/* * The simplest AC-3 encoder * Copyright (c) 2000 Fabrice Bellard * Copyright (c) 2006-2010 Justin Ruggles * Copyright (c) 2006-2010 Prakash Punnoor * * This file is part of Libav. * * 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. * * 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 Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * The simplest AC-3 encoder. */ //#define DEBUG //#define ASSERT_LEVEL 2 #include #include "libavutil/audioconvert.h" #include "libavutil/avassert.h" #include "libavutil/crc.h" #include "libavutil/opt.h" #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" #include "ac3dsp.h" #include "ac3.h" #include "audioconvert.h" #include "fft.h" #ifndef CONFIG_AC3ENC_FLOAT #define CONFIG_AC3ENC_FLOAT 0 #endif /** Maximum number of exponent groups. +1 for separate DC exponent. */ #define AC3_MAX_EXP_GROUPS 85 /* stereo rematrixing algorithms */ #define AC3_REMATRIXING_IS_STATIC 0x1 #define AC3_REMATRIXING_SUMS 0 #define AC3_REMATRIXING_NONE 1 #define AC3_REMATRIXING_ALWAYS 3 #if CONFIG_AC3ENC_FLOAT #define MAC_COEF(d,a,b) ((d)+=(a)*(b)) typedef float SampleType; typedef float CoefType; typedef float CoefSumType; #else #define MAC_COEF(d,a,b) MAC64(d,a,b) typedef int16_t SampleType; typedef int32_t CoefType; typedef int64_t CoefSumType; #endif typedef struct AC3MDCTContext { const SampleType *window; ///< MDCT window function FFTContext fft; ///< FFT context for MDCT calculation } AC3MDCTContext; /** * Encoding Options used by AVOption. */ typedef struct AC3EncOptions { /* AC-3 metadata options*/ int dialogue_level; int bitstream_mode; float center_mix_level; float surround_mix_level; int dolby_surround_mode; int audio_production_info; int mixing_level; int room_type; int copyright; int original; int extended_bsi_1; int preferred_stereo_downmix; float ltrt_center_mix_level; float ltrt_surround_mix_level; float loro_center_mix_level; float loro_surround_mix_level; int extended_bsi_2; int dolby_surround_ex_mode; int dolby_headphone_mode; int ad_converter_type; /* other encoding options */ int allow_per_frame_metadata; } AC3EncOptions; /** * Data for a single audio block. */ typedef struct AC3Block { uint8_t **bap; ///< bit allocation pointers (bap) CoefType **mdct_coef; ///< MDCT coefficients int32_t **fixed_coef; ///< fixed-point MDCT coefficients uint8_t **exp; ///< original exponents uint8_t **grouped_exp; ///< grouped exponents int16_t **psd; ///< psd per frequency bin int16_t **band_psd; ///< psd per critical band int16_t **mask; ///< masking curve uint16_t **qmant; ///< quantized mantissas uint8_t coeff_shift[AC3_MAX_CHANNELS]; ///< fixed-point coefficient shift values uint8_t new_rematrixing_strategy; ///< send new rematrixing flags in this block uint8_t rematrixing_flags[4]; ///< rematrixing flags struct AC3Block *exp_ref_block[AC3_MAX_CHANNELS]; ///< reference blocks for EXP_REUSE } AC3Block; /** * AC-3 encoder private context. */ typedef struct AC3EncodeContext { AVClass *av_class; ///< AVClass used for AVOption AC3EncOptions options; ///< encoding options PutBitContext pb; ///< bitstream writer context DSPContext dsp; AC3DSPContext ac3dsp; ///< AC-3 optimized functions AC3MDCTContext mdct; ///< MDCT context AC3Block blocks[AC3_MAX_BLOCKS]; ///< per-block info int bitstream_id; ///< bitstream id (bsid) int bitstream_mode; ///< bitstream mode (bsmod) int bit_rate; ///< target bit rate, in bits-per-second int sample_rate; ///< sampling frequency, in Hz int frame_size_min; ///< minimum frame size in case rounding is necessary int frame_size; ///< current frame size in bytes int frame_size_code; ///< frame size code (frmsizecod) uint16_t crc_inv[2]; int bits_written; ///< bit count (used to avg. bitrate) int samples_written; ///< sample count (used to avg. bitrate) int fbw_channels; ///< number of full-bandwidth channels (nfchans) int channels; ///< total number of channels (nchans) int lfe_on; ///< indicates if there is an LFE channel (lfeon) int lfe_channel; ///< channel index of the LFE channel int has_center; ///< indicates if there is a center channel int has_surround; ///< indicates if there are one or more surround channels int channel_mode; ///< channel mode (acmod) const uint8_t *channel_map; ///< channel map used to reorder channels int center_mix_level; ///< center mix level code int surround_mix_level; ///< surround mix level code int ltrt_center_mix_level; ///< Lt/Rt center mix level code int ltrt_surround_mix_level; ///< Lt/Rt surround mix level code int loro_center_mix_level; ///< Lo/Ro center mix level code int loro_surround_mix_level; ///< Lo/Ro surround mix level code int cutoff; ///< user-specified cutoff frequency, in Hz int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod) int nb_coefs[AC3_MAX_CHANNELS]; int rematrixing; ///< determines how rematrixing strategy is calculated int num_rematrixing_bands; ///< number of rematrixing bands /* bitrate allocation control */ int slow_gain_code; ///< slow gain code (sgaincod) int slow_decay_code; ///< slow decay code (sdcycod) int fast_decay_code; ///< fast decay code (fdcycod) int db_per_bit_code; ///< dB/bit code (dbpbcod) int floor_code; ///< floor code (floorcod) AC3BitAllocParameters bit_alloc; ///< bit allocation parameters int coarse_snr_offset; ///< coarse SNR offsets (csnroffst) int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod) int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst) int frame_bits_fixed; ///< number of non-coefficient bits for fixed parameters int frame_bits; ///< all frame bits except exponents and mantissas int exponent_bits; ///< number of bits used for exponents SampleType **planar_samples; uint8_t *bap_buffer; uint8_t *bap1_buffer; CoefType *mdct_coef_buffer; int32_t *fixed_coef_buffer; uint8_t *exp_buffer; uint8_t *grouped_exp_buffer; int16_t *psd_buffer; int16_t *band_psd_buffer; int16_t *mask_buffer; uint16_t *qmant_buffer; uint8_t exp_strategy[AC3_MAX_CHANNELS][AC3_MAX_BLOCKS]; ///< exponent strategies DECLARE_ALIGNED(16, SampleType, windowed_samples)[AC3_WINDOW_SIZE]; } AC3EncodeContext; typedef struct AC3Mant { uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr; ///< mantissa pointers for bap=1,2,4 int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4 } AC3Mant; #define CMIXLEV_NUM_OPTIONS 3 static const float cmixlev_options[CMIXLEV_NUM_OPTIONS] = { LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB }; #define SURMIXLEV_NUM_OPTIONS 3 static const float surmixlev_options[SURMIXLEV_NUM_OPTIONS] = { LEVEL_MINUS_3DB, LEVEL_MINUS_6DB, LEVEL_ZERO }; #define EXTMIXLEV_NUM_OPTIONS 8 static const float extmixlev_options[EXTMIXLEV_NUM_OPTIONS] = { LEVEL_PLUS_3DB, LEVEL_PLUS_1POINT5DB, LEVEL_ONE, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_ZERO }; #define OFFSET(param) offsetof(AC3EncodeContext, options.param) #define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM) static const AVOption options[] = { /* Metadata Options */ {"per_frame_metadata", "Allow Changing Metadata Per-Frame", OFFSET(allow_per_frame_metadata), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM}, /* downmix levels */ {"center_mixlev", "Center Mix Level", OFFSET(center_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_4POINT5DB, 0.0, 1.0, AC3ENC_PARAM}, {"surround_mixlev", "Surround Mix Level", OFFSET(surround_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_6DB, 0.0, 1.0, AC3ENC_PARAM}, /* audio production information */ {"mixing_level", "Mixing Level", OFFSET(mixing_level), FF_OPT_TYPE_INT, -1, -1, 111, AC3ENC_PARAM}, {"room_type", "Room Type", OFFSET(room_type), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "room_type"}, {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"}, {"large", "Large Room", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"}, {"small", "Small