels in a hierarchy (M)
+ int dmix_coeff[DCA_XLL_DMIX_COEFFS_MAX]; ///< Downmixing coefficients
+ int dmix_scale[DCA_XLL_DMIX_SCALES_MAX]; ///< Downmixing scales
+ int dmix_scale_inv[DCA_XLL_DMIX_SCALES_MAX]; ///< Inverse downmixing scales
+ int ch_mask; ///< Channel mask for set
+ int ch_remap[DCA_XLL_CHANNELS_MAX]; ///< Channel to speaker map
+
+ int nfreqbands; ///< Number of frequency bands (1 or 2)
+ int nabits; ///< Number of bits to read bit allocation coding parameter
+
+ DCAXllBand bands[DCA_XLL_BANDS_MAX]; ///< Frequency bands
+
+ // Frequency band coding parameters
+ int seg_common; ///< Segment type
+ int rice_code_flag[DCA_XLL_CHANNELS_MAX]; ///< Rice coding flag
+ int bitalloc_hybrid_linear[DCA_XLL_CHANNELS_MAX]; ///< Binary code length for isolated samples
+ int bitalloc_part_a[DCA_XLL_CHANNELS_MAX]; ///< Coding parameter for part A of segment
+ int bitalloc_part_b[DCA_XLL_CHANNELS_MAX]; ///< Coding parameter for part B of segment
+ int nsamples_part_a[DCA_XLL_CHANNELS_MAX]; ///< Number of samples in part A of segment
+
+ // Decimator history
+ DECLARE_ALIGNED(32, int32_t, deci_history)[DCA_XLL_CHANNELS_MAX][DCA_XLL_DECI_HISTORY_MAX]; ///< Decimator history for frequency band 1
+
+ // Sample buffers
+ unsigned int sample_size[DCA_XLL_SAMPLE_BUFFERS_MAX];
+ int32_t *sample_buffer[DCA_XLL_SAMPLE_BUFFERS_MAX];
+} DCAXllChSet;
+
+typedef struct DCAXllDecoder {
+ AVCodecContext *avctx;
+ GetBitContext gb;
+
+ int frame_size; ///< Number of bytes in a lossless frame
+ int nchsets; ///< Number of channels sets per frame
+ int nframesegs; ///< Number of segments per frame
+ int nsegsamples_log2; ///< log2(nsegsamples)
+ int nsegsamples; ///< Samples in segment per one frequency band
+ int nframesamples_log2; ///< log2(nframesamples)
+ int nframesamples; ///< Samples in frame per one frequency band
+ int seg_size_nbits; ///< Number of bits used to read segment size
+ int band_crc_present; ///< Presence of CRC16 within each frequency band
+ int scalable_lsbs; ///< MSB/LSB split flag
+ int ch_mask_nbits; ///< Number of bits used to read channel mask
+ int fixed_lsb_width; ///< Fixed LSB width
+
+ DCAXllChSet chset[DCA_XLL_CHSETS_MAX]; ///< Channel sets
+
+ int *navi; ///< NAVI table
+ unsigned int navi_size;
+
+ int nfreqbands; ///< Highest number of frequency bands
+ int nchannels; ///< Total number of channels in a hierarchy
+ int nreschsets; ///< Number of channel sets that have residual encoded channels
+ int nactivechsets; ///< Number of active channel sets to decode
+
+ int hd_stream_id; ///< Previous DTS-HD stream ID for detecting changes
+
+ uint8_t *pbr_buffer; ///< Peak bit rate (PBR) smoothing buffer
+ int pbr_length; ///< Length in bytes of data currently buffered
+ int pbr_delay; ///< Delay in frames before decoding buffered data
+
+ DCADSPContext *dcadsp;
+
+ int output_mask;
+ int32_t *output_samples[DCA_SPEAKER_COUNT];
+} DCAXllDecoder;
+
+int ff_dca_xll_parse(DCAXllDecoder *s, uint8_t *data, DCAExssAsset *asset);
+int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame);
+av_cold void ff_dca_xll_flush(DCAXllDecoder *s);
+av_cold void ff_dca_xll_close(DCAXllDecoder *s);
+
+#endif
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
new file mode 100644
index 0000000..f3c3972
--- /dev/null
+++ b/libavcodec/dcadec.c
@@ -0,0 +1,417 @@
+/*
