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FFmpeg/libavcodec/ac3enc.c
Michael Niedermayer 0665199e43 Merge remote branch 'qatar/master' 
* qatar/master:
  vorbisdec: Rename silly "class_" variable to plain "class".
  simple_idct_alpha: Drop some useless casts.
  Simplify av_log_missing_feature().
  ac3enc: remove check for mismatching channels and channel_layout
  If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout.
  If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother.
  cosmetics: indentation
  Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open().
  aacdec: remove sf_scale and sf_offset.
  aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec.
  Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere.
  Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway.
  qpeg: use reget_buffer() in decode_frame()
  ultimotion: use reget_buffer() in ulti_decode_frame()
  smacker: remove unnecessary call to avctx->release_buffer in decode_frame()
  avparser: don't av_malloc(0).

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-28 04:26:01 +02:00

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/*
* The simplest AC-3 encoder
* Copyright (c) 2000 Fabrice Bellard
* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* The simplest AC-3 encoder.
*/
//#define DEBUG
//#define ASSERT_LEVEL 2
#include <stdint.h>
#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;
/**
* 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(32, 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)
#if CONFIG_AC3ENC_FLOAT || !CONFIG_AC3_FLOAT_ENCODER //we need this exactly once compiled in
const AVOption ff_ac3_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}
};
#endif
#if CONFIG_AC3ENC_FLOAT
static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
ff_ac3_options, LIBAVUTIL_VERSION_INT };
#else
static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
ff_ac3_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.
*/
#if CONFIG_AC3ENC_FLOAT || !CONFIG_AC3_FLOAT_ENCODER //we need this exactly once compiled in
const int64_t ff_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
};
#endif
/**
* 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 FFmpeg'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);
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;
}
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