X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fac3dec.c;h=2966c33b25dc0901f0ddcc501f36c93ba5e690df;hb=75abcdb3915e3abb2dc6b5f7d101c177dcfdb626;hp=5feb1895199681e65a145a81d2b93e6ca47b0718;hpb=ae04de316f03d038b21078c803007cb8f955777a;p=ffmpeg diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index 5feb1895199..2966c33b25d 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -7,20 +7,20 @@ * Copyright (c) 2007-2008 Bartlomiej Wolowiec * Copyright (c) 2007 Justin Ruggles * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -35,6 +35,7 @@ #include "ac3_parser.h" #include "ac3dec.h" #include "ac3dec_data.h" +#include "kbdwin.h" /** Large enough for maximum possible frame size when the specification limit is ignored */ #define AC3_FRAME_BUFFER_SIZE 32768 @@ -66,16 +67,6 @@ static const uint8_t quantization_tab[16] = { static float dynamic_range_tab[256]; /** Adjustments in dB gain */ -#define LEVEL_PLUS_3DB 1.4142135623730950 -#define LEVEL_PLUS_1POINT5DB 1.1892071150027209 -#define LEVEL_MINUS_1POINT5DB 0.8408964152537145 -#define LEVEL_MINUS_3DB 0.7071067811865476 -#define LEVEL_MINUS_4POINT5DB 0.5946035575013605 -#define LEVEL_MINUS_6DB 0.5000000000000000 -#define LEVEL_MINUS_9DB 0.3535533905932738 -#define LEVEL_ZERO 0.0000000000000000 -#define LEVEL_ONE 1.0000000000000000 - static const float gain_levels[9] = { LEVEL_PLUS_3DB, LEVEL_PLUS_1POINT5DB, @@ -187,21 +178,23 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx) AC3DecodeContext *s = avctx->priv_data; s->avctx = avctx; - ac3_common_init(); + ff_ac3_common_init(); ac3_tables_init(); ff_mdct_init(&s->imdct_256, 8, 1, 1.0); ff_mdct_init(&s->imdct_512, 9, 1, 1.0); ff_kbd_window_init(s->window, 5.0, 256); dsputil_init(&s->dsp, avctx); + ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT); + ff_fmt_convert_init(&s->fmt_conv, avctx); av_lfg_init(&s->dith_state, 0); - /* set bias values for float to int16 conversion */ - if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { - s->add_bias = 385.0f; + /* set scale value for float to int16 conversion */ + if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT) { s->mul_bias = 1.0f; + avctx->sample_fmt = AV_SAMPLE_FMT_FLT; } else { - s->add_bias = 0.0f; s->mul_bias = 32767.0f; + avctx->sample_fmt = AV_SAMPLE_FMT_S16; } /* allow downmixing to stereo or mono */ @@ -213,13 +206,10 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx) s->downmixed = 1; /* allocate context input buffer */ - if (avctx->error_recognition >= FF_ER_CAREFUL) { s->input_buffer = av_mallocz(AC3_FRAME_BUFFER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); if (!s->input_buffer) - return AVERROR_NOMEM; - } + return AVERROR(ENOMEM); - avctx->sample_fmt = SAMPLE_FMT_S16; return 0; } @@ -279,6 +269,7 @@ static int parse_frame_header(AC3DecodeContext *s) /* get decoding parameters from header info */ s->bit_alloc_params.sr_code = hdr.sr_code; + s->bitstream_mode = hdr.bitstream_mode; s->channel_mode = hdr.channel_mode; s->channel_layout = hdr.channel_layout; s->lfe_on = hdr.lfe_on; @@ -314,9 +305,12 @@ static int parse_frame_header(AC3DecodeContext *s) s->skip_syntax = 1; memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht)); return ac3_parse_header(s); - } else { + } else if (CONFIG_EAC3_DECODER) { s->eac3 = 1; return ff_eac3_parse_header(s); + } else { + av_log(s->avctx, AV_LOG_ERROR, "E-AC-3 support not compiled in\n"); + return -1; } } @@ -409,24 +403,25 @@ static int decode_exponents(GetBitContext *gbc, int exp_strategy, int ngrps, */ static void calc_transform_coeffs_cpl(AC3DecodeContext *s) { - int i, j, ch, bnd, subbnd; + int bin, band, ch; - subbnd = -1; - i = s->start_freq[CPL_CH]; - for(bnd=0; bndnum_cpl_bands; bnd++) { - do { - subbnd++; - for(j=0; j<12; j++) { - for(ch=1; ch<=s->fbw_channels; ch++) { - if(s->channel_in_cpl[ch]) { - s->fixed_coeffs[ch][i] = ((int64_t)s->fixed_coeffs[CPL_CH][i] * (int64_t)s->cpl_coords[ch][bnd]) >> 