+ * 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 "libavutil/opt.h"
+#include "libavutil/channel_layout.h"
+
+#include "dcadec.h"
+#include "dcamath.h"
+#include "dca_syncwords.h"
+#include "profiles.h"
+
+#define MIN_PACKET_SIZE 16
+#define MAX_PACKET_SIZE 0x104000
+
+int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
+{
+ 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;
+ }
+
+ avctx->channels = nchannels;
+ return nchannels;
+}
+
+static uint16_t crc16(const uint8_t *data, int size)
+{
+ static const uint16_t crctab[16] = {
+ 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7,
+ 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef,
+ };
+
+ uint16_t res = 0xffff;
+ int i;
+
+ 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 res;
+}
+
+int ff_dca_check_crc(GetBitContext *s, int p1, int p2)
+{
+ 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;
+}
+
+void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples,
+ int *coeff_l, int nsamples, int ch_mask)
+{
+ int pos, spkr, max_spkr = av_log2(ch_mask);
+ int *coeff_r = coeff_l + av_popcount(ch_mask);
+
+ av_assert0(DCA_HAS_STEREO(ch_mask));
+
+ // 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);
+
+ // Downmix remaining channels
+ for (spkr = 0; spkr <= max_spkr; spkr++) {
+ if (!(ch_mask & (1U << spkr)))
+ continue;
+
+ if (*coeff_l && spkr != DCA_SPEAKER_L)
+ dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr],
+ *coeff_l, nsamples);
+
+ if (*coeff_r && spkr != DCA_SPEAKER_R)
+ dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr],
+ *coeff_r, nsamples);
+
+ coeff_l++;
+ coeff_r++;
+ }
+}
+
+void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples,
+ int *coeff_l, int nsamples, int ch_mask)
+{
+ 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);
+
+ av_assert0(DCA_HAS_STEREO(ch_mask));
+
+ // 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);
+
+ // Downmix remaining channels
+ for (spkr = 0; spkr <= max_spkr; spkr++) {
+ if (!(ch_mask & (1U << spkr)))
+ continue;
+
+ if (*coeff_l && spkr != DCA_SPEAKER_L)
+ fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr],
+ *coeff_l * scale, nsamples);
+
+ if (*coeff_r && spkr != DCA_SPEAKER_R)
+ fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr],
+ *coeff_r * scale, nsamples);
+
+ coeff_l++;
+ coeff_r++;
+ }
+}
+
+static int convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size)
+{
+ 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;
+ }
+}
+
+static int dcadec_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ 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;
+
+ if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ av_fast_malloc(&s->buffer, &s->buffer_size,
+ FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_SIZE);
+ if (!s->buffer)
+ return AVERROR(ENOMEM);
+
+ 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);
+
+ if (ret < 0)
+ return ret;
+
+ input = s->buffer;
+ input_size = ret;
+
+ s->packet = 0;
+
+ // 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;
+ }
+
+ s->packet |= DCA_PACKET_CORE;
+
+ // 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;
+ }
+ }
+
+ if (!s->core_only) {
+ DCAExssAsset *asset = NULL;
+
+ // 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];
+ }
+ }
+
+ // 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;
+ }
+ }
+
+ // 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;
