X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fdca_core.c;h=6cb1f30a3c161da7e73745d3e46f87f872048f63;hb=f386dd70acdc81d42d6bcb885d2889634cdf45b7;hp=d5e628e763b0c2f0f029cda6f13be54378266010;hpb=3f17751eeb7e3348576e2597884d5e5155aadcfb;p=ffmpeg diff --git a/libavcodec/dca_core.c b/libavcodec/dca_core.c index d5e628e763b..6cb1f30a3c1 100644 --- a/libavcodec/dca_core.c +++ b/libavcodec/dca_core.c @@ -18,6 +18,7 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ +#include "dcaadpcm.h" #include "dcadec.h" #include "dcadata.h" #include "dcahuff.h" @@ -34,35 +35,7 @@ enum HeaderType { 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] = { +static const int8_t prm_ch_to_spkr_map[DCA_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 }, @@ -75,7 +48,7 @@ static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = { { 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] = { +static const uint8_t audio_mode_ch_mask[DCA_AMODE_COUNT] = { DCA_SPEAKER_LAYOUT_MONO, DCA_SPEAKER_LAYOUT_STEREO, DCA_SPEAKER_LAYOUT_STEREO, @@ -108,114 +81,68 @@ static void get_array(GetBitContext *s, int32_t *array, int size, int 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); + DCACoreFrameHeader h = { 0 }; + int err = ff_dca_parse_core_frame_header(&h, &s->gb); - // HDCD mastering flag - skip_bits1(&s->gb); + if (err < 0) { + switch (err) { + case DCA_PARSE_ERROR_DEFICIT_SAMPLES: + av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n"); + return h.normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME; - // Extension audio descriptor flag - s->ext_audio_type = get_bits(&s->gb, 3); + case DCA_PARSE_ERROR_PCM_BLOCKS: + av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", h.npcmblocks); + return (h.npcmblocks < 6 || h.normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME; - // 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; - } + case DCA_PARSE_ERROR_FRAME_SIZE: + av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", h.frame_size); + return AVERROR_INVALIDDATA; - // Predictor history flag switch - s->predictor_history = get_bits1(&s->gb); + case DCA_PARSE_ERROR_AMODE: + av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", h.audio_mode); + return AVERROR_PATCHWELCOME; - // Header CRC check bytes - if (s->crc_present) - skip_bits(&s->gb, 16); + case DCA_PARSE_ERROR_SAMPLE_RATE: + av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n"); + return AVERROR_INVALIDDATA; - // Multirate interpolator switch - s->filter_perfect = get_bits1(&s->gb); + case DCA_PARSE_ERROR_RESERVED_BIT: + av_log(s->avctx, AV_LOG_ERROR, "Reserved bit set\n"); + return AVERROR_INVALIDDATA; - // Encoder software revision - skip_bits(&s->gb, 4); + case DCA_PARSE_ERROR_LFE_FLAG: + av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n"); + return AVERROR_INVALIDDATA; - // Copy history - skip_bits(&s->gb, 2); + case DCA_PARSE_ERROR_PCM_RES: + av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n"); + return AVERROR_INVALIDDATA; - // 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; + default: + av_log(s->avctx, AV_LOG_ERROR, "Unknown core frame header error\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); + s->crc_present = h.crc_present; + s->npcmblocks = h.npcmblocks; + s->frame_size = h.frame_size; + s->audio_mode = h.audio_mode; + s->sample_rate = avpriv_dca_sample_rates[h.sr_code]; + s->bit_rate = ff_dca_bit_rates[h.br_code]; + s->drc_present = h.drc_present; + s->ts_present = h.ts_present; + s->aux_present = h.aux_present; + s->ext_audio_type = h.ext_audio_type; + s->ext_audio_present = h.ext_audio_present; + s->sync_ssf = h.sync_ssf; + s->lfe_present = h.lfe_present; + s->predictor_history = h.predictor_history; + s->filter_perfect = h.filter_perfect; + s->source_pcm_res = ff_dca_bits_per_sample[h.pcmr_code]; + s->es_format = h.pcmr_code & 1; + s->sumdiff_front = h.sumdiff_front; + s->sumdiff_surround = h.sumdiff_surround; return 0; } @@ -670,46 +597,21 @@ static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, in 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; + int i, j; for (i = sb_start; i < sb_end; i++) { if (prediction_mode[i]) { - const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]]; + const int pred_id = 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))); + int32_t x = ff_dcaadpcm_predict(pred_id, ptr + j - DCA_ADPCM_COEFFS); + ptr[j] = clip23(ptr[j] + x); } } } @@ -817,8 +719,8 @@ static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType heade scale = clip23(adj * scale >> 22); } - dequantize(s->subband_samples[ch][band] + ofs, - audio, step_size, scale, 0); + ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs, + audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES); } } @@ -1146,8 +1048,8 @@ static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchann else scale = xbr_scale_factors[ch][band][1]; - dequantize(s->subband_samples[ch][band] + ofs, - audio, step_size, scale, 1); + ff_dca_core_dequantize(s->subband_samples[ch][band] + ofs, + audio, step_size, scale, 1, DCA_SUBBAND_SAMPLES); } } @@ -1326,8 +1228,8 @@ static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int // 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); + ff_dca_core_dequantize(s->x96_subband_samples[ch][band] + ofs, + audio, step_size, scale, 0, DCA_SUBBAND_SAMPLES); } } @@ -1802,12 +1704,13 @@ static int parse_optional_info(DCACoreDecoder *s) 