X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdcadec.c;h=187e1728be6e7c98bf44fd919066fa0ddb2a3e1a;hb=95a2b883e36d3499c6bf620c65ac9d21fa2bd808;hp=aca6ed325ffa3f3b83e3088169cab25719a8edc4;hpb=2c6811397bdf13d43ca206e48d6d6da9c2cd47c6;p=ffmpeg diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c index aca6ed325ff..187e1728be6 100644 --- a/libavcodec/dcadec.c +++ b/libavcodec/dcadec.c @@ -7,20 +7,20 @@ * Copyright (C) 2012 Paul B Mahol * Copyright (C) 2014 Niels Möller * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -50,7 +50,6 @@ #include "internal.h" #include "mathops.h" #include "profiles.h" -#include "put_bits.h" #include "synth_filter.h" #if ARCH_ARM @@ -71,62 +70,36 @@ enum DCAMode { DCA_4F2R }; -/* -1 are reserved or unknown */ -static const int dca_ext_audio_descr_mask[] = { - DCA_EXT_XCH, - -1, - DCA_EXT_X96, - DCA_EXT_XCH | DCA_EXT_X96, - -1, - -1, - DCA_EXT_XXCH, - -1, -}; -/* Tables for mapping dts channel configurations to libavcodec multichannel api. - * Some compromises have been made for special configurations. Most configurations - * are never used so complete accuracy is not needed. - * - * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead. - * S -> side, when both rear and back are configured move one of them to the side channel - * OV -> center back - * All 2 channel configurations -> AV_CH_LAYOUT_STEREO - */ -static const uint64_t dca_core_channel_layout[] = { - AV_CH_FRONT_CENTER, ///< 1, A - AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono) - AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo) - AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference) - AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total) - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R - AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT | - AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR - - AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | - AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV - - AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER | - AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | - AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | - AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | - AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | - AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2 - - AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | - AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | - AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR +enum DCAXxchSpeakerMask { + DCA_XXCH_FRONT_CENTER = 0x0000001, + DCA_XXCH_FRONT_LEFT = 0x0000002, + DCA_XXCH_FRONT_RIGHT = 0x0000004, + DCA_XXCH_SIDE_REAR_LEFT = 0x0000008, + DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010, + DCA_XXCH_LFE1 = 0x0000020, + DCA_XXCH_REAR_CENTER = 0x0000040, + DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080, + DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100, + DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200, + DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400, + DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800, + DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000, + DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000, + DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000, + DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000, + DCA_XXCH_LFE2 = 0x0010000, + DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000, + DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000, + DCA_XXCH_OVERHEAD = 0x0080000, + DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000, + DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000, + DCA_XXCH_REAR_HIGH_CENTER = 0x0400000, + DCA_XXCH_REAR_HIGH_LEFT = 0x0800000, + DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000, + DCA_XXCH_REAR_LOW_CENTER = 0x2000000, + DCA_XXCH_REAR_LOW_LEFT = 0x4000000, + DCA_XXCH_REAR_LOW_RIGHT = 0x8000000, }; #define DCA_DOLBY 101 /* FIXME */ @@ -162,6 +135,8 @@ static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, ba->offset; } +static float dca_dmix_code(unsigned code); + static av_cold void dca_init_vlcs(void) { static int vlcs_initialized = 0; @@ -223,16 +198,103 @@ static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) *dst++ = get_bits(gb, bits); } -static int dca_parse_audio_coding_header(DCAContext *s, int base_channel) +static inline int dca_xxch2index(DCAContext *s, int xxch_ch) +{ + int i, base, mask; + + /* locate channel set containing the channel */ + for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1); + i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i]) + base += av_popcount(mask); + + return base + av_popcount(mask & (xxch_ch - 1)); +} + +static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, + int xxch) { int i, j; - static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; + static const uint8_t adj_table[4] = { 16, 18, 20, 23 }; static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; + int hdr_pos = 0, hdr_size = 0; + float scale_factor; + int this_chans, acc_mask; + int embedded_downmix; + int nchans, mask[8]; + int coeff, ichan; + + /* xxch has arbitrary sized audio coding