+ }
+
+ // Filter the frame
+ if (s->packet & DCA_PACKET_XLL) {
+ if (s->packet & DCA_PACKET_CORE) {
+ int x96_synth = -1;
+
+ // Enable X96 synthesis if needed
+ if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000)
+ x96_synth = 1;
+
+ if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) {
+ s->core_residual_valid = 0;
+ return ret;
+ }
+
+ // 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;
+ }
+ }
+
+ 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;
+ }
+ }
+ } 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 {
+ return AVERROR_INVALIDDATA;
+ }
+
+ *got_frame_ptr = 1;
+
+ return avpkt->size;
+}
+
+static av_cold void dcadec_flush(AVCodecContext *avctx)
+{
+ DCAContext *s = avctx->priv_data;
+
+ ff_dca_core_flush(&s->core);
+ ff_dca_xll_flush(&s->xll);
+
+ s->core_residual_valid = 0;
+}
+
+static av_cold int dcadec_close(AVCodecContext *avctx)
+{
+ DCAContext *s = avctx->priv_data;
+
+ ff_dca_core_close(&s->core);
+ ff_dca_xll_close(&s->xll);
+
+ av_freep(&s->buffer);
+ s->buffer_size = 0;
+
+ return 0;
+}
+
+static av_cold int dcadec_init(AVCodecContext *avctx)
+{
+ DCAContext *s = avctx->priv_data;
+
+ s->avctx = avctx;
+ s->core.avctx = avctx;
+ s->exss.avctx = avctx;
+ s->xll.avctx = avctx;
+
+ if (ff_dca_core_init(&s->core) < 0)
+ return AVERROR(ENOMEM);
+
+ ff_dcadsp_init(&s->dcadsp);
+ s->core.dcadsp = &s->dcadsp;
+ s->xll.dcadsp = &s->dcadsp;
+
+ 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;
+ }
+
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ avctx->bits_per_raw_sample = 24;
+
+ return 0;
+}
+
+#define OFFSET(x) offsetof(DCAContext, x)
+#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+
+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 dcadec_class = {
+ .class_name = "DCA decoder",
+ .item_name = av_default_item_name,
+ .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 = 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,
+};
diff --git a/libavcodec/dcadec.h b/libavcodec/dcadec.h
new file mode 100644
index 0000000..6726121
--- /dev/null
+++ b/libavcodec/dcadec.h
@@ -0,0 +1,80 @@
+/*
+ * 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_DCADEC_H
+#define AVCODEC_DCADEC_H
+
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "dca.h"
+#include "dcadsp.h"
+#include "dca_core.h"
+#include "dca_exss.h"
+#include "dca_xll.h"
+
+#define DCA_BUFFER_PADDING_SIZE 1024
+
+#define DCA_PACKET_CORE 0x01
+#define DCA_PACKET_EXSS 0x02
+#define DCA_PACKET_XLL 0x04
+#define DCA_PACKET_RECOVERY 0x08
+
+typedef struct DCAContext {
+ const AVClass *class; ///< class for AVOptions
+ AVCodecContext *avctx;
+
+ DCACoreDecoder core; ///< Core decoder context
+ DCAExssParser exss; ///< EXSS parser context
+ DCAXllDecoder xll; ///< XLL decoder context
+
+ DCADSPContext dcadsp;
+
+ uint8_t *buffer; ///< Packet buffer
+ unsigned int buffer_size;
+
+ int packet; ///< Packet flags
+
+ int core_residual_valid; ///< Core valid for residual decoding
+
+ int request_channel_layout; ///< Converted from avctx.request_channel_layout
+ int core_only; ///< Core only decoding flag
+} DCAContext;
+
+int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask);
+
+int ff_dca_check_crc(GetBitContext *s, int p1, int p2);
+
+void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples,
+ int *coeff_l, int nsamples, int ch_mask);