23; - if (ch == 2 && s->phase_flags[bnd]) - s->fixed_coeffs[ch][i] = -s->fixed_coeffs[ch][i]; - } + bin = s->start_freq[CPL_CH]; + for (band = 0; band < s->num_cpl_bands; band++) { + int band_start = bin; + int band_end = bin + s->cpl_band_sizes[band]; + for (ch = 1; ch <= s->fbw_channels; ch++) { + if (s->channel_in_cpl[ch]) { + int cpl_coord = s->cpl_coords[ch][band] << 5; + for (bin = band_start; bin < band_end; bin++) { + s->fixed_coeffs[ch][bin] = MULH(s->fixed_coeffs[CPL_CH][bin] << 4, cpl_coord); + } + if (ch == 2 && s->phase_flags[band]) { + for (bin = band_start; bin < band_end; bin++) + s->fixed_coeffs[2][bin] = -s->fixed_coeffs[2][bin]; } - i++; } - } while(s->cpl_band_struct[subbnd]); + } + bin = band_end; } } @@ -453,6 +448,7 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma uint8_t *baps = s->bap[ch_index]; int8_t *exps = s->dexps[ch_index]; int *coeffs = s->fixed_coeffs[ch_index]; + int dither = (ch_index == CPL_CH) || s->dither_flag[ch_index]; GetBitContext *gbc = &s->gbc; int freq; @@ -461,7 +457,10 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma int mantissa; switch(bap){ case 0: - mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000; + if (dither) + mantissa = (av_lfg_get(&s->dith_state) & 0x7FFFFF) - 0x400000; + else + mantissa = 0; break; case 1: if(m->b1){ @@ -518,33 +517,18 @@ static void ac3_decode_transform_coeffs_ch(AC3DecodeContext *s, int ch_index, ma } /** - * Remove random dithering from coefficients with zero-bit mantissas + * Remove random dithering from coupling range coefficients with zero-bit + * mantissas for coupled channels which do not use dithering. * reference: Section 7.3.4 Dither for Zero Bit Mantissas (bap=0) */ static void remove_dithering(AC3DecodeContext *s) { int ch, i; - int end=0; - int *coeffs; - uint8_t *bap; for(ch=1; ch<=s->fbw_channels; ch++) { - if(!s->dither_flag[ch]) { - coeffs = s->fixed_coeffs[ch]; - bap = s->bap[ch]; - if(s->channel_in_cpl[ch]) - end = s->start_freq[CPL_CH]; - else - end = s->end_freq[ch]; - for(i=0; ichannel_in_cpl[ch]) { - bap = s->bap[CPL_CH]; - for(; iend_freq[CPL_CH]; i++) { - if(!bap[i]) - coeffs[i] = 0; - } + if(!s->dither_flag[ch] && s->channel_in_cpl[ch]) { + for(i = s->start_freq[CPL_CH]; iend_freq[CPL_CH]; i++) { + if(!s->bap[CPL_CH][i]) + s->fixed_coeffs[ch][i] = 0; } } } @@ -559,7 +543,7 @@ static void decode_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, /* if AHT is used, mantissas for all blocks are encoded in the first block of the frame. */ int bin; - if (!blk) + if (!blk && CONFIG_EAC3_DECODER) ff_eac3_decode_transform_coeffs_aht_ch(s, ch); for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) { s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin]; @@ -610,7 +594,6 @@ static void do_rematrixing(AC3DecodeContext *s) { int bnd, i; int end, bndend; - int tmp0, tmp1; end = FFMIN(s->end_freq[1], s->end_freq[2]); @@ -618,10 +601,9 @@ static void do_rematrixing(AC3DecodeContext *s) if(s->rematrixing_flags[bnd]) { bndend = FFMIN(end, ff_ac3_rematrix_band_tab[bnd+1]); for(i=ff_ac3_rematrix_band_tab[bnd]; ifixed_coeffs[1][i]; - tmp1 = s->fixed_coeffs[2][i]; - s->fixed_coeffs[1][i] = tmp0 + tmp1; - s->fixed_coeffs[2][i] = tmp0 - tmp1; + int tmp0 = s->fixed_coeffs[1][i]; + s->fixed_coeffs[1][i] += s->fixed_coeffs[2][i]; + s->fixed_coeffs[2][i] = tmp0 - s->fixed_coeffs[2][i]; } } } @@ -635,9 +617,6 @@ static void do_rematrixing(AC3DecodeContext *s) static inline void do_imdct(AC3DecodeContext *s, int channels) { int ch; - float add_bias = s->add_bias; - if(s->out_channels==1 && channels>1) - add_bias *= LEVEL_MINUS_3DB; // compensate for the gain in downmix for (ch=1; ch<=channels; ch++) { if (s->block_switch[ch]) { @@ -645,14 +624,14 @@ static inline void do_imdct(AC3DecodeContext *s, int