Room", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"}, /* other metadata options */ {"copyright", "Copyright Bit", OFFSET(copyright), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM}, {"dialnorm", "Dialogue Level (dB)", OFFSET(dialogue_level), FF_OPT_TYPE_INT, -31, -31, -1, AC3ENC_PARAM}, {"dsur_mode", "Dolby Surround Mode", OFFSET(dolby_surround_mode), FF_OPT_TYPE_INT, 0, 0, 2, AC3ENC_PARAM, "dsur_mode"}, {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"}, {"on", "Dolby Surround Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"}, {"off", "Not Dolby Surround Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"}, {"original", "Original Bit Stream", OFFSET(original), FF_OPT_TYPE_INT, 1, 0, 1, AC3ENC_PARAM}, /* extended bitstream information */ {"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dmix_mode"}, {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"}, {"ltrt", "Lt/Rt Downmix Preferred", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"}, {"loro", "Lo/Ro Downmix Preferred", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"}, {"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM}, {"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM}, {"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM}, {"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM}, {"dsurex_mode", "Dolby Surround EX Mode", OFFSET(dolby_surround_ex_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dsurex_mode"}, {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"}, {"on", "Dolby Surround EX Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"}, {"off", "Not Dolby Surround EX Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"}, {"dheadphone_mode", "Dolby Headphone Mode", OFFSET(dolby_headphone_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dheadphone_mode"}, {"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"}, {"on", "Dolby Headphone Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"}, {"off", "Not Dolby Headphone Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"}, {"ad_conv_type", "A/D Converter Type", OFFSET(ad_converter_type), FF_OPT_TYPE_INT, -1, -1, 1, AC3ENC_PARAM, "ad_conv_type"}, {"standard", "Standard (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"}, {"hdcd", "HDCD", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"}, {NULL} }; #if CONFIG_AC3ENC_FLOAT static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name, options, LIBAVUTIL_VERSION_INT }; #else static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name, options, LIBAVUTIL_VERSION_INT }; #endif /* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */ static av_cold void mdct_end(AC3MDCTContext *mdct); static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct, int nbits); static void apply_window(DSPContext *dsp, SampleType *output, const SampleType *input, const SampleType *window, unsigned int len); static int normalize_samples(AC3EncodeContext *s); static void scale_coefficients(AC3EncodeContext *s); /** * LUT for number of exponent groups. * exponent_group_tab[exponent strategy-1][number of coefficients] */ static uint8_t exponent_group_tab[3][256]; /** * List of supported channel layouts. */ static const int64_t ac3_channel_layouts[] = { AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_2_1, AV_CH_LAYOUT_SURROUND, AV_CH_LAYOUT_2_2, AV_CH_LAYOUT_QUAD, AV_CH_LAYOUT_4POINT0, AV_CH_LAYOUT_5POINT0, AV_CH_LAYOUT_5POINT0_BACK, (AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY), (AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY), AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_5POINT1_BACK, 0 }; /** * LUT to select the bandwidth code based on the bit rate, sample rate, and * number of full-bandwidth channels. * bandwidth_tab[fbw_channels-1][sample rate code][bit rate code] */ static const uint8_t ac3_bandwidth_tab[5][3][19] = { // 32 40 48 56 64 80 96 112 128 160 192 224 256 320 384 448 512 576 640 { { 0, 0, 0, 12, 16, 32, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48, 48 }, { 0, 0, 0, 16, 20, 36, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56, 56 }, { 0, 0, 0, 32, 40, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60 } }, { { 0, 0, 0, 0, 0, 0, 0, 20, 24, 32, 48, 48, 48, 48, 48, 48, 48, 48, 48 }, { 0, 0, 0, 0, 0, 0, 4, 24, 28, 36, 56, 56, 56, 56, 56, 56, 56, 56, 56 }, { 0, 0, 0, 0, 0, 0, 20, 44, 52, 60, 60, 60, 60, 60, 60, 60, 60, 60, 60 } }, { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 16, 24, 32, 40, 48, 48, 48, 48, 48, 48 }, { 0, 0, 0, 0, 0, 0, 0, 0, 4, 20, 28, 36, 44, 56, 56, 56, 56, 56, 56 }, { 0, 0, 0, 0, 0, 0, 0, 0, 20, 40, 48, 60, 60, 60, 60, 60, 60, 60, 60 } }, { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 24, 32, 48, 48, 48, 48, 48, 48 }, { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 16, 28, 36, 56, 56, 56, 56, 56, 56 }, { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 32, 48, 60, 60, 60, 60, 60, 60, 60 } }, { { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 8, 20, 32, 40, 48, 48, 48, 48 }, { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 12, 24, 36, 44, 56, 56, 56, 56 }, { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 44, 60, 60, 60, 60, 60, 60 } } }; /** * Adjust the frame size to make the average bit rate match the target bit rate. * This is only needed for 11025, 22050, and 44100 sample rates. */ static void adjust_frame_size(AC3EncodeContext *s) { while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) { s->bits_written -= s->bit_rate; s->samples_written -= s->sample_rate; } s->frame_size = s->frame_size_min + 2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate); s->bits_written += s->frame_size * 8; s->samples_written += AC3_FRAME_SIZE; } /** * Deinterleave input samples. * Channels are reordered from Libav's default order to AC-3 order. */ static void deinterleave_input_samples(AC3EncodeContext *s, const SampleType *samples) { int ch, i; /* deinterleave and remap input samples */ for (ch = 0; ch < s->channels; ch++) { const SampleType *sptr; int sinc; /* copy last 256 samples of previous frame to the start of the current frame */ memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][AC3_FRAME_SIZE], AC3_BLOCK_SIZE * sizeof(s->planar_samples[0][0])); /* deinterleave */ sinc = s->channels; sptr = samples + s->channel_map[ch]; for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) { s->planar_samples[ch][i] = *sptr; sptr += sinc; } } } /** * Apply the MDCT to input samples to generate frequency coefficients. * This applies the KBD window and normalizes the input to reduce precision * loss due to fixed-point calculations. */ static void apply_mdct(AC3EncodeContext *s) { int blk, ch; for (ch = 0; ch < s->channels; ch++) { for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE]; apply_window(&s->dsp, s->windowed_samples, input_samples, s->mdct.window, AC3_WINDOW_SIZE); block->coeff_shift[ch] = normalize_samples(s); s->mdct.fft.mdct_calcw(&s->mdct.fft, block->mdct_coef[ch], s->windowed_samples); } } } /** * Initialize stereo rematrixing. * If the strategy does not change for each frame, set the rematrixing flags. */ static void rematrixing_init(AC3EncodeContext *s) { if (s->channel_mode == AC3_CHMODE_STEREO) s->rematrixing = AC3_REMATRIXING_SUMS; else s->rematrixing = AC3_REMATRIXING_NONE; /* NOTE: AC3_REMATRIXING_ALWAYS might be used in the future in conjunction with channel coupling. */ if (s->rematrixing & AC3_REMATRIXING_IS_STATIC) { int flag = (s->rematrixing == AC3_REMATRIXING_ALWAYS); s->blocks[0].new_rematrixing_strategy = 1; memset(s->blocks[0].rematrixing_flags, flag, sizeof(s->blocks[0].rematrixing_flags)); } } /** * Determine rematrixing flags for each block and band. */ static void compute_rematrixing_strategy(AC3EncodeContext *s) { int nb_coefs; int blk, bnd, i; AC3Block *block, *block0; s->num_rematrixing_bands = 4; if (s->rematrixing & AC3_REMATRIXING_IS_STATIC) return; nb_coefs = FFMIN(s->nb_coefs[0], s->nb_coefs[1]); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { block = &s->blocks[blk]; block->new_rematrixing_strategy = !blk; for (bnd = 0; bnd < s->num_rematrixing_bands; bnd++) { /* calculate calculate sum of squared coeffs for one band in one block */ int start = ff_ac3_rematrix_band_tab[bnd]; int end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]); CoefSumType sum[4] = {0,}; for (i = start; i < end; i++) { CoefType lt = block->mdct_coef[0][i]; CoefType rt = block->mdct_coef[1][i]; CoefType md = lt + rt; CoefType sd = lt - rt; MAC_COEF(sum[0], lt, lt); MAC_COEF(sum[1], rt, rt); MAC_COEF(sum[2], md, md); MAC_COEF(sum[3], sd, sd); } /* compare sums to determine if rematrixing will be used for this band */ if (FFMIN(sum[2], sum[3]) < FFMIN(sum[0], sum[1])) block->rematrixing_flags[bnd] = 1; else block->rematrixing_flags[bnd] = 0; /* determine if new rematrixing flags will be sent */ if (blk && block->rematrixing_flags[bnd] != block0->rematrixing_flags[bnd]) { block->new_rematrixing_strategy = 1; } } block0 = block; } } /** * Apply stereo rematrixing to coefficients based on rematrixing flags. */ static void apply_rematrixing(AC3EncodeContext *s) { int nb_coefs; int blk, bnd, i; int start, end; uint8_t *flags; if (s->rematrixing == AC3_REMATRIXING_NONE) return; nb_coefs = FFMIN(s->nb_coefs[0], s->nb_coefs[1]); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; if (block->new_rematrixing_strategy) flags = block->rematrixing_flags; for (bnd = 0; bnd < s->num_rematrixing_bands; bnd++) { if (flags[bnd]) { start = ff_ac3_rematrix_band_tab[bnd]; end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]); for (i = start; i < end; i++) { int32_t lt = block->fixed_coef[0][i]; int32_t rt = block->fixed_coef[1][i]; block->fixed_coef[0][i] = (lt + rt) >> 1; block->fixed_coef[1][i] = (lt - rt) >> 1; } } } } } /** * Initialize exponent tables. */ static av_cold void exponent_init(AC3EncodeContext *s) { int i; for (i = 73; i < 256; i++) { exponent_group_tab[0][i] = (i - 1) / 3; exponent_group_tab[1][i] = (i + 2) / 6; exponent_group_tab[2][i] = (i + 8) / 12; } /* LFE */ exponent_group_tab[0][7] = 2; } /** * Extract exponents from the MDCT coefficients. * This takes into account the normalization that was done to the input samples * by adjusting the exponents by the exponent shift values. */ static void extract_exponents(AC3EncodeContext *s) { int blk, ch; for (ch = 0; ch < s->channels; ch++) { for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; s->ac3dsp.extract_exponents(block->exp[ch], block->fixed_coef[ch], AC3_MAX_COEFS); } } } /** * Exponent Difference Threshold. * New exponents are sent if their SAD exceed this number. */ #define EXP_DIFF_THRESHOLD 500 /** * Calculate exponent strategies for all blocks in a single channel. */ static void compute_exp_strategy_ch(AC3EncodeContext *s, uint8_t *exp_strategy, uint8_t *exp) { int blk, blk1; int exp_diff; /* estimate if the exponent variation & decide if they should be reused in the next frame */ exp_strategy[0] = EXP_NEW; exp += AC3_MAX_COEFS; for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) { exp_diff = s->dsp.sad[0](NULL, exp, exp - AC3_MAX_COEFS, 16, 16); if (exp_diff > EXP_DIFF_THRESHOLD) exp_strategy[blk] = EXP_NEW; else exp_strategy[blk] = EXP_REUSE; exp += AC3_MAX_COEFS; } /* now select the encoding strategy type : if exponents are often recoded, we use a coarse encoding */ blk = 0; while (blk < AC3_MAX_BLOCKS) { blk1 = blk + 1; while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1] == EXP_REUSE) blk1++; switch (blk1 - blk) { case 1: exp_strategy[blk] = EXP_D45; break; case 2: case 3: exp_strategy[blk] = EXP_D25; break; default: exp_strategy[blk] = EXP_D15; break; } blk = blk1; } } /** * Calculate exponent strategies for all channels. * Array arrangement is reversed to simplify the per-channel calculation. */ static void compute_exp_strategy(AC3EncodeContext *s) { int ch, blk; for (ch = 0; ch < s->fbw_channels; ch++) { compute_exp_strategy_ch(s, s->exp_strategy[ch], s->blocks[0].exp[ch]); } if (s->lfe_on) { ch = s->lfe_channel; s->exp_strategy[ch][0] = EXP_D15; for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) s->exp_strategy[ch][blk] = EXP_REUSE; } } /** * Update the exponents so that they are the ones the decoder will decode. */ static void encode_exponents_blk_ch(uint8_t *exp, int nb_exps, int exp_strategy) { int nb_groups, i, k; nb_groups = exponent_group_tab[exp_strategy-1][nb_exps] * 3; /* for each group, compute the minimum exponent */ switch(exp_strategy) { case EXP_D25: for (i = 1, k = 1; i <= nb_groups; i++) { uint8_t exp_min = exp[k]; if (exp[k+1] < exp_min) exp_min = exp[k+1]; exp[i] = exp_min; k += 2; } break; case EXP_D45: for (i = 1, k = 1; i <= nb_groups; i++) { uint8_t exp_min = exp[k]; if (exp[k+1] < exp_min) exp_min = exp[k+1]; if (exp[k+2] < exp_min) exp_min = exp[k+2]; if (exp[k+3] < exp_min) exp_min = exp[k+3]; exp[i] = exp_min; k += 4; } break; } /* constraint for DC exponent */ if (exp[0] > 15) exp[0] = 15; /* decrease the delta between each groups to within 2 so that they can be differentially encoded */ for (i = 1; i <= nb_groups; i++) exp[i] = FFMIN(exp[i], exp[i-1] + 2); i--; while (--i >= 0) exp[i] = FFMIN(exp[i], exp[i+1] + 2); /* now we have the exponent values the decoder will see */ switch (exp_strategy) { case EXP_D25: for (i = nb_groups, k = nb_groups * 2; i > 0; i--) { uint8_t exp1 = exp[i]; exp[k--] = exp1; exp[k--] = exp1; } break; case EXP_D45: for (i = nb_groups, k = nb_groups * 4; i > 0; i--) { exp[k] = exp[k-1] = exp[k-2] = exp[k-3] = exp[i]; k -= 4; } break; } } /** * Encode exponents from original extracted form to what the decoder will see. * This copies and groups exponents based on exponent strategy and reduces * deltas between adjacent exponent groups so that they can be differentially * encoded. */ static void encode_exponents(AC3EncodeContext *s) { int blk, blk1, ch; uint8_t *exp, *exp_strategy; int nb_coefs, num_reuse_blocks; for (ch = 0; ch < s->channels; ch++) { exp = s->blocks[0].exp[ch]; exp_strategy = s->exp_strategy[ch]; nb_coefs = s->nb_coefs[ch]; blk = 0; while (blk < AC3_MAX_BLOCKS) { blk1 = blk + 1; /* count the number of EXP_REUSE blocks after the current block and set exponent reference block pointers */ s->blocks[blk].exp_ref_block[ch] = &s->blocks[blk]; while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1] == EXP_REUSE) { s->blocks[blk1].exp_ref_block[ch] = &s->blocks[blk]; blk1++; } num_reuse_blocks = blk1 - blk - 1; /* for the EXP_REUSE case we select the min of the exponents */ s->ac3dsp.ac3_exponent_min(exp, num_reuse_blocks, nb_coefs); encode_exponents_blk_ch(exp, nb_coefs, exp_strategy[blk]); exp += AC3_MAX_COEFS * (num_reuse_blocks + 1); blk = blk1; } } } /** * Group exponents. * 3 delta-encoded exponents are in each 7-bit group. The number of groups * varies depending on exponent strategy and bandwidth. */ static void group_exponents(AC3EncodeContext *s) { int blk, ch, i; int group_size, nb_groups, bit_count; uint8_t *p; int delta0, delta1, delta2; int exp0, exp1; bit_count = 0; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; for (ch = 0; ch < s->channels; ch++) { int exp_strategy = s->exp_strategy[ch][blk]; if (exp_strategy == EXP_REUSE) continue; group_size = exp_strategy + (exp_strategy == EXP_D45); nb_groups = exponent_group_tab[exp_strategy-1][s->nb_coefs[ch]]; bit_count += 4 + (nb_groups * 7); p = block->exp[ch]; /* DC exponent */ exp1 = *p++; block->grouped_exp[ch][0] = exp1; /* remaining exponents are delta encoded */ for (i = 1; i <= nb_groups; i++) { /* merge three delta in one code */ exp0 = exp1; exp1 = p[0]; p += group_size; delta0 = exp1 - exp0 + 2; av_assert2(delta0 >= 0 && delta0 <= 4); exp0 = exp1; exp1 = p[0]; p += group_size; delta1 = exp1 - exp0 + 2; av_assert2(delta1 >= 0 && delta1 <= 4); exp0 = exp1; exp1 = p[0]; p += group_size; delta2 = exp1 - exp0 + 2; av_assert2(delta2 >= 0 && delta2 <= 4); block->grouped_exp[ch][i] = ((delta0 * 5 + delta1) * 5) + delta2; } } } s->exponent_bits = bit_count; } /** * Calculate final exponents from the supplied MDCT coefficients and exponent shift. * Extract exponents from MDCT coefficients, calculate exponent strategies, * and encode final exponents. */ static void process_exponents(AC3EncodeContext *s) { extract_exponents(s); compute_exp_strategy(s); encode_exponents(s); group_exponents(s); emms_c(); } /** * Count frame bits that are based solely on fixed parameters. * This only has to be run once when the encoder is initialized. */ static void count_frame_bits_fixed(AC3EncodeContext *s) { static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 }; int blk; int frame_bits; /* assumptions: * no dynamic range codes * no channel coupling * bit allocation parameters do not change between blocks * SNR offsets do not change between blocks * no delta bit allocation * no skipped data * no auxilliary data */ /* header size */ frame_bits = 65; frame_bits += frame_bits_inc[s->channel_mode]; /* audio blocks */ for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */ if (s->channel_mode == AC3_CHMODE_STEREO) { frame_bits++; /* rematstr */ } frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */ if (s->lfe_on) frame_bits++; /* lfeexpstr */ frame_bits++; /* baie */ frame_bits++; /* snr */ frame_bits += 2; /* delta / skip */ } frame_bits++; /* cplinu for block 0 */ /* bit alloc info */ /* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */ /* csnroffset[6] */ /* (fsnoffset[4] + fgaincod[4]) * c */ frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3); /* auxdatae, crcrsv */ frame_bits += 2; /* CRC */ frame_bits += 16; s->frame_bits_fixed = frame_bits; } /** * Initialize bit allocation. * Set default parameter codes and calculate parameter values. */ static void bit_alloc_init(AC3EncodeContext *s) { int ch; /* init default parameters */ s->slow_decay_code = 2; s->fast_decay_code = 1; s->slow_gain_code = 1; s->db_per_bit_code = 3; s->floor_code = 7; for (ch = 0; ch < s->channels; ch++) s->fast_gain_code[ch] = 4; /* initial snr offset */ s->coarse_snr_offset = 40; /* compute real values */ /* currently none of these values change during encoding, so we can just set them once at initialization */ s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift; s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift; s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code]; s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code]; s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code]; count_frame_bits_fixed(s); } /** * Count the bits used to encode the frame, minus exponents and mantissas. * Bits based on fixed parameters have already been counted, so now we just * have to add the bits based on parameters that change during encoding. */ static void count_frame_bits(AC3EncodeContext *s) { AC3EncOptions *opt = &s->options; int blk, ch; int frame_bits = 0; if (opt->audio_production_info) frame_bits += 7; if (s->bitstream_id == 6) { if (opt->extended_bsi_1) frame_bits += 14; if (opt->extended_bsi_2) frame_bits += 14; } for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { /* stereo rematrixing */ if (s->channel_mode == AC3_CHMODE_STEREO && s->blocks[blk].new_rematrixing_strategy) { frame_bits += s->num_rematrixing_bands; } for (ch = 0; ch < s->fbw_channels; ch++) { if (s->exp_strategy[ch][blk] != EXP_REUSE) frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */ } } s->frame_bits = s->frame_bits_fixed + frame_bits; } /** * Finalize the mantissa bit count by adding in the grouped mantissas. */ static int compute_mantissa_size_final(int mant_cnt[5]) { // bap=1 : 3 mantissas in 5 bits int bits = (mant_cnt[1] / 3) * 5; // bap=2 : 3 mantissas in 7 bits // bap=4 : 2 mantissas in 7 bits bits += ((mant_cnt[2] / 3) + (mant_cnt[4] >> 1)) * 7; // bap=3 : each mantissa is 3 bits bits += mant_cnt[3] * 3; return bits; } /** * Calculate masking curve based on the final exponents. * Also calculate the power spectral densities to use in future calculations. */ static void bit_alloc_masking(AC3EncodeContext *s) { int blk, ch; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; for (ch = 0; ch < s->channels; ch++) { /* We only need psd and mask for calculating bap. Since we currently do not calculate bap when exponent strategy is EXP_REUSE we do not need to calculate psd or mask. */ if (s->exp_strategy[ch][blk] != EXP_REUSE) { ff_ac3_bit_alloc_calc_psd(block->exp[ch], 0, s->nb_coefs[ch], block->psd[ch], block->band_psd[ch]); ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, block->band_psd[ch], 0, s->nb_coefs[ch], ff_ac3_fast_gain_tab[s->fast_gain_code[ch]], ch == s->lfe_channel, DBA_NONE, 0, NULL, NULL, NULL, block->mask[ch]); } } } } /** * Ensure that bap for each block and channel point to the current bap_buffer. * They may have been switched during the bit allocation search. */ static void reset_block_bap(AC3EncodeContext *s) { int blk, ch; if (s->blocks[0].bap[0] == s->bap_buffer) return; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { for (ch = 0; ch < s->channels; ch++) { s->blocks[blk].bap[ch] = &s->bap_buffer[AC3_MAX_COEFS * (blk * s->channels + ch)]; } } } /** * Run the bit allocation with a given SNR offset. * This calculates the bit allocation pointers that will be used to determine * the quantization of each mantissa. * @return the number of bits needed for mantissas if the given SNR offset is * is used. */ static int bit_alloc(AC3EncodeContext *s, int snr_offset) { int blk, ch; int mantissa_bits; int mant_cnt[5]; snr_offset = (snr_offset - 240) << 2; reset_block_bap(s); mantissa_bits = 0; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block; // initialize grouped mantissa counts. these are set so that they are // padded to the next whole group size when bits are counted in // compute_mantissa_size_final mant_cnt[0] = mant_cnt[3] = 0; mant_cnt[1] = mant_cnt[2] = 2; mant_cnt[4] = 1; for (ch = 0; ch < s->channels; ch++) { /* Currently the only bit allocation parameters which vary across blocks within a frame are the exponent values. We can take advantage of that by reusing the bit allocation pointers whenever we reuse exponents. */ block = s->blocks[blk].exp_ref_block[ch]; if (s->exp_strategy[ch][blk] != EXP_REUSE) { s->ac3dsp.bit_alloc_calc_bap(block->mask[ch], block->psd[ch], 0, s->nb_coefs[ch], snr_offset, s->bit_alloc.floor, ff_ac3_bap_tab, block->bap[ch]); } mantissa_bits += s->ac3dsp.compute_mantissa_size(mant_cnt, block->bap[ch], s->nb_coefs[ch]); } mantissa_bits += compute_mantissa_size_final(mant_cnt); } return mantissa_bits; } /** * Constant bitrate bit allocation search. * Find the largest SNR offset that will allow data to fit in the frame. */ static int cbr_bit_allocation(AC3EncodeContext *s) { int ch; int bits_left; int snr_offset, snr_incr; bits_left = 8 * s->frame_size - (s->frame_bits + s->exponent_bits); av_assert2(bits_left >= 0); snr_offset = s->coarse_snr_offset << 4; /* if previous frame SNR offset was 1023, check if current frame can also use SNR offset of 1023. if so, skip the search. */ if ((snr_offset | s->fine_snr_offset[0]) == 1023) { if (bit_alloc(s, 1023) <= bits_left) return 0; } while (snr_offset >= 0 && bit_alloc(s, snr_offset) > bits_left) { snr_offset -= 64; } if (snr_offset < 0) return AVERROR(EINVAL); FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer); for (snr_incr = 64; snr_incr > 0; snr_incr >>= 2) { while (snr_offset + snr_incr <= 1023 && bit_alloc(s, snr_offset + snr_incr) <= bits_left) { snr_offset += snr_incr; FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer); } } FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer); reset_block_bap(s); s->coarse_snr_offset = snr_offset >> 4; for (ch = 0; ch < s->channels; ch++) s->fine_snr_offset[ch] = snr_offset & 0xF; return 0; } /** * Downgrade exponent strategies to reduce the bits used by the exponents. * This is a fallback for when bit allocation fails with the normal exponent * strategies. Each time this function is run it only downgrades the * strategy in 1 channel of 1 block. * @return non-zero if downgrade was unsuccessful */ static int downgrade_exponents(AC3EncodeContext *s) { int ch, blk; for (ch = 0; ch < s->fbw_channels; ch++) { for (blk = AC3_MAX_BLOCKS-1; blk >= 0; blk--) { if (s->exp_strategy[ch][blk] == EXP_D15) { s->exp_strategy[ch][blk] = EXP_D25; return 0; } } } for (ch = 0; ch < s->fbw_channels; ch++) { for (blk = AC3_MAX_BLOCKS-1; blk >= 0; blk--) { if (s->exp_strategy[ch][blk] == EXP_D25) { s->exp_strategy[ch][blk] = EXP_D45; return 0; } } } for (ch = 0; ch < s->fbw_channels; ch++) { /* block 0 cannot reuse exponents, so only downgrade D45 to REUSE if the block number > 0 */ for (blk = AC3_MAX_BLOCKS-1; blk > 0; blk--) { if (s->exp_strategy[ch][blk] > EXP_REUSE) { s->exp_strategy[ch][blk] = EXP_REUSE; return 0; } } } return -1; } /** * Reduce the bandwidth to reduce the number of bits used for a given SNR offset. * This is a second fallback for when bit allocation still fails after exponents * have been downgraded. * @return non-zero if bandwidth reduction was unsuccessful */ static int reduce_bandwidth(AC3EncodeContext *s, int min_bw_code) { int ch; if (s->bandwidth_code[0] > min_bw_code) { for (ch = 0; ch < s->fbw_channels; ch++) { s->bandwidth_code[ch]--; s->nb_coefs[ch] = s->bandwidth_code[ch] * 3 + 73; } return 0; } return -1; } /** * Perform bit allocation search. * Finds the SNR offset value that maximizes quality and fits in the specified * frame size. Output is the SNR offset and a set of bit allocation pointers * used to quantize the mantissas. */ static int compute_bit_allocation(AC3EncodeContext *s) { int ret; count_frame_bits(s); bit_alloc_masking(s); ret = cbr_bit_allocation(s); while (ret) { /* fallback 1: downgrade exponents */ if (!downgrade_exponents(s)) { extract_exponents(s); encode_exponents(s); group_exponents(s); ret = compute_bit_allocation(s); continue; } /* fallback 2: reduce bandwidth */ /* only do this if the user has not specified a specific cutoff frequency */ if (!s->cutoff && !reduce_bandwidth(s, 0)) { process_exponents(s); ret = compute_bit_allocation(s); continue; } /* fallbacks were not enough... */ break; } return ret; } /** * Symmetric quantization on 'levels' levels. */ static inline int sym_quant(int c, int e, int levels) { int v = (((levels * c) >> (24 - e)) + levels) >> 1; av_assert2(v >= 0 && v < levels); return v; } /** * Asymmetric quantization on 2^qbits levels. */ static inline int asym_quant(int c, int e, int qbits) { int lshift, m, v; lshift = e + qbits - 24; if (lshift >= 0) v = c << lshift; else v = c >> (-lshift); /* rounding */ v = (v + 1) >> 1; m = (1 << (qbits-1)); if (v >= m) v = m - 1; av_assert2(v >= -m); return v & ((1 << qbits)-1); } /** * Quantize a set of mantissas for a single channel in a single block. */ static void quantize_mantissas_blk_ch(AC3Mant *s, int32_t *fixed_coef, uint8_t *exp, uint8_t *bap, uint16_t *qmant, int n) { int i; for (i = 0; i < n; i++) { int v; int c = fixed_coef[i]; int e = exp[i]; int b = bap[i]; switch (b) { case 0: v = 0; break; case 1: v = sym_quant(c, e, 3); switch (s->mant1_cnt) { case 0: s->qmant1_ptr = &qmant[i]; v = 9 * v; s->mant1_cnt = 1; break; case 1: *s->qmant1_ptr += 3 * v; s->mant1_cnt = 2; v = 128; break; default: *s->qmant1_ptr += v; s->mant1_cnt = 0; v = 128; break; } break; case 2: v = sym_quant(c, e, 5); switch (s->mant2_cnt) { case 0: s->qmant2_ptr = &qmant[i]; v = 25 * v; s->mant2_cnt = 1; break; case 1: *s->qmant2_ptr += 5 * v; s->mant2_cnt = 2; v = 128; break; default: *s->qmant2_ptr += v; s->mant2_cnt = 0; v = 128; break; } break; case 3: v = sym_quant(c, e, 7); break; case 4: v = sym_quant(c, e, 11); switch (s->mant4_cnt) { case 0: s->qmant4_ptr = &qmant[i]; v = 11 * v; s->mant4_cnt = 1; break; default: *s->qmant4_ptr += v; s->mant4_cnt = 0; v = 128; break; } break; case 5: v = sym_quant(c, e, 15); break; case 14: v = asym_quant(c, e, 14); break; case 15: v = asym_quant(c, e, 16); break; default: v = asym_quant(c, e, b - 1); break; } qmant[i] = v; } } /** * Quantize mantissas using coefficients, exponents, and bit allocation pointers. */ static void quantize_mantissas(AC3EncodeContext *s) { int blk, ch; for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; AC3Block *ref_block; AC3Mant m = { 0 }; for (ch = 0; ch < s->channels; ch++) { ref_block = block->exp_ref_block[ch]; quantize_mantissas_blk_ch(&m, block->fixed_coef[ch], ref_block->exp[ch], ref_block->bap[ch], block->qmant[ch], s->nb_coefs[ch]); } } } /** * Write the AC-3 frame header to the output bitstream. */ static void output_frame_header(AC3EncodeContext *s) { AC3EncOptions *opt = &s->options; put_bits(&s->pb, 16, 0x0b77); /* frame header */ put_bits(&s->pb, 16, 0); /* crc1: will be filled later */ put_bits(&s->pb, 2, s->bit_alloc.sr_code); put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2); put_bits(&s->pb, 5, s->bitstream_id); put_bits(&s->pb, 3, s->bitstream_mode); put_bits(&s->pb, 3, s->channel_mode); if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO) put_bits(&s->pb, 2, s->center_mix_level); if (s->channel_mode & 0x04) put_bits(&s->pb, 2, s->surround_mix_level); if (s->channel_mode == AC3_CHMODE_STEREO) put_bits(&s->pb, 2, opt->dolby_surround_mode); put_bits(&s->pb, 1, s->lfe_on); /* LFE */ put_bits(&s->pb, 5, -opt->dialogue_level); put_bits(&s->pb, 1, 0); /* no compression control word */ put_bits(&s->pb, 1, 0); /* no lang code */ put_bits(&s->pb, 1, opt->audio_production_info); if (opt->audio_production_info) { put_bits(&s->pb, 5, opt->mixing_level - 80); put_bits(&s->pb, 2, opt->room_type); } put_bits(&s->pb, 1, opt->copyright); put_bits(&s->pb, 1, opt->original); if (s->bitstream_id == 6) { /* alternate bit stream syntax */ put_bits(&s->pb, 1, opt->extended_bsi_1); if (opt->extended_bsi_1) { put_bits(&s->pb, 2, opt->preferred_stereo_downmix); put_bits(&s->pb, 3, s->ltrt_center_mix_level); put_bits(&s->pb, 3, s->ltrt_surround_mix_level); put_bits(&s->pb, 3, s->loro_center_mix_level); put_bits(&s->pb, 3, s->loro_surround_mix_level); } put_bits(&s->pb, 1, opt->extended_bsi_2); if (opt->extended_bsi_2) { put_bits(&s->pb, 2, opt->dolby_surround_ex_mode); put_bits(&s->pb, 2, opt->dolby_headphone_mode); put_bits(&s->pb, 1, opt->ad_converter_type); put_bits(&s->pb, 9, 0); /* xbsi2 and encinfo : reserved */ } } else { put_bits(&s->pb, 1, 0); /* no time code 1 */ put_bits(&s->pb, 1, 0); /* no time code 2 */ } put_bits(&s->pb, 1, 0); /* no additional bit stream info */ } /** * Write one audio block to the output bitstream. */ static void output_audio_block(AC3EncodeContext *s, int blk) { int ch, i, baie, rbnd; AC3Block *block = &s->blocks[blk]; /* block switching */ for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 1, 0); /* dither flags */ for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 1, 1); /* dynamic range codes */ put_bits(&s->pb, 1, 0); /* channel coupling */ if (!blk) { put_bits(&s->pb, 1, 1); /* coupling strategy present */ put_bits(&s->pb, 1, 0); /* no coupling strategy */ } else { put_bits(&s->pb, 1, 0); /* no new coupling strategy */ } /* stereo rematrixing */ if (s->channel_mode == AC3_CHMODE_STEREO) { put_bits(&s->pb, 1, block->new_rematrixing_strategy); if (block->new_rematrixing_strategy) { /* rematrixing flags */ for (rbnd = 0; rbnd < s->num_rematrixing_bands; rbnd++) put_bits(&s->pb, 1, block->rematrixing_flags[rbnd]); } } /* exponent strategy */ for (ch = 0; ch < s->fbw_channels; ch++) put_bits(&s->pb, 2, s->exp_strategy[ch][blk]); if (s->lfe_on) put_bits(&s->pb, 1, s->exp_strategy[s->lfe_channel][blk]); /* bandwidth */ for (ch = 0; ch < s->fbw_channels; ch++) { if (s->exp_strategy[ch][blk] != EXP_REUSE) put_bits(&s->pb, 6, s->bandwidth_code[ch]); } /* exponents */ for (ch = 0; ch < s->channels; ch++) { int nb_groups; if (s->exp_strategy[ch][blk] == EXP_REUSE) continue; /* DC exponent */ put_bits(&s->pb, 4, block->grouped_exp[ch][0]); /* exponent groups */ nb_groups = exponent_group_tab[s->exp_strategy[ch][blk]-1][s->nb_coefs[ch]]; for (i = 1; i <= nb_groups; i++) put_bits(&s->pb, 7, block->grouped_exp[ch][i]); /* gain range info */ if (ch != s->lfe_channel) put_bits(&s->pb, 2, 0); } /* bit allocation info */ baie = (blk == 0); put_bits(&s->pb, 1, baie); if (baie) { put_bits(&s->pb, 2, s->slow_decay_code); put_bits(&s->pb, 2, s->fast_decay_code); put_bits(&s->pb, 2, s->slow_gain_code); put_bits(&s->pb, 2, s->db_per_bit_code); put_bits(&s->pb, 3, s->floor_code); } /* snr offset */ put_bits(&s->pb, 1, baie); if (baie) { put_bits(&s->pb, 6, s->coarse_snr_offset); for (ch = 0; ch < s->channels; ch++) { put_bits(&s->pb, 4, s->fine_snr_offset[ch]); put_bits(&s->pb, 3, s->fast_gain_code[ch]); } } put_bits(&s->pb, 1, 0); /* no delta bit allocation */ put_bits(&s->pb, 1, 0); /* no data to skip */ /* mantissas */ for (ch = 0; ch < s->channels; ch++) { int b, q; AC3Block *ref_block = block->exp_ref_block[ch]; for (i = 0; i < s->nb_coefs[ch]; i++) { q = block->qmant[ch][i]; b = ref_block->bap[ch][i]; switch (b) { case 0: break; case 1: if (q != 128) put_bits(&s->pb, 5, q); break; case 2: if (q != 128) put_bits(&s->pb, 7, q); break; case 3: put_bits(&s->pb, 3, q); break; case 4: if (q != 128) put_bits(&s->pb, 7, q); break; case 14: put_bits(&s->pb, 14, q); break; case 15: put_bits(&s->pb, 16, q); break; default: put_bits(&s->pb, b-1, q); break; } } } } /** CRC-16 Polynomial */ #define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16)) static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly) { unsigned int c; c = 0; while (a) { if (a & 1) c ^= b; a = a >> 1; b = b << 1; if (b & (1 << 16)) b ^= poly; } return c; } static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly) { unsigned int r; r = 1; while (n) { if (n & 1) r = mul_poly(r, a, poly); a = mul_poly(a, a, poly); n >>= 1; } return r; } /** * Fill the end of the frame with 0's and compute the two CRCs. */ static void output_frame_end(AC3EncodeContext *s) { const AVCRC *crc_ctx = av_crc_get_table(AV_CRC_16_ANSI); int frame_size_58, pad_bytes, crc1, crc2_partial, crc2, crc_inv; uint8_t *frame; frame_size_58 = ((s->frame_size >> 2) + (s->frame_size >> 4)) << 1; /* pad the remainder of the frame with zeros */ av_assert2(s->frame_size * 8 - put_bits_count(&s->pb) >= 18); flush_put_bits(&s->pb); frame = s->pb.buf; pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2; av_assert2(pad_bytes >= 0); if (pad_bytes > 0) memset(put_bits_ptr(&s->pb), 0, pad_bytes); /* compute crc1 */ /* this is not so easy because it is at the beginning of the data... */ crc1 = av_bswap16(av_crc(crc_ctx, 0, frame + 4, frame_size_58 - 4)); crc_inv = s->crc_inv[s->frame_size > s->frame_size_min]; crc1 = mul_poly(crc_inv, crc1, CRC16_POLY); AV_WB16(frame + 2, crc1); /* compute crc2 */ crc2_partial = av_crc(crc_ctx, 0, frame + frame_size_58, s->frame_size - frame_size_58 - 3); crc2 = av_crc(crc_ctx, crc2_partial, frame + s->frame_size - 3, 1); /* ensure crc2 does not match sync word by flipping crcrsv bit if needed */ if (crc2 == 0x770B) { frame[s->frame_size - 3] ^= 0x1; crc2 = av_crc(crc_ctx, crc2_partial, frame + s->frame_size - 3, 1); } crc2 = av_bswap16(crc2); AV_WB16(frame + s->frame_size - 2, crc2); } /** * Write the frame to the output bitstream. */ static void output_frame(AC3EncodeContext *s, unsigned char *frame) { int blk; init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE); output_frame_header(s); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) output_audio_block(s, blk); output_frame_end(s); } static void dprint_options(AVCodecContext *avctx) { #ifdef DEBUG AC3EncodeContext *s = avctx->priv_data; AC3EncOptions *opt = &s->options; char strbuf[32]; switch (s->bitstream_id) { case 6: strncpy(strbuf, "AC-3 (alt syntax)", 32); break; case 8: strncpy(strbuf, "AC-3 (standard)", 32); break; case 9: strncpy(strbuf, "AC-3 (dnet half-rate)", 32); break; case 10: strncpy(strbuf, "AC-3 (dnet quater-rate", 32); break; default: snprintf(strbuf, 32, "ERROR"); } av_dlog(avctx, "bitstream_id: %s (%d)\n", strbuf, s->bitstream_id); av_dlog(avctx, "sample_fmt: %s\n", av_get_sample_fmt_name(avctx->sample_fmt)); av_get_channel_layout_string(strbuf, 32, s->channels, avctx->channel_layout); av_dlog(avctx, "channel_layout: %s\n", strbuf); av_dlog(avctx, "sample_rate: %d\n", s->sample_rate); av_dlog(avctx, "bit_rate: %d\n", s->bit_rate); if (s->cutoff) av_dlog(avctx, "cutoff: %d\n", s->cutoff); av_dlog(avctx, "per_frame_metadata: %s\n", opt->allow_per_frame_metadata?"on":"off"); if (s->has_center) av_dlog(avctx, "center_mixlev: %0.3f (%d)\n", opt->center_mix_level, s->center_mix_level); else av_dlog(avctx, "center_mixlev: {not written}\n"); if (s->has_surround) av_dlog(avctx, "surround_mixlev: %0.3f (%d)\n", opt->surround_mix_level, s->surround_mix_level); else av_dlog(avctx, "surround_mixlev: {not written}\n"); if (opt->audio_production_info) { av_dlog(avctx, "mixing_level: %ddB\n", opt->mixing_level); switch (opt->room_type) { case 0: strncpy(strbuf, "notindicated", 32); break; case 1: strncpy(strbuf, "large", 32); break; case 2: strncpy(strbuf, "small", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->room_type); } av_dlog(avctx, "room_type: %s\n", strbuf); } else { av_dlog(avctx, "mixing_level: {not written}\n"); av_dlog(avctx, "room_type: {not written}\n"); } av_dlog(avctx, "copyright: %s\n", opt->copyright?"on":"off"); av_dlog(avctx, "dialnorm: %ddB\n", opt->dialogue_level); if (s->channel_mode == AC3_CHMODE_STEREO) { switch (opt->dolby_surround_mode) { case 0: strncpy(strbuf, "notindicated", 32); break; case 1: strncpy(strbuf, "on", 32); break; case 2: strncpy(strbuf, "off", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_mode); } av_dlog(avctx, "dsur_mode: %s\n", strbuf); } else { av_dlog(avctx, "dsur_mode: {not written}\n"); } av_dlog(avctx, "original: %s\n", opt->original?"on":"off"); if (s->bitstream_id == 6) { if (opt->extended_bsi_1) { switch (opt->preferred_stereo_downmix) { case 0: strncpy(strbuf, "notindicated", 32); break; case 1: strncpy(strbuf, "ltrt", 32); break; case 2: strncpy(strbuf, "loro", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->preferred_stereo_downmix); } av_dlog(avctx, "dmix_mode: %s\n", strbuf); av_dlog(avctx, "ltrt_cmixlev: %0.3f (%d)\n", opt->ltrt_center_mix_level, s->ltrt_center_mix_level); av_dlog(avctx, "ltrt_surmixlev: %0.3f (%d)\n", opt->ltrt_surround_mix_level, s->ltrt_surround_mix_level); av_dlog(avctx, "loro_cmixlev: %0.3f (%d)\n", opt->loro_center_mix_level, s->loro_center_mix_level); av_dlog(avctx, "loro_surmixlev: %0.3f (%d)\n", opt->loro_surround_mix_level, s->loro_surround_mix_level); } else { av_dlog(avctx, "extended bitstream info 1: {not written}\n"); } if (opt->extended_bsi_2) { switch (opt->dolby_surround_ex_mode) { case 0: strncpy(strbuf, "notindicated", 32); break; case 1: strncpy(strbuf, "on", 32); break; case 2: strncpy(strbuf, "off", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_ex_mode); } av_dlog(avctx, "dsurex_mode: %s\n", strbuf); switch (opt->dolby_headphone_mode) { case 0: strncpy(strbuf, "notindicated", 32); break; case 1: strncpy(strbuf, "on", 32); break; case 2: strncpy(strbuf, "off", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_headphone_mode); } av_dlog(avctx, "dheadphone_mode: %s\n", strbuf); switch (opt->ad_converter_type) { case 0: strncpy(strbuf, "standard", 32); break; case 1: strncpy(strbuf, "hdcd", 32); break; default: snprintf(strbuf, 32, "ERROR (%d)", opt->ad_converter_type); } av_dlog(avctx, "ad_conv_type: %s\n", strbuf); } else { av_dlog(avctx, "extended bitstream info 2: {not written}\n"); } } #endif } #define FLT_OPTION_THRESHOLD 0.01 static int validate_float_option(float v, const float *v_list, int v_list_size) { int i; for (i = 0; i < v_list_size; i++) { if (v < (v_list[i] + FLT_OPTION_THRESHOLD) && v > (v_list[i] - FLT_OPTION_THRESHOLD)) break; } if (i == v_list_size) return -1; return i; } static void validate_mix_level(void *log_ctx, const char *opt_name, float *opt_param, const float *list, int list_size, int default_value, int min_value, int *ctx_param) { int mixlev = validate_float_option(*opt_param, list, list_size); if (mixlev < min_value) { mixlev = default_value; if (*opt_param >= 0.0) { av_log(log_ctx, AV_LOG_WARNING, "requested %s is not valid. using " "default value: %0.3f\n", opt_name, list[mixlev]); } } *opt_param = list[mixlev]; *ctx_param = mixlev; } /** * Validate metadata options as set by AVOption system. * These values can optionally be changed per-frame. */ static int validate_metadata(AVCodecContext *avctx) { AC3EncodeContext *s = avctx->priv_data; AC3EncOptions *opt = &s->options; /* validate mixing levels */ if (s->has_center) { validate_mix_level(avctx, "center_mix_level", &opt->center_mix_level, cmixlev_options, CMIXLEV_NUM_OPTIONS, 1, 0, &s->center_mix_level); } if (s->has_surround) { validate_mix_level(avctx, "surround_mix_level", &opt->surround_mix_level, surmixlev_options, SURMIXLEV_NUM_OPTIONS, 1, 0, &s->surround_mix_level); } /* set audio production info flag */ if (opt->mixing_level >= 0 || opt->room_type >= 0) { if (opt->mixing_level < 0) { av_log(avctx, AV_LOG_ERROR, "mixing_level must be set if " "room_type is set\n"); return AVERROR(EINVAL); } if (opt->mixing_level < 80) { av_log(avctx, AV_LOG_ERROR, "invalid mixing level. must be between " "80dB and 111dB\n"); return AVERROR(EINVAL); } /* default room type */ if (opt->room_type < 0) opt->room_type = 0; opt->audio_production_info = 1; } else { opt->audio_production_info = 0; } /* set extended bsi 1 flag */ if ((s->has_center || s->has_surround) && (opt->preferred_stereo_downmix >= 0 || opt->ltrt_center_mix_level >= 0 || opt->ltrt_surround_mix_level >= 0 || opt->loro_center_mix_level >= 0 || opt->loro_surround_mix_level >= 0)) { /* default preferred stereo downmix */ if (opt->preferred_stereo_downmix < 0) opt->preferred_stereo_downmix = 0; /* validate Lt/Rt center mix level */ validate_mix_level(avctx, "ltrt_center_mix_level", &opt->ltrt_center_mix_level, extmixlev_options, EXTMIXLEV_NUM_OPTIONS, 5, 0, &s->ltrt_center_mix_level); /* validate Lt/Rt surround mix level */ validate_mix_level(avctx, "ltrt_surround_mix_level", &opt->ltrt_surround_mix_level, extmixlev_options, EXTMIXLEV_NUM_OPTIONS, 6, 3, &s->ltrt_surround_mix_level); /* validate Lo/Ro center mix level */ validate_mix_level(avctx, "loro_center_mix_level", &opt->loro_center_mix_level, extmixlev_options, EXTMIXLEV_NUM_OPTIONS, 5, 0, &s->loro_center_mix_level); /* validate Lo/Ro surround mix level */ validate_mix_level(avctx, "loro_surround_mix_level", &opt->loro_surround_mix_level, extmixlev_options, EXTMIXLEV_NUM_OPTIONS, 6, 3, &s->loro_surround_mix_level); opt->extended_bsi_1 = 1; } else { opt->extended_bsi_1 = 0; } /* set extended bsi 2 flag */ if (opt->dolby_surround_ex_mode >= 0 || opt->dolby_headphone_mode >= 0 || opt->ad_converter_type >= 0) { /* default dolby surround ex mode */ if (opt->dolby_surround_ex_mode < 0) opt->dolby_surround_ex_mode = 0; /* default dolby headphone mode */ if (opt->dolby_headphone_mode < 0) opt->dolby_headphone_mode = 0; /* default A/D converter type */ if (opt->ad_converter_type < 0) opt->ad_converter_type = 0; opt->extended_bsi_2 = 1; } else { opt->extended_bsi_2 = 0; } /* set bitstream id for alternate bitstream syntax */ if (opt->extended_bsi_1 || opt->extended_bsi_2) { if (s->bitstream_id > 8 && s->bitstream_id < 11) { static int warn_once = 1; if (warn_once) { av_log(avctx, AV_LOG_WARNING, "alternate bitstream syntax is " "not compatible with reduced samplerates. writing of " "extended bitstream information will be disabled.\n"); warn_once = 0; } } else { s->bitstream_id = 6; } } return 0; } /** * Encode a single AC-3 frame. */ static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { AC3EncodeContext *s = avctx->priv_data; const SampleType *samples = data; int ret; if (s->options.allow_per_frame_metadata) { ret = validate_metadata(avctx); if (ret) return ret; } if (s->bit_alloc.sr_code == 1) adjust_frame_size(s); deinterleave_input_samples(s, samples); apply_mdct(s); scale_coefficients(s); compute_rematrixing_strategy(s); apply_rematrixing(s); process_exponents(s); ret = compute_bit_allocation(s); if (ret) { av_log(avctx, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n"); return ret; } quantize_mantissas(s); output_frame(s, frame); return s->frame_size; } /** * Finalize encoding and free any memory allocated by the encoder. */ static av_cold int ac3_encode_close(AVCodecContext *avctx) { int blk, ch; AC3EncodeContext *s = avctx->priv_data; for (ch = 0; ch < s->channels; ch++) av_freep(&s->planar_samples[ch]); av_freep(&s->planar_samples); av_freep(&s->bap_buffer); av_freep(&s->bap1_buffer); av_freep(&s->mdct_coef_buffer); av_freep(&s->fixed_coef_buffer); av_freep(&s->exp_buffer); av_freep(&s->grouped_exp_buffer); av_freep(&s->psd_buffer); av_freep(&s->band_psd_buffer); av_freep(&s->mask_buffer); av_freep(&s->qmant_buffer); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; av_freep(&block->bap); av_freep(&block->mdct_coef); av_freep(&block->fixed_coef); av_freep(&block->exp); av_freep(&block->grouped_exp); av_freep(&block->psd); av_freep(&block->band_psd); av_freep(&block->mask); av_freep(&block->qmant); } mdct_end(&s->mdct); av_freep(&avctx->coded_frame); return 0; } /** * Set channel information during initialization. */ static av_cold int set_channel_info(AC3EncodeContext *s, int channels, int64_t *channel_layout) { int ch_layout; if (channels < 1 || channels > AC3_MAX_CHANNELS) return AVERROR(EINVAL); if ((uint64_t)*channel_layout > 0x7FF) return AVERROR(EINVAL); ch_layout = *channel_layout; if (!ch_layout) ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL); if (av_get_channel_layout_nb_channels(ch_layout) != channels) return AVERROR(EINVAL); s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY); s->channels = channels; s->fbw_channels = channels - s->lfe_on; s->lfe_channel = s->lfe_on ? s->fbw_channels : -1; if (s->lfe_on) ch_layout -= AV_CH_LOW_FREQUENCY; switch (ch_layout) { case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break; case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break; case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break; case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break; case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break; case AV_CH_LAYOUT_QUAD: case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break; case AV_CH_LAYOUT_5POINT0: case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break; default: return AVERROR(EINVAL); } s->has_center = (s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO; s->has_surround = s->channel_mode & 0x04; s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on]; *channel_layout = ch_layout; if (s->lfe_on) *channel_layout |= AV_CH_LOW_FREQUENCY; return 0; } static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s) { int i, ret; /* validate channel layout */ if (!avctx->channel_layout) { av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The " "encoder will guess the layout, but it " "might be incorrect.