+void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples,
+ int *coeff_l, int nsamples, int ch_mask);
+
+static inline int ff_dca_seek_bits(GetBitContext *s, int p)
+{
+ if (p < s->index || p > s->size_in_bits)
+ return -1;
+ s->index = p;
+ return 0;
+}
+
+#endif
diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
new file mode 100644
index 0000000..cee3d60
--- /dev/null
+++ b/libavcodec/dcadsp.c
@@ -0,0 +1,413 @@
+/*
+ * 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 "libavutil/mem.h"
+
+#include "dcadsp.h"
+#include "dcamath.h"
+
+static void decode_hf_c(int32_t **dst,
+ const int32_t *vq_index,
+ const int8_t hf_vq[1024][32],
+ int32_t scale_factors[32][2],
+ intptr_t sb_start, intptr_t sb_end,
+ intptr_t ofs, intptr_t len)
+{
+ int i, j;
+
+ for (i = sb_start; i < sb_end; i++) {
+ const int8_t *coeff = hf_vq[vq_index[i]];
+ int32_t scale = scale_factors[i][0];
+ for (j = 0; j < len; j++)
+ dst[i][j + ofs] = clip23(coeff[j] * scale + (1 << 3) >> 4);
+ }
+}
+
+static void decode_joint_c(int32_t **dst, int32_t **src,
+ const int32_t *scale_factors,
+ intptr_t sb_start, intptr_t sb_end,
+ intptr_t ofs, intptr_t len)
+{
+ int i, j;
+
+ for (i = sb_start; i < sb_end; i++) {
+ int32_t scale = scale_factors[i];
+ for (j = 0; j < len; j++)
+ dst[i][j + ofs] = clip23(mul17(src[i][j + ofs], scale));
+ }
+}
+
+static void lfe_fir_float_c(float *pcm_samples, int32_t *lfe_samples,
+ const float *filter_coeff, intptr_t npcmblocks,
+ int dec_select)
+{
+ // Select decimation factor
+ int factor = 64 << dec_select;
+ int ncoeffs = 8 >> dec_select;
+ int nlfesamples = npcmblocks >> (dec_select + 1);
+ int i, j, k;
+
+ for (i = 0; i < nlfesamples; i++) {
+ // One decimated sample generates 64 or 128 interpolated ones
+ for (j = 0; j < factor / 2; j++) {
+ float a = 0;
+ float b = 0;
+
+ for (k = 0; k < ncoeffs; k++) {
+ a += filter_coeff[ j * ncoeffs + k] * lfe_samples[-k];
+ b += filter_coeff[255 - j * ncoeffs - k] * lfe_samples[-k];
+ }
+
+ pcm_samples[ j] = a;
+ pcm_samples[factor / 2 + j] = b;
+ }
+
+ lfe_samples++;
+ pcm_samples += factor;
+ }
+}
+
+static void lfe_fir1_float_c(float *pcm_samples, int32_t *lfe_samples,
+ const float *filter_coeff, intptr_t npcmblocks)
+{
+ lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 0);
+}
+
+static void lfe_fir2_float_c(float *pcm_samples, int32_t *lfe_samples,
+ const float *filter_coeff, intptr_t npcmblocks)
+{
+ lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 1);
+}
+
+static void lfe_x96_float_c(float *dst, const float *src,
+ float *hist, intptr_t len)
+{
+ float prev = *hist;
+ int i;
+
+ for (i = 0; i < len; i++) {
+ float a = 0.25f * src[i] + 0.75f * prev;
+ float b = 0.75f * src[i] + 0.25f * prev;
+ prev = src[i];
+ *dst++ = a;
+ *dst++ = b;
+ }
+
+ *hist = prev;
+}
+
+static void sub_qmf32_float_c(SynthFilterContext *synth,
+ FFTContext *imdct,
+ float *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ float *hist1, int *offset, float *hist2,
+ const float *filter_coeff, intptr_t npcmblocks,
+ float scale)
+{
+ LOCAL_ALIGNED(32, float, input, [32]);
+ int i, j;
+
+ for (j = 0; j < npcmblocks; j++) {
+ // Load in one sample from each subband
+ for (i = 0; i < 32; i++) {
+ if ((i - 1) & 2)
+ input[i] = -subband_samples_lo[i][j];
+ else
+ input[i] = subband_samples_lo[i][j];
+ }
+
+ // One subband sample generates 32 interpolated ones