channels) float *x = s->tmp_output+128; for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i]; - ff_imdct_half(&s->imdct_256, s->tmp_output, x); - s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128); + s->imdct_256.imdct_half(&s->imdct_256, s->tmp_output, x); + s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128); for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i+1]; - ff_imdct_half(&s->imdct_256, s->delay[ch-1], x); + s->imdct_256.imdct_half(&s->imdct_256, s->delay[ch-1], x); } else { - ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]); - s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, add_bias, 128); + s->imdct_512.imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]); + s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 128); memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float)); } } @@ -715,6 +694,10 @@ static void ac3_upmix_delay(AC3DecodeContext *s) /** * Decode band structure for coupling, spectral extension, or enhanced coupling. + * The band structure defines how many subbands are in each band. For each + * subband in the range, 1 means it is combined with the previous band, and 0 + * means that it starts a new band. + * * @param[in] gbc bit reader context * @param[in] blk block number * @param[in] eac3 flag to indicate E-AC-3 @@ -722,32 +705,33 @@ static void ac3_upmix_delay(AC3DecodeContext *s) * @param[in] start_subband subband number for start of range * @param[in] end_subband subband number for end of range * @param[in] default_band_struct default band structure table - * @param[out] band_struct decoded band structure * @param[out] num_bands number of bands (optionally NULL) * @param[out] band_sizes array containing the number of bins in each band (optionally NULL) */ static void decode_band_structure(GetBitContext *gbc, int blk, int eac3, int ecpl, int start_subband, int end_subband, const uint8_t *default_band_struct, - uint8_t *band_struct, int *num_bands, - uint8_t *band_sizes) + int *num_bands, uint8_t *band_sizes) { int subbnd, bnd, n_subbands, n_bands=0; uint8_t bnd_sz[22]; + uint8_t coded_band_struct[22]; + const uint8_t *band_struct; n_subbands = end_subband - start_subband; /* decode band structure from bitstream or use default */ if (!eac3 || get_bits1(gbc)) { for (subbnd = 0; subbnd < n_subbands - 1; subbnd++) { - band_struct[subbnd] = get_bits1(gbc); + coded_band_struct[subbnd] = get_bits1(gbc); } + band_struct = coded_band_struct; } else if (!blk) { - memcpy(band_struct, - &default_band_struct[start_subband+1], - n_subbands-1); + band_struct = &default_band_struct[start_subband+1]; + } else { + /* no change in band structure */ + return; } - band_struct[n_subbands-1] = 0; /* calculate number of bands and band sizes based on band structure. note that the first 4 subbands in enhanced coupling span only 6 bins @@ -819,14 +803,105 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* spectral extension strategy */ if (s->eac3 && (!blk || get_bits1(gbc))) { - if (get_bits1(gbc)) { - ff_log_missing_feature(s->avctx, "Spectral extension", 1); - return -1; + s->spx_in_use = get_bits1(gbc); + if (s->spx_in_use) { + int dst_start_freq, dst_end_freq, src_start_freq, + start_subband, end_subband; + + /* determine which channels use spx */ + if (s->channel_mode == AC3_CHMODE_MONO) { + s->channel_uses_spx[1] = 1; + } else { + for (ch = 1; ch <= fbw_channels; ch++) + s->channel_uses_spx[ch] = get_bits1(gbc); + } + + /* get the frequency bins of the spx copy region and the spx start + and end subbands */ + dst_start_freq = get_bits(gbc, 2); + start_subband = get_bits(gbc, 3) + 2; + if (start_subband > 7) + start_subband += start_subband - 7; + end_subband = get_bits(gbc, 3) + 5; + if (end_subband > 7) + end_subband += end_subband - 7; + dst_start_freq = dst_start_freq * 12 + 25; + src_start_freq = start_subband * 12 + 25; + dst_end_freq = end_subband * 12 + 25; + + /* check validity of spx ranges */ + if (start_subband >= end_subband) { + av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension " + "range (%d >= %d)\n", start_subband, end_subband); + return -1; + } + if (dst_start_freq >= src_start_freq) { + av_log(s->avctx, AV_LOG_ERROR, "invalid spectral extension " + "copy start bin (%d >= %d)\n", dst_start_freq, src_start_freq); + return -1; + } + + s->spx_dst_start_freq = dst_start_freq; + s->spx_src_start_freq = src_start_freq; + s->spx_dst_end_freq = dst_end_freq; + + decode_band_structure(gbc, blk, s->eac3, 0, + start_subband, end_subband, + ff_eac3_default_spx_band_struct, + &s->num_spx_bands, + s->spx_band_sizes); + } else { + for (ch = 1; ch <= fbw_channels; ch++) { + s->channel_uses_spx[ch] = 0; + s->first_spx_coords[ch] = 1; + } } - /* TODO: parse spectral extension strategy info */ } - /* TODO: spectral extension coordinates */ + /* spectral extension coordinates */ + if (s->spx_in_use) { + for (ch = 1; ch <= fbw_channels; ch++) { + if (s->channel_uses_spx[ch]) { + if (s->first_spx_coords[ch] || get_bits1(gbc)) { + float spx_blend; + int bin, master_spx_coord; + + s->first_spx_coords[ch] = 0; + spx_blend = get_bits(gbc, 5) * (1.0f/32); + master_spx_coord = get_bits(gbc, 2) * 3; + + bin = s->spx_src_start_freq; + for (bnd = 0; bnd < s->num_spx_bands; bnd++) { + int bandsize; + int spx_coord_exp, spx_coord_mant; + float nratio, sblend, nblend, spx_coord; + + /* calculate blending factors */ + bandsize = s->spx_band_sizes[bnd]; + nratio = ((float)((bin + (bandsize >> 1))) / s->spx_dst_end_freq) - spx_blend; + nratio = av_clipf(nratio, 0.0f, 1.0f); + nblend = sqrtf(3.0f * nratio); // noise is scaled by sqrt(3) to give unity variance + sblend = sqrtf(1.0f - nratio); + bin += bandsize; + + /* decode spx coordinates */ + spx_coord_exp = get_bits(gbc, 4); + spx_coord_mant = get_bits(gbc, 2); + if (spx_coord_exp == 15) spx_coord_mant <<= 1; + else spx_coord_mant += 4; + spx_coord_mant <<= (25 - spx_coord_exp - master_spx_coord); + spx_coord = spx_coord_mant * (1.0f/(1<<23)); + + /* multiply noise and signal blending factors by spx coordinate */ + s->spx_noise_blend [ch][bnd] = nblend * spx_coord; + s->spx_signal_blend[ch][bnd] = sblend * spx_coord; + } + } + } else { + s->first_spx_coords[ch] = 1; + } + } + } /* coupling strategy */ if (s->eac3 ? s->cpl_strategy_exists[blk] : get_bits1(gbc)) { @@ -845,7 +920,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* check for enhanced coupling */ if (s->eac3 && get_bits1(gbc)) { /* TODO: parse enhanced coupling strategy info */ - ff_log_missing_feature(s->avctx, "Enhanced coupling", 1); + av_log_missing_feature(s->avctx, "Enhanced coupling", 1); return -1; } @@ -863,9 +938,9 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) s->phase_flags_in_use = get_bits1(gbc); /* coupling frequency range */ - /* TODO: modify coupling end freq if spectral extension is used */ cpl_start_subband = get_bits(gbc, 4); - cpl_end_subband = get_bits(gbc, 4) + 3; + cpl_end_subband = s->spx_in_use ? (s->spx_src_start_freq - 37) / 12 : + get_bits(gbc, 4) + 3; if (cpl_start_subband >= cpl_end_subband) { av_log(s->avctx, AV_LOG_ERROR, "invalid coupling range (%d >= %d)\n", cpl_start_subband, cpl_end_subband); @@ -877,7 +952,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) decode_band_structure(gbc, blk, s->eac3, 0, cpl_start_subband, cpl_end_subband, ff_eac3_default_cpl_band_struct, - s->cpl_band_struct, &s->num_cpl_bands, NULL); + &s->num_cpl_bands, s->cpl_band_sizes); } else { /* coupling not in use */ for (ch = 1; ch <= fbw_channels; ch++) { @@ -938,13 +1013,16 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) if (channel_mode == AC3_CHMODE_STEREO) { if ((s->eac3 && !blk) || get_bits1(gbc)) { s->num_rematrixing_bands = 