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; + uint32_t w1, w2 = 0; // 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: + case DCA_EXT_AUDIO_XCH: if (dca->request_channel_layout) break; @@ -1816,15 +1719,15 @@ static int parse_optional_info(DCACoreDecoder *s) // 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; + for (; sync_pos >= last_pos; sync_pos--, w2 = w1) { + w1 = AV_RB32(s->gb.buffer + sync_pos * 4); + if (w1 == DCA_SYNCWORD_XCH) { + size = (w2 >> 22) + 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); + && (w2 >> 15 & 0x7f) == 0x08) { + s->xch_pos = sync_pos * 32 + 49; break; } } @@ -1837,17 +1740,17 @@ static int parse_optional_info(DCACoreDecoder *s) } break; - case EXT_AUDIO_X96: + case DCA_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; + for (; sync_pos >= last_pos; sync_pos--, w2 = w1) { + w1 = AV_RB32(s->gb.buffer + sync_pos * 4); + if (w1 == DCA_SYNCWORD_X96) { + size = (w2 >> 20) + 1; dist = s->frame_size - sync_pos * 4; if (size >= 96 && size == dist) { - s->x96_pos = get_bits_count(&s->gb); + s->x96_pos = sync_pos * 32 + 44; break; } } @@ -1860,16 +1763,16 @@ static int parse_optional_info(DCACoreDecoder *s) } break; - case EXT_AUDIO_XXCH: + case DCA_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; + for (; sync_pos >= last_pos; sync_pos--, w2 = w1) { + w1 = AV_RB32(s->gb.buffer + sync_pos * 4); + if (w1 == DCA_SYNCWORD_XXCH) { + size = (w2 >> 26) + 1; dist = s->gb.size_in_bits / 8 - sync_pos * 4; if (size >= 11 && size <= dist && !av_crc(dca->crctab, 0xffff, s->gb.buffer + @@ -1901,8 +1804,8 @@ int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size) if ((ret = init_get_bits8(&s->gb, data, size)) < 0) return ret; + s->gb_in = s->gb; - skip_bits_long(&s->gb, 32); if ((ret = parse_frame_header(s)) < 0) return ret; if ((ret = alloc_sample_buffer(s)) < 0) @@ -1929,7 +1832,6 @@ 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; @@ -1941,11 +1843,13 @@ int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset ret = parse_xxch_frame(s); ext = DCA_EXSS_XXCH; } else if (s->xxch_pos) { - s->gb.index = s->xxch_pos; + s->gb = s->gb_in; + skip_bits_long(&s->gb, s->xxch_pos); ret = parse_xxch_frame(s); ext = DCA_CSS_XXCH; } else if (s->xch_pos) { - s->gb.index = s->xch_pos; + s->gb = s->gb_in; + skip_bits_long(&s->gb, s->xch_pos); ret = parse_xch_frame(s); ext = DCA_CSS_XCH; } @@ -1987,8 +1891,8 @@ int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset s->ext_audio_mask |= DCA_EXSS_X96; } } else if (s->x96_pos) { - s->gb = gb; - s->gb.index = s->x96_pos; + s->gb = s->gb_in; + skip_bits_long(&s->gb, s->x96_pos); if ((ret = parse_x96_frame(s)) < 0) { if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) return ret; @@ -2126,7 +2030,7 @@ int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth) int nlfesamples = s->npcmblocks >> 1; // Check LFF - if (s->lfe_present == LFE_FLAG_128) { + if (s->lfe_present == DCA_LFE_FLAG_128) { av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n"); return AVERROR(EINVAL); } @@ -2176,7 +2080,7 @@ static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame) // Undo embedded XCH downmix if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH) - && s->audio_mode >= AMODE_2F2R) { + && s->audio_mode >= DCA_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], @@ -2220,15 +2124,15 @@ static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame) 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) { + if ((s->sumdiff_front && s->audio_mode > DCA_AMODE_MONO) + || s->audio_mode == DCA_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) { + if (s->sumdiff_surround && s->audio_mode >= DCA_AMODE_2F2R) { s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls], s->output_samples[DCA_SPEAKER_Rs], nsamples); @@ -2332,7 +2236,7 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame) // Filter LFE channel if (s->lfe_present) { - int dec_select = (s->lfe_present == LFE_FLAG_128); + int dec_select = (s->lfe_present == DCA_LFE_FLAG_128); float *samples = output_samples[DCA_SPEAKER_LFE1]; int nlfesamples = s->npcmblocks >> (dec_select + 1); @@ -2366,7 +2270,7 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame) // Undo embedded XCH downmix if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH) - && s->audio_mode >= AMODE_2F2R) { + && s->audio_mode >= DCA_AMODE_2F2R) { s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls], output_samples[DCA_SPEAKER_Cs], -M_SQRT1_2, nsamples); @@ -2413,15 +2317,15 @@ static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame) 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) { + if ((s->sumdiff_front && s->audio_mode > DCA_AMODE_MONO) + || s->audio_mode == DCA_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) { + if (s->sumdiff_surround && s->audio_mode >= DCA_AMODE_2F2R) { s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls], output_samples[DCA_SPEAKER_Rs], nsamples); @@ -2448,7 +2352,7 @@ int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame) // Handle downmixing to stereo request if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO - && s->audio_mode > AMODE_MONO && s->prim_dmix_embedded + && s->audio_mode > DCA_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; @@ -2481,8 +2385,8 @@ int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame) 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)) + if (s->audio_mode == DCA_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;