+ synth->synth_filter_float(imdct, hist1, offset,
+ hist2, filter_coeff,
+ pcm_samples, input, scale);
+ pcm_samples += 32;
+ }
+}
+
+static void sub_qmf64_float_c(SynthFilterContext *synth,
+ FFTContext *imdct,
+ float *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ float *hist1, int *offset, float *hist2,
+ const float *filter_coeff, intptr_t npcmblocks,
+ float scale)
+{
+ LOCAL_ALIGNED(32, float, input, [64]);
+ int i, j;
+
+ if (!subband_samples_hi)
+ memset(&input[32], 0, sizeof(input[0]) * 32);
+
+ for (j = 0; j < npcmblocks; j++) {
+ // Load in one sample from each subband
+ if (subband_samples_hi) {
+ // Full 64 subbands, first 32 are residual coded
+ for (i = 0; i < 32; i++) {
+ if ((i - 1) & 2)
+ input[i] = -subband_samples_lo[i][j] - subband_samples_hi[i][j];
+ else
+ input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j];
+ }
+ for (i = 32; i < 64; i++) {
+ if ((i - 1) & 2)
+ input[i] = -subband_samples_hi[i][j];
+ else
+ input[i] = subband_samples_hi[i][j];
+ }
+ } else {
+ // Only first 32 subbands
+ for (i = 0; i < 32; i++) {
+ if ((i - 1) & 2)
+ input[i] = -subband_samples_lo[i][j];
+ else
+ input[i] = subband_samples_lo[i][j];
+ }
+ }
+
+ // One subband sample generates 64 interpolated ones
+ synth->synth_filter_float_64(imdct, hist1, offset,
+ hist2, filter_coeff,
+ pcm_samples, input, scale);
+ pcm_samples += 64;
+ }
+}
+
+static void lfe_fir_fixed_c(int32_t *pcm_samples, int32_t *lfe_samples,
+ const int32_t *filter_coeff, intptr_t npcmblocks)
+{
+ // Select decimation factor
+ int nlfesamples = npcmblocks >> 1;
+ int i, j, k;
+
+ for (i = 0; i < nlfesamples; i++) {
+ // One decimated sample generates 64 interpolated ones
+ for (j = 0; j < 32; j++) {
+ int64_t a = 0;
+ int64_t b = 0;
+
+ for (k = 0; k < 8; k++) {
+ a += (int64_t)filter_coeff[ j * 8 + k] * lfe_samples[-k];
+ b += (int64_t)filter_coeff[255 - j * 8 - k] * lfe_samples[-k];
+ }
+
+ pcm_samples[ j] = clip23(norm23(a));
+ pcm_samples[32 + j] = clip23(norm23(b));
+ }
+
+ lfe_samples++;
+ pcm_samples += 64;
+ }
+}
+
+static void lfe_x96_fixed_c(int32_t *dst, const int32_t *src,
+ int32_t *hist, intptr_t len)
+{
+ int32_t prev = *hist;
+ int i;
+
+ for (i = 0; i < len; i++) {
+ int64_t a = INT64_C(2097471) * src[i] + INT64_C(6291137) * prev;
+ int64_t b = INT64_C(6291137) * src[i] + INT64_C(2097471) * prev;
+ prev = src[i];
+ *dst++ = clip23(norm23(a));
+ *dst++ = clip23(norm23(b));
+ }
+
+ *hist = prev;
+}
+
+static void sub_qmf32_fixed_c(SynthFilterContext *synth,
+ DCADCTContext *imdct,
+ int32_t *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ int32_t *hist1, int *offset, int32_t *hist2,
+ const int32_t *filter_coeff, intptr_t npcmblocks)
+{
+ LOCAL_ALIGNED(32, int32_t, input, [32]);
+ int i, j;
+
+ for (j = 0; j < npcmblocks; j++) {
+ // Load in one sample from each subband
+ for (i = 0; i < 32; i++)
+ input[i] = subband_samples_lo[i][j];
+
+ // One subband sample generates 32 interpolated ones
+ synth->synth_filter_fixed(imdct, hist1, offset,
+ hist2, filter_coeff,
+ pcm_samples, input);
+ pcm_samples += 32;
+ }
+}
+
+static void sub_qmf64_fixed_c(SynthFilterContext *synth,
+ DCADCTContext *imdct,
+ int32_t *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ int32_t *hist1, int *offset, int32_t *hist2,
+ const int32_t *filter_coeff, intptr_t npcmblocks)
+{