\n"); } ret = set_channel_info(s, avctx->channels, &avctx->channel_layout); if (ret) { av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n"); return ret; } /* validate sample rate */ for (i = 0; i < 9; i++) { if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate) break; } if (i == 9) { av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n"); return AVERROR(EINVAL); } s->sample_rate = avctx->sample_rate; s->bit_alloc.sr_shift = i % 3; s->bit_alloc.sr_code = i / 3; s->bitstream_id = 8 + s->bit_alloc.sr_shift; /* validate bit rate */ for (i = 0; i < 19; i++) { if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate) break; } if (i == 19) { av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n"); return AVERROR(EINVAL); } s->bit_rate = avctx->bit_rate; s->frame_size_code = i << 1; /* validate cutoff */ if (avctx->cutoff < 0) { av_log(avctx, AV_LOG_ERROR, "invalid cutoff frequency\n"); return AVERROR(EINVAL); } s->cutoff = avctx->cutoff; if (s->cutoff > (s->sample_rate >> 1)) s->cutoff = s->sample_rate >> 1; /* validate audio service type / channels combination */ if ((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_KARAOKE && avctx->channels == 1) || ((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_COMMENTARY || avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_EMERGENCY || avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_VOICE_OVER) && avctx->channels > 1)) { av_log(avctx, AV_LOG_ERROR, "invalid audio service type for the " "specified number of channels\n"); return AVERROR(EINVAL); } ret = validate_metadata(avctx); if (ret) return ret; return 0; } /** * Set bandwidth for all channels. * The user can optionally supply a cutoff frequency. Otherwise an appropriate * default value will be used. */ static av_cold void set_bandwidth(AC3EncodeContext *s) { int ch, bw_code; if (s->cutoff) { /* calculate bandwidth based on user-specified cutoff frequency */ int fbw_coeffs; fbw_coeffs = s->cutoff * 2 * AC3_MAX_COEFS / s->sample_rate; bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60); } else { /* use default bandwidth setting */ bw_code = ac3_bandwidth_tab[s->fbw_channels-1][s->bit_alloc.sr_code][s->frame_size_code/2]; } /* set number of coefficients for each channel */ for (ch = 0; ch < s->fbw_channels; ch++) { s->bandwidth_code[ch] = bw_code; s->nb_coefs[ch] = bw_code * 3 + 73; } if (s->lfe_on) s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */ } static av_cold int allocate_buffers(AVCodecContext *avctx) { int blk, ch; AC3EncodeContext *s = avctx->priv_data; FF_ALLOC_OR_GOTO(avctx, s->planar_samples, s->channels * sizeof(*s->planar_samples), alloc_fail); for (ch = 0; ch < s->channels; ch++) { FF_ALLOCZ_OR_GOTO(avctx, s->planar_samples[ch], (AC3_FRAME_SIZE+AC3_BLOCK_SIZE) * sizeof(**s->planar_samples), alloc_fail); } FF_ALLOC_OR_GOTO(avctx, s->bap_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->bap_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->bap1_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->bap1_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->mdct_coef_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->mdct_coef_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->exp_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->exp_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->grouped_exp_buffer, AC3_MAX_BLOCKS * s->channels * 128 * sizeof(*s->grouped_exp_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->psd_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->psd_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->band_psd_buffer, AC3_MAX_BLOCKS * s->channels * 64 * sizeof(*s->band_psd_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->mask_buffer, AC3_MAX_BLOCKS * s->channels * 64 * sizeof(*s->mask_buffer), alloc_fail); FF_ALLOC_OR_GOTO(avctx, s->qmant_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->qmant_buffer), alloc_fail); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; FF_ALLOC_OR_GOTO(avctx, block->bap, s->channels * sizeof(*block->bap), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->mdct_coef, s->channels * sizeof(*block->mdct_coef), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->exp, s->channels * sizeof(*block->exp), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->grouped_exp, s->channels * sizeof(*block->grouped_exp), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->psd, s->channels * sizeof(*block->psd), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->band_psd, s->channels * sizeof(*block->band_psd), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->mask, s->channels * sizeof(*block->mask), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, block->qmant, s->channels * sizeof(*block->qmant), alloc_fail); for (ch = 0; ch < s->channels; ch++) { /* arrangement: block, channel, coeff */ block->bap[ch] = &s->bap_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)]; block->mdct_coef[ch] = &s->mdct_coef_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)]; block->grouped_exp[ch] = &s->grouped_exp_buffer[128 * (blk * s->channels + ch)]; block->psd[ch] = &s->psd_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)]; block->band_psd[ch] = &s->band_psd_buffer [64 * (blk * s->channels + ch)]; block->mask[ch] = &s->mask_buffer [64 * (blk * s->channels + ch)]; block->qmant[ch] = &s->qmant_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)]; /* arrangement: channel, block, coeff */ block->exp[ch] = &s->exp_buffer [AC3_MAX_COEFS * (AC3_MAX_BLOCKS * ch + blk)]; } } if (CONFIG_AC3ENC_FLOAT) { FF_ALLOC_OR_GOTO(avctx, s->fixed_coef_buffer, AC3_MAX_BLOCKS * s->channels * AC3_MAX_COEFS * sizeof(*s->fixed_coef_buffer), alloc_fail); for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; FF_ALLOCZ_OR_GOTO(avctx, block->fixed_coef, s->channels * sizeof(*block->fixed_coef), alloc_fail); for (ch = 0; ch < s->channels; ch++) block->fixed_coef[ch] = &s->fixed_coef_buffer[AC3_MAX_COEFS * (blk * s->channels + ch)]; } } else { for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) { AC3Block *block = &s->blocks[blk]; FF_ALLOCZ_OR_GOTO(avctx, block->fixed_coef, s->channels * sizeof(*block->fixed_coef), alloc_fail); for (ch = 0; ch < s->channels; ch++) block->fixed_coef[ch] = (int32_t *)block->mdct_coef[ch]; } } return 0; alloc_fail: return AVERROR(ENOMEM); } /** * Initialize the encoder. */ static av_cold int ac3_encode_init(AVCodecContext *avctx) { AC3EncodeContext *s = avctx->priv_data; int ret, frame_size_58; avctx->frame_size = AC3_FRAME_SIZE; ff_ac3_common_init(); ret = validate_options(avctx, s); if (ret) return ret; s->bitstream_mode = avctx->audio_service_type; if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE) s->bitstream_mode = 0x7; s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code]; s->bits_written = 0; s->samples_written = 0; s->frame_size = s->frame_size_min; /* calculate crc_inv for both possible frame sizes */ frame_size_58 = (( s->frame_size >> 2) + ( s->frame_size >> 4)) << 1; s->crc_inv[0] = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY); if (s->bit_alloc.sr_code == 1) { frame_size_58 = (((s->frame_size+2) >> 2) + ((s->frame_size+2) >> 4)) << 1; s->crc_inv[1] = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY); } set_bandwidth(s); rematrixing_init(s); exponent_init(s); bit_alloc_init(s); ret = mdct_init(avctx, &s->mdct, 9); if (ret) goto init_fail; ret = allocate_buffers(avctx); if (ret) goto init_fail; avctx->coded_frame= avcodec_alloc_frame(); dsputil_init(&s->dsp, avctx); ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT); dprint_options(avctx); return 0; init_fail: ac3_encode_close(avctx); return ret; }