headers */ + if (xxch) { + hdr_pos = get_bits_count(&s->gb); + hdr_size = get_bits(&s->gb, 7) + 1; + } + + nchans = get_bits(&s->gb, 3) + 1; + if (xxch && nchans >= 3) { + av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans); + return AVERROR_INVALIDDATA; + } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) { + av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel); + return AVERROR_INVALIDDATA; + } - s->audio_header.total_channels = get_bits(&s->gb, 3) + 1 + base_channel; + s->audio_header.total_channels = nchans + base_channel; s->audio_header.prim_channels = s->audio_header.total_channels; + /* obtain speaker layout mask & downmix coefficients for XXCH */ + if (xxch) { + acc_mask = s->xxch_core_spkmask; + + this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6; + s->xxch_spk_masks[s->xxch_chset] = this_chans; + s->xxch_chset_nch[s->xxch_chset] = nchans; + + for (i = 0; i <= s->xxch_chset; i++) + acc_mask |= s->xxch_spk_masks[i]; + + /* check for downmixing information */ + if (get_bits1(&s->gb)) { + embedded_downmix = get_bits1(&s->gb); + coeff = get_bits(&s->gb, 6); + + if (coeff<1 || coeff>61) { + av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff); + return AVERROR_INVALIDDATA; + } + + scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3); + + s->xxch_dmix_sf[s->xxch_chset] = scale_factor; + + for (i = base_channel; i < s->audio_header.prim_channels; i++) { + mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); + } + + for (j = base_channel; j < s->audio_header.prim_channels; j++) { + memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); + s->xxch_dmix_embedded |= (embedded_downmix << j); + for (i = 0; i < s->xxch_nbits_spk_mask; i++) { + if (mask[j] & (1 << i)) { + if ((1 << i) == DCA_XXCH_LFE1) { + av_log(s->avctx, AV_LOG_WARNING, + "DCA-XXCH: dmix to LFE1 not supported.\n"); + continue; + } + + coeff = get_bits(&s->gb, 7); + ichan = dca_xxch2index(s, 1 << i); + if ((coeff&63)<1 || (coeff&63)>61) { + av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff); + return AVERROR_INVALIDDATA; + } + s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3); + } + } + } + } + } + if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX) s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX; @@ -265,16 +327,23 @@ static int dca_parse_audio_coding_header(DCAContext *s, int base_channel) /* Get scale factor adjustment */ for (j = 0; j < 11; j++) for (i = base_channel; i < s->audio_header.prim_channels; i++) - s->audio_header.scalefactor_adj[i][j] = 1; + s->audio_header.scalefactor_adj[i][j] = 16; for (j = 1; j < 11; j++) for (i = base_channel; i < s->audio_header.prim_channels; i++) if (s->audio_header.quant_index_huffman[i][j] < thr[j]) s->audio_header.scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; - if (s->crc_present) { - /* Audio header CRC check */ - get_bits(&s->gb, 16); + if (!xxch) { + if (s->crc_present) { + /* Audio header CRC check */ + get_bits(&s->gb, 16); + } + } else { + /* Skip to the end of the header, also ignore CRC if present */ + i = get_bits_count(&s->gb); + if (hdr_pos + 8 * hdr_size > i) + skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i); } s->current_subframe = 0; @@ -319,6 +388,7 @@ static int dca_parse_frame_header(DCAContext *s) s->predictor_history = get_bits(&s->gb, 1); if (s->lfe > 2) { + s->lfe = 0; av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe); return AVERROR_INVALIDDATA; } @@ -343,7 +413,7 @@ static int dca_parse_frame_header(DCAContext *s) /* Primary audio coding header */ s->audio_header.subframes = get_bits(&s->gb, 4) + 1; - return dca_parse_audio_coding_header(s, 0); + return dca_parse_audio_coding_header(s, 0, 0); } static inline int get_scale(GetBitContext *gb, int level, int value, int log2range) @@ -373,6 +443,10 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) if (!base_channel) { s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1; + if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) { + s->subsubframes[s->current_subframe] = 1; + return AVERROR_INVALIDDATA; + } s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); } @@ -519,6 +593,7 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) /* Low frequency effect data */ if (!base_channel && s->lfe) { + int quant7; /* LFE samples */ int lfe_samples = 2 * s->lfe * (4 + block_index); int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]); @@ -530,8 +605,12 @@ static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) } /* Scale factor index */ - skip_bits(&s->gb, 1); - s->lfe_scale_factor = ff_dca_scale_factor_quant7[get_bits(&s->gb, 7)]; + quant7 = get_bits(&s->gb, 8); + if (quant7 > 127) { + avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127"); + return AVERROR_INVALIDDATA; + } + s->lfe_scale_factor = ff_dca_scale_factor_quant7[quant7]; /* Quantization step size * scale factor */ lfe_scale = 0.035 * s->lfe_scale_factor; @@ -709,7 +788,7 @@ static void dca_downmix(float **samples, int srcfmt, int lfe_present, switch (srcfmt) { case DCA_MONO: case DCA_4F2R: - av_log(NULL, 0, "Not implemented!\n"); + av_log(NULL, AV_LOG_ERROR, "Not implemented!