+ LOCAL_ALIGNED(32, int32_t, input, [64]);
+ int i, j;
+
+ if (!subband_samples_hi)
+ memset(&input[32], 0, sizeof(input[0]) * 32);
+
+ for (j = 0; j < npcmblocks; j++) {
+ // Load in one sample from each subband
+ if (subband_samples_hi) {
+ // Full 64 subbands, first 32 are residual coded
+ for (i = 0; i < 32; i++)
+ input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j];
+ for (i = 32; i < 64; i++)
+ input[i] = subband_samples_hi[i][j];
+ } else {
+ // Only first 32 subbands
+ for (i = 0; i < 32; i++)
+ input[i] = subband_samples_lo[i][j];
+ }
+
+ // One subband sample generates 64 interpolated ones
+ synth->synth_filter_fixed_64(imdct, hist1, offset,
+ hist2, filter_coeff,
+ pcm_samples, input);
+ pcm_samples += 64;
+ }
+}
+
+static void decor_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] += src[i] * coeff + (1 << 2) >> 3;
+}
+
+static void dmix_sub_xch_c(int32_t *dst1, int32_t *dst2,
+ const int32_t *src, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++) {
+ int32_t cs = mul23(src[i], 5931520 /* M_SQRT1_2 * (1 << 23) */);
+ dst1[i] -= cs;
+ dst2[i] -= cs;
+ }
+}
+
+static void dmix_sub_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] -= mul15(src[i], coeff);
+}
+
+static void dmix_add_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] += mul15(src[i], coeff);
+}
+
+static void dmix_scale_c(int32_t *dst, intptr_t scale, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] = mul15(dst[i], scale);
+}
+
+static void dmix_scale_inv_c(int32_t *dst, intptr_t scale_inv, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] = mul16(dst[i], scale_inv);
+}
+
+static void filter0(int32_t *dst, const int32_t *src, int32_t coeff, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] -= mul22(src[i], coeff);
+}
+
+static void filter1(int32_t *dst, const int32_t *src, int32_t coeff, intptr_t len)
+{
+ int i;
+
+ for (i = 0; i < len; i++)
+ dst[i] -= mul23(src[i], coeff);
+}
+
+static void assemble_freq_bands_c(int32_t *dst, int32_t *src0, int32_t *src1,
+ const int32_t *coeff, intptr_t len)
+{
+ int i;
+
+ filter0(src0, src1, coeff[0], len);
+ filter0(src1, src0, coeff[1], len);
+ filter0(src0, src1, coeff[2], len);
+ filter0(src1, src0, coeff[3], len);
+
+ for (i = 0; i < 8; i++, src0--) {
+ filter1(src0, src1, coeff[i + 4], len);
+ filter1(src1, src0, coeff[i + 12], len);
+ filter1(src0, src1, coeff[i + 4], len);
+ }
+
+ for (i = 0; i < len; i++) {
+ *dst++ = *src1++;
+ *dst++ = *++src0;
+ }
+}
+
+av_cold void ff_dcadsp_init(DCADSPContext *s)
+{
+ s->decode_hf = decode_hf_c;
+ s->decode_joint = decode_joint_c;
+
+ s->lfe_fir_float[0] = lfe_fir1_float_c;
+ s->lfe_fir_float[1] = lfe_fir2_float_c;
+ s->lfe_x96_float = lfe_x96_float_c;
+ s->sub_qmf_float[0] = sub_qmf32_float_c;
+ s->sub_qmf_float[1] = sub_qmf64_float_c;
+
+ s->lfe_fir_fixed = lfe_fir_fixed_c;
+ s->lfe_x96_fixed = lfe_x96_fixed_c;
+ s->sub_qmf_fixed[0] = sub_qmf32_fixed_c;
+ s->sub_qmf_fixed[1] = sub_qmf64_fixed_c;
+
+ s->decor = decor_c;
+
+ s->dmix_sub_xch = dmix_sub_xch_c;
+ s->dmix_sub = dmix_sub_c;
+ s->dmix_add = dmix_add_c;
+ s->dmix_scale = dmix_scale_c;
+ s->dmix_scale_inv = dmix_scale_inv_c;
+
+ s->assemble_freq_bands = assemble_freq_bands_c;
+}
diff --git a/libavcodec/dcadsp.h b/libavcodec/dcadsp.h
new file mode 100644
index 0000000..d8acf37
--- /dev/null
+++ b/libavcodec/dcadsp.h
@@ -0,0 +1,91 @@
+/*