4; - if(cpl_in_use && s->start_freq[CPL_CH] <= 61) + if (cpl_in_use && s->start_freq[CPL_CH] <= 61) { s->num_rematrixing_bands -= 1 + (s->start_freq[CPL_CH] == 37); + } else if (s->spx_in_use && s->spx_src_start_freq <= 61) { + s->num_rematrixing_bands--; + } for(bnd=0; bndnum_rematrixing_bands; bnd++) s->rematrixing_flags[bnd] = get_bits1(gbc); } else if (!blk) { - av_log(s->avctx, AV_LOG_ERROR, "new rematrixing strategy must be present in block 0\n"); - return -1; + av_log(s->avctx, AV_LOG_WARNING, "Warning: new rematrixing strategy not present in block 0\n"); + s->num_rematrixing_bands = 0; } } @@ -964,6 +1042,8 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) int prev = s->end_freq[ch]; if (s->channel_in_cpl[ch]) s->end_freq[ch] = s->start_freq[CPL_CH]; + else if (s->channel_uses_spx[ch]) + s->end_freq[ch] = s->spx_src_start_freq; else { int bandwidth_code = get_bits(gbc, 6); if (bandwidth_code > 60) { @@ -1098,8 +1178,8 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* channel delta offset, len and bit allocation */ for (ch = !cpl_in_use; ch <= fbw_channels; ch++) { if (s->dba_mode[ch] == DBA_NEW) { - s->dba_nsegs[ch] = get_bits(gbc, 3); - for (seg = 0; seg <= s->dba_nsegs[ch]; seg++) { + s->dba_nsegs[ch] = get_bits(gbc, 3) + 1; + for (seg = 0; seg < s->dba_nsegs[ch]; seg++) { s->dba_offsets[ch][seg] = get_bits(gbc, 5); s->dba_lengths[ch][seg] = get_bits(gbc, 4); s->dba_values[ch][seg] = get_bits(gbc, 3); @@ -1139,7 +1219,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* Compute bit allocation */ const uint8_t *bap_tab = s->channel_uses_aht[ch] ? ff_eac3_hebap_tab : ff_ac3_bap_tab; - ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch], + s->ac3dsp.bit_alloc_calc_bap(s->mask[ch], s->psd[ch], s->start_freq[ch], s->end_freq[ch], s->snr_offset[ch], s->bit_alloc_params.floor, @@ -1160,8 +1240,6 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* TODO: generate enhanced coupling coordinates and uncouple */ - /* TODO: apply spectral extension */ - /* recover coefficients if rematrixing is in use */ if(s->channel_mode == AC3_CHMODE_STEREO) do_rematrixing(s); @@ -1170,11 +1248,16 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) for(ch=1; ch<=s->channels; ch++) { float gain = s->mul_bias / 4194304.0f; if(s->channel_mode == AC3_CHMODE_DUALMONO) { - gain *= s->dynamic_range[ch-1]; + gain *= s->dynamic_range[2-ch]; } else { gain *= s->dynamic_range[0]; } - s->dsp.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256); + s->fmt_conv.int32_to_float_fmul_scalar(s->transform_coeffs[ch], s->fixed_coeffs[ch], gain, 256); + } + + /* apply spectral extension to high frequency bins */ + if (s->spx_in_use && CONFIG_EAC3_DECODER) { + ff_eac3_apply_spectral_extension(s); } /* downmix and MDCT. order depends on whether block switching is used for @@ -1221,40 +1304,29 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; AC3DecodeContext *s = avctx->priv_data; - int16_t *out_samples = (int16_t *)data; + float *out_samples_flt = data; + int16_t *out_samples_s16 = data; int blk, ch, err; const uint8_t *channel_map; const float *output[AC3_MAX_CHANNELS]; - /* initialize the GetBitContext with the start of valid AC-3 Frame */ - if (s->input_buffer) { - /* copy input buffer to decoder context to avoid reading past the end - of the buffer, which can be caused by a damaged input stream. */ + /* copy input buffer to decoder context to avoid reading past the end + of the buffer, which can be caused by a damaged input stream. */ + if (buf_size >= 2 && AV_RB16(buf) == 0x770B) { + // seems to be byte-swapped AC-3 + int cnt = FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE) >> 1; + s->dsp.bswap16_buf((uint16_t *)s->input_buffer, (const uint16_t *)buf, cnt); + } else memcpy(s->input_buffer, buf, FFMIN(buf_size, AC3_FRAME_BUFFER_SIZE)); - init_get_bits(&s->gbc, s->input_buffer, buf_size * 8); - } else { - init_get_bits(&s->gbc, buf, buf_size * 8); - } + buf = s->input_buffer; + /* initialize the GetBitContext with the start of valid AC-3 Frame */ + init_get_bits(&s->gbc, buf, buf_size * 8); /* parse the syncinfo */ *data_size = 0; err = parse_frame_header(s); - /* check that reported frame size fits in input buffer */ - if(s->frame_size > buf_size) { - av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); - err = AAC_AC3_PARSE_ERROR_FRAME_SIZE; - } - - /* check for crc mismatch */ - if(err != AAC_AC3_PARSE_ERROR_FRAME_SIZE && avctx->error_recognition >= FF_ER_CAREFUL) { - if(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) { - av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n"); - err = AAC_AC3_PARSE_ERROR_CRC; - } - } - - if(err && err != AAC_AC3_PARSE_ERROR_CRC) { + if (err) { switch(err) { case AAC_AC3_PARSE_ERROR_SYNC: av_log(avctx, AV_LOG_ERROR, "frame sync error\n"); @@ -1282,6 +1354,18 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, av_log(avctx, AV_LOG_ERROR, "invalid header\n"); break; } + } else { + /* check that reported frame size fits in input buffer */ + if (s->frame_size > buf_size) { + av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); + err = AAC_AC3_PARSE_ERROR_FRAME_SIZE; + } else if (avctx->error_recognition >= FF_ER_CAREFUL) { + /* check for crc mismatch */ + if (av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0, &buf[2], s->frame_size-2)) { + av_log(avctx, AV_LOG_ERROR, "frame CRC mismatch\n"); + err = AAC_AC3_PARSE_ERROR_CRC; + } + } } /* if frame is ok, set audio parameters */ @@ -1313,6 +1397,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, if(s->out_channels < s->channels) s->output_mode = s->out_channels == 1 ? AC3_CHMODE_MONO : AC3_CHMODE_STEREO; } + /* set audio service type based on bitstream mode for AC-3 */ + avctx->audio_service_type = s->bitstream_mode; + if (s->bitstream_mode == 0x7 && s->channels > 1) + avctx->audio_service_type = AV_AUDIO_SERVICE_TYPE_KARAOKE; /* decode the audio blocks */ channel_map = ff_ac3_dec_channel_map[s->output_mode & ~AC3_OUTPUT_LFEON][s->lfe_on]; @@ -1323,11 +1411,19 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size, av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); err = 1; } - s->dsp.float_to_int16_interleave(out_samples, output, 256, s->out_channels); - out_samples += 256 * s->out_channels; + if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) { + s->fmt_conv.float_interleave(out_samples_flt, output, 256, + s->out_channels); + out_samples_flt += 256 * s->out_channels; + } else { + s->fmt_conv.float_to_int16_interleave(out_samples_s16, output, 256, + s->out_channels); + out_samples_s16 += 256 * s->out_channels; + } } - *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t); - return s->frame_size; + *data_size = s->num_blocks * 256 * avctx->channels * + (av_get_bits_per_sample_fmt(avctx->sample_fmt) / 8); + return FFMIN(buf_size, s->frame_size); } /** @@ -1344,24 +1440,32 @@ static av_cold int ac3_decode_end(AVCodecContext *avctx) return 0; } -AVCodec ac3_decoder = { +AVCodec ff_ac3_decoder = { .name = "ac3", - .type = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_AC3, .priv_data_size = sizeof (AC3DecodeContext), .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"), + .sample_fmts = (const enum AVSampleFormat[]) { + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE + }, }; -AVCodec eac3_decoder = { +#if CONFIG_EAC3_DECODER +AVCodec ff_eac3_decoder = { .name = "eac3", - .type = CODEC_TYPE_AUDIO, + .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_EAC3, .priv_data_size = sizeof (AC3DecodeContext), .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52B (AC-3, E-AC-3)"), + .sample_fmts = (const enum AVSampleFormat[]) { + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE + }, }; +#endif