\n"); break; case DCA_CHANNEL: case DCA_STEREO: @@ -790,10 +869,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) { int k, l; int subsubframe = s->current_subsubframe; - - const float *quant_step_table; - - LOCAL_ALIGNED_16(int32_t, block, [SAMPLES_PER_SUBBAND * DCA_SUBBANDS]); + const uint32_t *quant_step_table; /* * Audio data @@ -801,13 +877,12 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) /* Select quantization step size table */ if (s->bit_rate_index == 0x1f) - quant_step_table = ff_dca_lossless_quant_d; + quant_step_table = ff_dca_lossless_quant; else - quant_step_table = ff_dca_lossy_quant_d; + quant_step_table = ff_dca_lossy_quant; for (k = base_channel; k < s->audio_header.prim_channels; k++) { - float (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; - float rscale[DCA_SUBBANDS]; + int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index]; if (get_bits_left(&s->gb) < 0) return AVERROR_INVALIDDATA; @@ -818,27 +893,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) /* Select the mid-tread linear quantizer */ int abits = s->dca_chan[k].bitalloc[l]; - float quant_step_size = quant_step_table[abits]; - - /* - * Determine quantization index code book and its type - */ - - /* Select quantization index code book */ - int sel = s->audio_header.quant_index_huffman[k][abits]; + uint32_t quant_step_size = quant_step_table[abits]; /* * Extract bits from the bit stream */ - if (!abits) { - rscale[l] = 0; - memset(block + SAMPLES_PER_SUBBAND * l, 0, SAMPLES_PER_SUBBAND * sizeof(block[0])); - } else { + if (!abits) + memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND * + sizeof(subband_samples[l][0])); + else { + uint32_t rscale; /* Deal with transients */ int sfi = s->dca_chan[k].transition_mode[l] && subsubframe >= s->dca_chan[k].transition_mode[l]; - rscale[l] = quant_step_size * s->dca_chan[k].scale_factor[l][sfi] * - s->audio_header.scalefactor_adj[k][sel]; + /* Determine quantization index code book and its type. + Select quantization index code book */ + int sel = s->audio_header.quant_index_huffman[k][abits]; + + rscale = (s->dca_chan[k].scale_factor[l][sfi] * + s->audio_header.scalefactor_adj[k][sel] + 8) >> 4; if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) { if (abits <= 7) { @@ -851,7 +924,7 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) block_code1 = get_bits(&s->gb, size); block_code2 = get_bits(&s->gb, size); err = decode_blockcodes(block_code1, block_code2, - levels, block + SAMPLES_PER_SUBBAND * l); + levels, subband_samples[l]); if (err) { av_log(s->avctx, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); @@ -860,20 +933,18 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) } else { /* no coding */ for (m = 0; m < SAMPLES_PER_SUBBAND; m++) - block[SAMPLES_PER_SUBBAND * l + m] = get_sbits(&s->gb, abits - 3); + subband_samples[l][m] = get_sbits(&s->gb, abits - 3); } } else { /* Huffman coded */ for (m = 0; m < SAMPLES_PER_SUBBAND; m++) - block[SAMPLES_PER_SUBBAND * l + m] = get_bitalloc(&s->gb, - &dca_smpl_bitalloc[abits], sel); + subband_samples[l][m] = get_bitalloc(&s->gb, + &dca_smpl_bitalloc[abits], sel); } + s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale); } } - s->fmt_conv.int32_to_float_fmul_array8(&s->fmt_conv, subband_samples[0], - block, rscale, SAMPLES_PER_SUBBAND * s->audio_header.vq_start_subband[k]); - for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) { int m; /* @@ -883,25 +954,25 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) int n; if (s->predictor_history) subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * - s->dca_chan[k].subband_samples_hist[l][3] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * - s->dca_chan[k].subband_samples_hist[l][2] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * - s->dca_chan[k].subband_samples_hist[l][1] + - ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * - s->dca_chan[k].subband_samples_hist[l][0]) * - (1.0f / 8192); + (int64_t)s->dca_chan[k].subband_samples_hist[l][3] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] * + (int64_t)s->dca_chan[k].subband_samples_hist[l][2] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] * + (int64_t)s->dca_chan[k].subband_samples_hist[l][1] + + ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] * + (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) + + (1 << 12) >> 13; for (m = 1; m < SAMPLES_PER_SUBBAND; m++) { - float sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * - subband_samples[l][m - 1]; + int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] * + (int64_t)subband_samples[l][m - 1]; for (n = 2; n <= 4; n++) if (m >= n) sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * - subband_samples[l][m - n]; + (int64_t)subband_samples[l][m - n]; else if (s->predictor_history) sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] * - s->dca_chan[k].subband_samples_hist[l][m - n + 4]; - subband_samples[l][m] += sum * 1.0f / 8192; + (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4]; + subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13); } } @@ -915,14 +986,15 @@ static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) * Decode VQ encoded high frequencies */ if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) { - if (!s->debug_flag & 0x01) { + if (!