+ * 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_DCADSP_H
+#define AVCODEC_DCADSP_H
+
+#include "libavutil/common.h"
+
+#include "fft.h"
+#include "dcadct.h"
+#include "synth_filter.h"
+
+typedef struct DCADSPContext {
+ void (*decode_hf)(int32_t **dst,
+ const int32_t *vq_index,
+ const int8_t hf_vq[1024][32],
+ int32_t scale_factors[32][2],
+ intptr_t sb_start, intptr_t sb_end,
+ intptr_t ofs, intptr_t len);
+
+ void (*decode_joint)(int32_t **dst, int32_t **src,
+ const int32_t *scale_factors,
+ intptr_t sb_start, intptr_t sb_end,
+ intptr_t ofs, intptr_t len);
+
+ void (*lfe_fir_float[2])(float *pcm_samples, int32_t *lfe_samples,
+ const float *filter_coeff, intptr_t npcmblocks);
+
+ void (*lfe_x96_float)(float *dst, const float *src,
+ float *hist, intptr_t len);
+
+ void (*sub_qmf_float[2])(SynthFilterContext *synth,
+ FFTContext *imdct,
+ float *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ float *hist1, int *offset, float *hist2,
+ const float *filter_coeff, intptr_t npcmblocks,
+ float scale);
+
+ void (*lfe_fir_fixed)(int32_t *pcm_samples, int32_t *lfe_samples,
+ const int32_t *filter_coeff, intptr_t npcmblocks);
+
+ void (*lfe_x96_fixed)(int32_t *dst, const int32_t *src,
+ int32_t *hist, intptr_t len);
+
+ void (*sub_qmf_fixed[2])(SynthFilterContext *synth,
+ DCADCTContext *imdct,
+ int32_t *pcm_samples,
+ int32_t **subband_samples_lo,
+ int32_t **subband_samples_hi,
+ int32_t *hist1, int *offset, int32_t *hist2,
+ const int32_t *filter_coeff, intptr_t npcmblocks);
+
+ void (*decor)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len);
+
+ void (*dmix_sub_xch)(int32_t *dst1, int32_t *dst2,
+ const int32_t *src, intptr_t len);
+
+ void (*dmix_sub)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len);
+
+ void (*dmix_add)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len);
+
+ void (*dmix_scale)(int32_t *dst, intptr_t scale, intptr_t len);
+
+ void (*dmix_scale_inv)(int32_t *dst, intptr_t scale_inv, intptr_t len);
+
+ void (*assemble_freq_bands)(int32_t *dst, int32_t *src0, int32_t *src1,
+ const int32_t *coeff, intptr_t len);
+} DCADSPContext;
+
+av_cold void ff_dcadsp_init(DCADSPContext *s);
+
+#endif
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 5740137..02063c8 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -30,7 +30,7 @@
#define LIBAVCODEC_VERSION_MAJOR 57
#define LIBAVCODEC_VERSION_MINOR 24
-#define LIBAVCODEC_VERSION_MICRO 100
+#define LIBAVCODEC_VERSION_MICRO 101
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile
index eec98cb..ce06b90 100644
--- a/libavcodec/x86/Makefile
+++ b/libavcodec/x86/Makefile
@@ -44,7 +44,7 @@ OBJS-$(CONFIG_ADPCM_G722_ENCODER) += x86/g722dsp_init.o
OBJS-$(CONFIG_ALAC_DECODER) += x86/alacdsp_init.o
OBJS-$(CONFIG_APNG_DECODER) += x86/pngdsp_init.o
OBJS-$(CONFIG_CAVS_DECODER) += x86/cavsdsp.o
-#OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter_init.o
+OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter_init.o
OBJS-$(CONFIG_DNXHD_ENCODER) += x86/dnxhdenc_init.o
OBJS-$(CONFIG_HEVC_DECODER) += x86/hevcdsp_init.o
OBJS-$(CONFIG_JPEG2000_DECODER) += x86/jpeg2000dsp_init.o
@@ -132,7 +132,7 @@ YASM-OBJS-$(CONFIG_ADPCM_G722_DECODER) += x86/g722dsp.o
YASM-OBJS-$(CONFIG_ADPCM_G722_ENCODER) += x86/g722dsp.o
YASM-OBJS-$(CONFIG_ALAC_DECODER) += x86/alacdsp.o
YASM-OBJS-$(CONFIG_APNG_DECODER) += x86/pngdsp.o
-#YASM-OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter.o
+YASM-OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter.o
YASM-OBJS-$(CONFIG_DIRAC_DECODER) += x86/diracdsp_mmx.o x86/diracdsp_yasm.o \
x86/dwt_yasm.o
YASM-OBJS-$(CONFIG_DNXHD_ENCODER) += x86/dnxhdenc.o
diff --git a/tests/checkasm/Makefile b/tests/checkasm/Makefile
index 14a11d6..07fe5bc 100644
--- a/tests/checkasm/Makefile
+++ b/tests/checkasm/Makefile
@@ -1,7 +1,7 @@
# libavcodec tests
AVCODECOBJS-$(CONFIG_ALAC_DECODER) += alacdsp.o
AVCODECOBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
-#AVCODECOBJS-$(CONFIG_DCA_DECODER) += synth_filter.o
+AVCODECOBJS-$(CONFIG_DCA_DECODER) += synth_filter.o
AVCODECOBJS-$(CONFIG_FLACDSP) += flacdsp.o
AVCODECOBJS-$(CONFIG_FMTCONVERT) += fmtconvert.o
AVCODECOBJS-$(CONFIG_H264PRED) += h264pred.o
diff --git a/tests/checkasm/checkasm.c b/tests/checkasm/checkasm.c
index f7d1331..49fd2af 100644
--- a/tests/checkasm/checkasm.c
+++ b/tests/checkasm/checkasm.c
@@ -71,9 +71,9 @@ static const struct {
#if CONFIG_BSWAPDSP
{ "bswapdsp", checkasm_check_bswapdsp },
#endif
-/* #if CONFIG_DCA_DECODER
+ #if CONFIG_DCA_DECODER
{ "synth_filter", checkasm_check_synth_filter },
- #endif*/
+ #endif
#if CONFIG_FLACDSP
{ "flacdsp", checkasm_check_flacdsp },
#endif
diff --git a/tests/fate/acodec.mak b/tests/fate/acodec.mak
index 62b1bc1..e0f2320 100644
--- a/tests/fate/acodec.mak
+++ b/tests/fate/acodec.mak
@@ -99,14 +99,14 @@ FATE_ACODEC-$(call ENCDEC, ALAC, MOV) += fate-acodec-alac
fate-acodec-alac: FMT = mov
fate-acodec-alac: CODEC = alac -compression_level 1
-#FATE_ACODEC-$(call ENCDEC, DCA, DTS) += fate-acodec-dca
+FATE_ACODEC-$(call ENCDEC, DCA, DTS) += fate-acodec-dca
fate-acodec-dca: tests/data/asynth-44100-2.wav
fate-acodec-dca: SRC = tests/data/asynth-44100-2.wav
fate-acodec-dca: CMD = md5 -i $(TARGET_PATH)/$(SRC) -c:a dca -strict -2 -f dts -flags +bitexact
fate-acodec-dca: CMP = oneline
fate-acodec-dca: REF = 7ffdefdf47069289990755c79387cc90
-#FATE_ACODEC-$(call ENCDEC, DCA, WAV) += fate-acodec-dca2
+FATE_ACODEC-$(call ENCDEC, DCA, WAV) += fate-acodec-dca2
fate-acodec-dca2: CMD = enc_dec_pcm dts wav s16le $(SRC) -c:a dca -strict -2 -flags +bitexact
fate-acodec-dca2: REF = $(SRC)
fate-acodec-dca2: CMP = stddev
diff --git a/tests/fate/audio.mak b/tests/fate/audio.mak
index 686b7df..93c19a0 100644
--- a/tests/fate/audio.mak
+++ b/tests/fate/audio.mak
@@ -21,7 +21,7 @@ fate-dca-core: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts.ts
fate-dca-core: CMP = oneoff
fate-dca-core: REF = $(SAMPLES)/dts/dts.pcm
-#FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
+FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
fate-dca: $(FATE_DCA-yes)
FATE_SAMPLES_AUDIO-$(call DEMDEC, DSICIN, DSICINAUDIO) += fate-delphine-cin-audio
@@ -31,7 +31,7 @@ FATE_SAMPLES_AUDIO-$(call DEMDEC, DSS, DSS_SP) += fate-dss-lp fate-dss-sp
fate-dss-lp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/lp.dss -frames 30
fate-dss-sp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/sp.dss -frames 30
-#FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
+FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
fate-dts_es: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts_es.dts
fate-dts_es: CMP = oneoff
fate-dts_es: REF = $(SAMPLES)/dts/dts_es_2.pcm
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