(s->debug_flag & 0x01)) { av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); s->debug_flag |= 0x01; } s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq, - ff_dca_high_freq_vq, subsubframe * SAMPLES_PER_SUBBAND, + ff_dca_high_freq_vq, + subsubframe * SAMPLES_PER_SUBBAND, s->dca_chan[k].scale_factor, s->audio_header.vq_start_subband[k], s->audio_header.subband_activity[k]); @@ -945,6 +1017,8 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) int k; if (upsample) { + LOCAL_ALIGNED(32, float, samples, [64], [SAMPLES_PER_SUBBAND]); + if (!s->qmf64_table) { s->qmf64_table = qmf64_precompute(); if (!s->qmf64_table) @@ -953,21 +1027,31 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) /* 64 subbands QMF */ for (k = 0; k < s->audio_header.prim_channels; k++) { - float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; + int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] = + s->dca_chan[k].subband_samples[block_index]; + + s->fmt_conv.int32_to_float(samples[0], subband_samples[0], + 64 * SAMPLES_PER_SUBBAND); if (s->channel_order_tab[k] >= 0) - qmf_64_subbands(s, k, subband_samples, + qmf_64_subbands(s, k, samples, s->samples_chanptr[s->channel_order_tab[k]], /* Upsampling needs a factor 2 here. */ M_SQRT2 / 32768.0); } } else { /* 32 subbands QMF */ + LOCAL_ALIGNED(32, float, samples, [32], [SAMPLES_PER_SUBBAND]); + for (k = 0; k < s->audio_header.prim_channels; k++) { - float (*subband_samples)[SAMPLES_PER_SUBBAND] = s->dca_chan[k].subband_samples[block_index]; + int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] = + s->dca_chan[k].subband_samples[block_index]; + + s->fmt_conv.int32_to_float(samples[0], subband_samples[0], + 32 * SAMPLES_PER_SUBBAND); if (s->channel_order_tab[k] >= 0) - qmf_32_subbands(s, k, subband_samples, + qmf_32_subbands(s, k, samples, s->samples_chanptr[s->channel_order_tab[k]], M_SQRT1_2 / 32768.0); } @@ -975,7 +1059,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) /* Generate LFE samples for this subsubframe FIXME!!! */ if (s->lfe) { - float *samples = s->samples_chanptr[ff_dca_lfe_index[s->amode]]; + float *samples = s->samples_chanptr[s->lfe_index]; lfe_interpolation_fir(s, s->lfe_data + 2 * s->lfe * (block_index + 4), samples); @@ -983,7 +1067,7 @@ static int dca_filter_channels(DCAContext *s, int block_index, int upsample) unsigned i; /* Should apply the filter in Table 6-11 when upsampling. For * now, just duplicate. */ - for (i = 511; i > 0; i--) { + for (i = 255; i > 0; i--) { samples[2 * i] = samples[2 * i + 1] = samples[i]; } @@ -1148,11 +1232,242 @@ static int dca_decode_block(DCAContext *s, int base_channel, int block_index) return 0; } +int ff_dca_xbr_parse_frame(DCAContext *s) +{ + int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2]; + int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX]; + int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS]; + int anctemp[DCA_CHSET_CHANS_MAX]; + int chset_fsize[DCA_CHSETS_MAX]; + int n_xbr_ch[DCA_CHSETS_MAX]; + int hdr_size, num_chsets, xbr_tmode, hdr_pos; + int i, j, k, l, chset, chan_base; + + av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n"); + + /* get bit position of sync header */ + hdr_pos = get_bits_count(&s->gb) - 32; + + hdr_size = get_bits(&s->gb, 6) + 1; + num_chsets = get_bits(&s->gb, 2) + 1; + + for(i = 0; i < num_chsets; i++) + chset_fsize[i] = get_bits(&s->gb, 14) + 1; + + xbr_tmode = get_bits1(&s->gb); + + for(i = 0; i < num_chsets; i++) { + n_xbr_ch[i] = get_bits(&s->gb, 3) + 1; + k = get_bits(&s->gb, 2) + 5; + for(j = 0; j < n_xbr_ch[i]; j++) { + active_bands[i][j] = get_bits(&s->gb, k) + 1; + if (active_bands[i][j] > DCA_SUBBANDS) { + av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]); + return AVERROR_INVALIDDATA; + } + } + } + + /* skip to the end of the header */ + i = get_bits_count(&s->gb); + if(hdr_pos + hdr_size * 8 > i) + skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); + + /* loop over the channel data sets */ + /* only decode as many channels as we've decoded base data for */ + for(chset = 0, chan_base = 0; + chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->audio_header.prim_channels; + chan_base += n_xbr_ch[chset++]) { + int start_posn = get_bits_count(&s->gb); + int subsubframe = 0; + int subframe = 0; + + /* loop over subframes */ + for (k = 0; k < (s->sample_blocks / 8); k++) { + /* parse header if we're on first subsubframe of a block */ + if(subsubframe == 0) { + /* Parse subframe header */ + for(i = 0; i < n_xbr_ch[chset]; i++) { + anctemp[i] = get_bits(&s->gb, 2) + 2; + } + + for(i = 0; i < n_xbr_ch[chset]; i++) { + get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]); + } + + for(i = 0; i < n_xbr_ch[chset]; i++) { + anctemp[i] = get_bits(&s->gb, 3); + if(anctemp[i] < 1) { + av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n"); + return AVERROR_INVALIDDATA; + } + } + + /* generate scale factors */ + for(i = 0; i < n_xbr_ch[chset]; i++) { + const uint32_t *scale_table; + int nbits; + int scale_table_size; + + if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) { + scale_table = ff_dca_scale_factor_quant7; + scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); + } else { + scale_table = ff_dca_scale_factor_quant6; + scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6); + } + + nbits = anctemp[i]; + + for(j = 0; j < active_bands[chset][i]; j++) { + if(abits_high[i][j] > 0) { + int index = get_bits(&s->gb, nbits); + if (index >= scale_table_size) { + av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); + return AVERROR_INVALIDDATA; + } + scale_table_high[i][j][0] = scale_table[index]; + + if(xbr_tmode && s->dca_chan[i].transition_mode[j]) { + int index = get_bits(&s->gb, nbits); + if (index >= scale_table_size) { + av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index); + return AVERROR_INVALIDDATA; + } + scale_table_high[i][j][1] = scale_table[index]; + } + } + } + } + } + + /* decode audio array for this block */ + for(i = 0; i < n_xbr_ch[chset]; i++) { + for(j = 0; j < active_bands[chset][i]; j++) { + const int xbr_abits = abits_high[i][j]; + const float quant_step_size = ff_dca_lossless_quant_d[xbr_abits]; + const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j]; + const float rscale = quant_step_size * scale_table_high[i][j][sfi]; + float *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j]; + int block[8]; + + if(xbr_abits <= 0) + continue; + + if(xbr_abits > 7) { + get_array(&s->gb, block, 8, xbr_abits - 3); + } else { + int block_code1, block_code2, size, levels, err; + + size = abits_sizes[xbr_abits - 1]; + levels = abits_levels[xbr_abits - 1]; + + block_code1 = get_bits(&s->gb, size); + block_code2 = get_bits(&s->gb, size); + err = decode_blockcodes(block_code1, block_code2, + levels, block); + if (err) { + av_log(s->avctx, AV_LOG_ERROR, + "ERROR: DTS-XBR: block code look-up failed\n"); + return AVERROR_INVALIDDATA; + } + } + + /* scale & sum into subband */ + for(l = 0; l < 8; l++) + subband_samples[l] += (float)block[l] * rscale; + } + } + + /* check DSYNC marker */ + if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) { + if(get_bits(&s->gb, 16) != 0xffff) { + av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n"); + return AVERROR_INVALIDDATA; + } + } + + /* advance sub-sub-frame index */ + if(++subsubframe >= s->subsubframes[subframe]) { + subsubframe = 0; + subframe++; + } + } + + /* skip to next channel set */ + i = get_bits_count(&s->gb); + if(start_posn + chset_fsize[chset] * 8 != i) { + j = start_posn + chset_fsize[chset] * 8 - i; + if(j < 0 || j >= 8) + av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set," + " skipping further than expected (%d bits)\n", j); + skip_bits_long(&s->gb, j); + } + } + + return 0; +} + + +/* parse initial header for XXCH and dump details */ +int ff_dca_xxch_decode_frame(DCAContext *s) +{ + int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos; + int i, chset, base_channel, chstart, fsize[8]; + + /* assume header word has already been parsed */ + hdr_pos = get_bits_count(&s->gb) - 32; + hdr_size = get_bits(&s->gb, 6) + 1; + /*chhdr_crc =*/ skip_bits1(&s->gb); + spkmsk_bits = get_bits(&s->gb, 5) + 1; + num_chsets = get_bits(&s->gb, 2) + 1; + + for (i = 0; i < num_chsets; i++) + fsize[i] = get_bits(&s->gb, 14) + 1; + + core_spk = get_bits(&s->gb, spkmsk_bits); + s->xxch_core_spkmask = core_spk; + s->xxch_nbits_spk_mask = spkmsk_bits; + s->xxch_dmix_embedded = 0; + + /* skip to the end of the header */ + i = get_bits_count(&s->gb); + if (hdr_pos + hdr_size * 8 > i) + skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); + + for (chset = 0; chset < num_chsets; chset++) { + chstart = get_bits_count(&s->gb); + base_channel = s->audio_header.prim_channels; + s->xxch_chset = chset; + + /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. + 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */ + dca_parse_audio_coding_header(s, base_channel, 1); + + /* decode channel data */ + for (i = 0; i < (s->sample_blocks / 8); i++) { + if (dca_decode_block(s, base_channel, i)) { + av_log(s->avctx, AV_LOG_ERROR, + "Error decoding DTS-XXCH extension\n"); + continue; + } + } + + /* skip to end of this section */ + i = get_bits_count(&s->gb); + if (chstart + fsize[chset] * 8 > i) + skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i); + } + s->xxch_chset = num_chsets; + + return 0; +} + static float dca_dmix_code(unsigned code) { int sign = (code >> 8) - 1; code &= 0xff; - return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1U << 15)); + return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15)); } static int scan_for_extensions(AVCodecContext *avctx) @@ -1164,7 +1479,7 @@ static int scan_for_extensions(AVCodecContext *avctx) /* only scan for extensions if ext_descr was unknown or indicated a * supported XCh extension */ - if (s->core_ext_mask < 0 || s->core_ext_mask & DCA_EXT_XCH) { + if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) { /* if ext_descr was unknown, clear s->core_ext_mask so that the * extensions scan can fill it up */ s->core_ext_mask = FFMAX(s->core_ext_mask, 0); @@ -1202,8 +1517,13 @@ static int scan_for_extensions(AVCodecContext *avctx) continue; } + if (s->xch_base_channel < 2) { + avpriv_request_sample(avctx, "XCh with fewer than 2 base channels"); + continue; + } + /* much like core primary audio coding header */ - dca_parse_audio_coding_header(s, s->xch_base_channel); + dca_parse_audio_coding_header(s, s->xch_base_channel, 0); for (i = 0; i < (s->sample_blocks / 8); i++) if ((ret = dca_decode_block(s, s->xch_base_channel, i))) { @@ -1219,6 +1539,7 @@ static int scan_for_extensions(AVCodecContext *avctx) /* usually found either in core or HD part in DTS-HD HRA streams, * but not in DTS-ES which contains XCh extensions instead */ s->core_ext_mask |= DCA_EXT_XXCH; + ff_dca_xxch_decode_frame(s); break; case 0x1d95f262: { @@ -1257,95 +1578,134 @@ static int scan_for_extensions(AVCodecContext *avctx) return ret; } -static int set_channel_layout(AVCodecContext *avctx, int channels, int num_core_channels) +static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels) { DCAContext *s = avctx->priv_data; - int i; + int i, j, chset, mask; + int channel_layout, channel_mask; + int posn, lavc; + + /* If we have XXCH then the channel layout is managed differently */ + /* note that XLL will also have another way to do things */ + if (!(s->core_ext_mask & DCA_EXT_XXCH)) { + /* xxx should also do MA extensions */ + if (s->amode < 16) { + avctx->channel_layout = ff_dca_core_channel_layout[s->amode]; + + if (s->audio_header.prim_channels + !!s->lfe > 2 && + avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { + /* + * Neither the core's auxiliary data nor our default tables contain + * downmix coefficients for the additional channel coded in the XCh + * extension, so when we're doing a Stereo downmix, don't decode it. + */ + s->xch_disable = 1; + } - if (s->amode < 16) { - avctx->channel_layout = dca_core_channel_layout[s->amode]; + if (s->xch_present && !s->xch_disable) { + if (avctx->channel_layout & AV_CH_BACK_CENTER) { + avpriv_request_sample(avctx, "XCh with Back center channel"); + return AVERROR_INVALIDDATA; + } + avctx->channel_layout |= AV_CH_BACK_CENTER; + if (s->lfe) { + avctx->channel_layout |= AV_CH_LOW_FREQUENCY; + s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode]; + } else { + s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode]; + } + if (s->channel_order_tab[s->xch_base_channel] < 0) + return AVERROR_INVALIDDATA; + } else { + *channels = num_core_channels + !!s->lfe; + s->xch_present = 0; /* disable further xch processing */ + if (s->lfe) { + avctx->channel_layout |= AV_CH_LOW_FREQUENCY; + s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode]; + } else + s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode]; + } - if (s->audio_header.prim_channels + !!s->lfe > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - /* - * Neither the core's auxiliary data nor our default tables contain - * downmix coefficients for the additional channel coded in the XCh - * extension, so when we're doing a Stereo downmix, don't decode it. - */ - s->xch_disable = 1; - } + if (*channels > !!s->lfe && + s->channel_order_tab[*channels - 1 - !!s->lfe] < 0) + return AVERROR_INVALIDDATA; - if (s->xch_present && !s->xch_disable) { - avctx->channel_layout |= AV_CH_BACK_CENTER; - if (s->lfe) { - avctx->channel_layout |= AV_CH_LOW_FREQUENCY; - s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode]; - } else { - s->channel_order_tab = ff_dca_channel_reorder_nolfe_xch[s->amode]; + if (av_get_channel_layout_nb_channels(avctx->channel_layout) != *channels) { + av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", *channels, av_get_channel_layout_nb_channels(avctx->channel_layout)); + return AVERROR_INVALIDDATA; } + + if (num_core_channels + !!s->lfe > 2 && + avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { + *channels = 2; + s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO; + avctx->channel_layout = AV_CH_LAYOUT_STEREO; + } + else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { + static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 }; + s->channel_order_tab = dca_channel_order_native; + } + s->lfe_index = ff_dca_lfe_index[s->amode]; } else { - channels = num_core_channels + !!s->lfe; - s->xch_present = 0; /* disable further xch processing */ - if (s->lfe) { - avctx->channel_layout |= AV_CH_LOW_FREQUENCY; - s->channel_order_tab = ff_dca_channel_reorder_lfe[s->amode]; - } else - s->channel_order_tab = ff_dca_channel_reorder_nolfe[s->amode]; + av_log(avctx, AV_LOG_ERROR, + "Non standard configuration %d !\n", s->amode); + return AVERROR_INVALIDDATA; } - if (channels > !!s->lfe && - s->channel_order_tab[channels - 1 - !!s->lfe] < 0) - return AVERROR_INVALIDDATA; + s->xxch_dmix_embedded = 0; + } else { + /* we only get here if an XXCH channel set can be added to the mix */ + channel_mask = s->xxch_core_spkmask; - if (num_core_channels + !!s->lfe > 2 && - avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { - channels = 2; - s->output = s->audio_header.prim_channels == 2 ? s->amode : DCA_STEREO; - avctx->channel_layout = AV_CH_LAYOUT_STEREO; + { + *channels = s->audio_header.prim_channels + !!s->lfe; + for (i = 0; i < s->xxch_chset; i++) { + channel_mask |= s->xxch_spk_masks[i]; + } + } - /* Stereo downmix coefficients - * - * The decoder can only downmix to 2-channel, so we need to ensure - * embedded downmix coefficients are actually targeting 2-channel. - */ - if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO || - s->core_downmix_amode == DCA_STEREO_TOTAL)) { - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - /* Range checked earlier */ - s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]); - s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]); - } - s->output = s->core_downmix_amode; - } else { - int am = s->amode & DCA_CHANNEL_MASK; - if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) { - av_log(s->avctx, AV_LOG_ERROR, - "Invalid channel mode %d\n", am); - return AVERROR_INVALIDDATA; - } - if (num_core_channels + !!s->lfe > - FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) { - avpriv_request_sample(s->avctx, "Downmixing %d channels", - s->audio_header.prim_channels + !!s->lfe); - return AVERROR_PATCHWELCOME; - } - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0]; - s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1]; - } + /* Given the DTS spec'ed channel mask, generate an avcodec version */ + channel_layout = 0; + for (i = 0; i < s->xxch_nbits_spk_mask; ++i) { + if (channel_mask & (1 << i)) { + channel_layout |= ff_dca_map_xxch_to_native[i]; } - ff_dlog(s->avctx, "Stereo downmix coeffs:\n"); - for (i = 0; i < num_core_channels + !!s->lfe; i++) { - ff_dlog(s->avctx, "L, input channel %d = %f\n", i, - s->downmix_coef[i][0]); - ff_dlog(s->avctx, "R, input channel %d = %f\n", i, - s->downmix_coef[i][1]); + } + + /* make sure that we have managed to get equivalent dts/avcodec channel + * masks in some sense -- unfortunately some channels could overlap */ + if (av_popcount(channel_mask) != av_popcount(channel_layout)) { + av_log(avctx, AV_LOG_DEBUG, + "DTS-XXCH: Inconsistent avcodec/dts channel layouts\n"); + return AVERROR_INVALIDDATA; + } + + avctx->channel_layout = channel_layout; + + if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) { + /* Estimate DTS --> avcodec ordering table */ + for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) { + mask = chset >= 0 ? s->xxch_spk_masks[chset] + : s->xxch_core_spkmask; + for (i = 0; i < s->xxch_nbits_spk_mask; i++) { + if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) { + lavc = ff_dca_map_xxch_to_native[i]; + posn = av_popcount(channel_layout & (lavc - 1)); + s->xxch_order_tab[j++] = posn; + } + } + } - ff_dlog(s->avctx, "\n"); + + s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1)); + } else { /* native ordering */ + for (i = 0; i < *channels; i++) + s->xxch_order_tab[i] = i; + + s->lfe_index = *channels - 1; } - } else { - av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n", s->amode); - return AVERROR_INVALIDDATA; + + s->channel_order_tab = s->xxch_order_tab; } return 0; @@ -1361,20 +1721,30 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, AVFrame *frame = data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; - int lfe_samples; int num_core_channels = 0; int i, ret; - float **samples_flt; + float **samples_flt; + float *src_chan; + float *dst_chan; DCAContext *s = avctx->priv_data; int channels, full_channels; + float scale; + int achan; + int chset; + int mask; + int j, k; + int endch; int upsample = 0; s->exss_ext_mask = 0; s->xch_present = 0; - s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer, - DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE); + s->dca_buffer_size = AVERROR_INVALIDDATA; + for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++) + s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, s->dca_buffer, + DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE); + if (s->dca_buffer_size == AVERROR_INVALIDDATA) { av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); return AVERROR_INVALIDDATA; @@ -1386,7 +1756,6 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, } // set AVCodec values with parsed data avctx->sample_rate = s->sample_rate; - avctx->bit_rate = s->bit_rate; s->profile = FF_PROFILE_DTS; @@ -1400,8 +1769,51 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, /* record number of core channels incase less than max channels are requested */ num_core_channels = s->audio_header.prim_channels; + if (s->audio_header.prim_channels + !!s->lfe > 2 && + avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) { + /* Stereo downmix coefficients + * + * The decoder can only downmix to 2-channel, so we need to ensure + * embedded downmix coefficients are actually targeting 2-channel. + */ + if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO || + s->core_downmix_amode == DCA_STEREO_TOTAL)) { + for (i = 0; i < num_core_channels + !!s->lfe; i++) { + /* Range checked earlier */ + s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]); + s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]); + } + s->output = s->core_downmix_amode; + } else { + int am = s->amode & DCA_CHANNEL_MASK; + if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) { + av_log(s->avctx, AV_LOG_ERROR, + "Invalid channel mode %d\n", am); + return AVERROR_INVALIDDATA; + } + if (num_core_channels + !!s->lfe > + FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) { + avpriv_request_sample(s->avctx, "Downmixing %d channels", + s->audio_header.prim_channels + !!s->lfe); + return AVERROR_PATCHWELCOME; + } + for (i = 0; i < num_core_channels + !!s->lfe; i++) { + s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0]; + s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1]; + } + } + ff_dlog(s->avctx, "Stereo downmix coeffs:\n"); + for (i = 0; i < num_core_channels + !!s->lfe; i++) { + ff_dlog(s->avctx, "L, input channel %d = %f\n", i, + s->downmix_coef[i][0]); + ff_dlog(s->avctx, "R, input channel %d = %f\n", i, + s->downmix_coef[i][1]); + } + ff_dlog(s->avctx, "\n"); + } + if (s->ext_coding) - s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr]; + s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr]; else s->core_ext_mask = 0; @@ -1411,10 +1823,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, full_channels = channels = s->audio_header.prim_channels + !!s->lfe; - ret = set_channel_layout(avctx, channels, num_core_channels); + ret = set_channel_layout(avctx, &channels, num_core_channels); if (ret < 0) return ret; - avctx->channels = channels; /* get output buffer */ frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND); @@ -1445,20 +1856,24 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, /* If downmixing to stereo, don't decode additional channels. * FIXME: Using the xch_disable flag for this doesn't seem right. */ if (!s->xch_disable) - avctx->channels += s->xll_channels - s->xll_residual_channels; + channels = s->xll_channels; } } + if (avctx->channels != channels) { + if (avctx->channels) + av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels); + avctx->channels = channels; + } + /* FIXME: This is an ugly hack, to just revert to the default * layout if we have additional channels. Need to convert the XLL - * channel masks to libav channel_layout mask. */ + * channel masks to ffmpeg channel_layout mask. */ if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels) avctx->channel_layout = 0; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) { - av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; - } samples_flt = (float **) frame->extended_data; /* allocate buffer for extra channels if downmixing */ @@ -1499,8 +1914,55 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]]; float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]]; float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]]; - s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256); - s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256); + s->fdsp->vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256); + s->fdsp->vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256); + } + + /* If stream contains XXCH, we might need to undo an embedded downmix */ + if (s->xxch_dmix_embedded) { + /* Loop over channel sets in turn */ + ch = num_core_channels; + for (chset = 0; chset < s->xxch_chset; chset++) { + endch = ch + s->xxch_chset_nch[chset]; + mask = s->xxch_dmix_embedded; + + /* undo downmix */ + for (j = ch; j < endch; j++) { + if (mask & (1 << j)) { /* this channel has been mixed-out */ + src_chan = s->samples_chanptr[s->channel_order_tab[j]]; + for (k = 0; k < endch; k++) { + achan = s->channel_order_tab[k]; + scale = s->xxch_dmix_coeff[j][k]; + if (scale != 0.0) { + dst_chan = s->samples_chanptr[achan]; + s->fdsp->vector_fmac_scalar(dst_chan, src_chan, + -scale, 256); + } + } + } + } + + /* if a downmix has been embedded then undo the pre-scaling */ + if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) { + scale = s->xxch_dmix_sf[chset]; + + for (j = 0; j < ch; j++) { + src_chan = s->samples_chanptr[s->channel_order_tab[j]]; + for (k = 0; k < 256; k++) + src_chan[k] *= scale; + } + + /* LFE channel is always part of core, scale if it exists */ + if (s->lfe) { + src_chan = s->samples_chanptr[s->lfe_index]; + for (k = 0; k < 256; k++) + src_chan[k] *= scale; + } + } + + ch = endch; + } + } } @@ -1523,6 +1985,9 @@ static int dca_decode_frame(AVCodecContext *avctx, void *data, if (ret < 0) return ret; + if ( avctx->profile != FF_PROFILE_DTS_HD_MA + && avctx->profile != FF_PROFILE_DTS_HD_HRA) + avctx->bit_rate = s->bit_rate; *got_frame_ptr = 1; return buf_size; @@ -1541,7 +2006,10 @@ static av_cold int dca_decode_init(AVCodecContext *avctx) s->avctx = avctx; dca_init_vlcs(); - avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT); + s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); + if (!s->fdsp) + return AVERROR(ENOMEM); + ff_mdct_init(&s->imdct, 6, 1, 1.0); ff_synth_filter_init(&s->synth); ff_dcadsp_init(&s->dcadsp); @@ -1562,14 +2030,15 @@ static av_cold int dca_decode_end(AVCodecContext *avctx) DCAContext *s = avctx->priv_data; ff_mdct_end(&s->imdct); av_freep(&s->extra_channels_buffer); + av_freep(&s->fdsp); av_freep(&s->xll_sample_buf); av_freep(&s->qmf64_table); return 0; } static const AVOption options[] = { - { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, - { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, + { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, + { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM }, { NULL }, }; @@ -1578,6 +2047,7 @@ static const AVClass dca_decoder_class = { .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, + .category = AV_CLASS_CATEGORY_DECODER, }; AVCodec ff_dca_decoder = {