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FFmpeg/libavcodec/dcaenc.c

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/*
* DCA encoder
* Copyright (C) 2008 Alexander E. Patrakov
* 2010 Benjamin Larsson
* 2011 Xiang Wang
*
* 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
*/
#include "libavutil/common.h"
#include "libavutil/avassert.h"
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "put_bits.h"
#include "dcaenc.h"
#include "dcadata.h"
#undef NDEBUG
#define MAX_CHANNELS 6
#define DCA_SUBBANDS_32 32
#define DCA_MAX_FRAME_SIZE 16383
#define DCA_HEADER_SIZE 13
#define DCA_SUBBANDS 32 ///< Subband activity count
#define QUANTIZER_BITS 16
#define SUBFRAMES 1
#define SUBSUBFRAMES 4
#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
#define LFE_BITS 8
#define LFE_INTERPOLATION 64
#define LFE_PRESENT 2
#define LFE_MISSING 0
static const int8_t dca_lfe_index[] = {
1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
};
static const int8_t dca_channel_reorder_lfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, 2, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, -1, -1, -1, -1 },
{ 0, 1, -1, 2, 3, -1, -1, -1, -1 },
{ 1, 2, 0, -1, 3, 4, -1, -1, -1 },
{ 2, 3, -1, 0, 1, 4, 5, -1, -1 },
{ 1, 2, 0, -1, 3, 4, 5, -1, -1 },
{ 0, -1, 4, 5, 2, 3, 1, -1, -1 },
{ 3, 4, 1, -1, 0, 2, 5, 6, -1 },
{ 2, 3, -1, 5, 7, 0, 1, 4, 6 },
{ 3, 4, 1, -1, 0, 2, 5, 7, 6 },
};
static const int8_t dca_channel_reorder_nolfe[][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
{ 0, 1, 2, -1, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, -1, -1, -1, -1, -1 },
{ 0, 1, 2, 3, -1, -1, -1, -1, -1 },
{ 1, 2, 0, 3, 4, -1, -1, -1, -1 },
{ 2, 3, 0, 1, 4, 5, -1, -1, -1 },
{ 1, 2, 0, 3, 4, 5, -1, -1, -1 },
{ 0, 4, 5, 2, 3, 1, -1, -1, -1 },
{ 3, 4, 1, 0, 2, 5, 6, -1, -1 },
{ 2, 3, 5, 7, 0, 1, 4, 6, -1 },
{ 3, 4, 1, 0, 2, 5, 7, 6, -1 },
};
typedef struct {
PutBitContext pb;
int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
int start[MAX_CHANNELS];
int frame_size;
int prim_channels;
int lfe_channel;
int sample_rate_code;
int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
int lfe_scale_factor;
int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
int a_mode; ///< audio channels arrangement
int num_channel;
int lfe_state;
int lfe_offset;
const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
} DCAContext;
static int32_t cos_table[128];
static inline int32_t mul32(int32_t a, int32_t b)
{
int64_t r = (int64_t) a * b;
/* round the result before truncating - improves accuracy */
return (r + 0x80000000) >> 32;
}
/* Integer version of the cosine modulated Pseudo QMF */
static void qmf_init(void)
{
int i;
int32_t c[17], s[17];
s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
for (i = 1; i <= 16; i++) {
s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
}
for (i = 0; i < 16; i++) {
cos_table[i ] = c[i] >> 3; /* avoid output overflow */
cos_table[i + 16] = s[16 - i] >> 3;
cos_table[i + 32] = -s[i] >> 3;
cos_table[i + 48] = -c[16 - i] >> 3;
cos_table[i + 64] = -c[i] >> 3;
cos_table[i + 80] = -s[16 - i] >> 3;
cos_table[i + 96] = s[i] >> 3;
cos_table[i + 112] = c[16 - i] >> 3;
}
}
static int32_t band_delta_factor(int band, int sample_num)
{
int index = band * (2 * sample_num + 1);
if (band == 0)
return 0x07ffffff;
else
return cos_table[index & 127];
}
static void add_new_samples(DCAContext *c, const int32_t *in,
int count, int channel)
{
int i;
/* Place new samples into the history buffer */
for (i = 0; i < count; i++) {
c->history[channel][c->start[channel] + i] = in[i];
av_assert0(c->start[channel] + i < 512);
}
c->start[channel] += count;
if (c->start[channel] == 512)
c->start[channel] = 0;
av_assert0(c->start[channel] < 512);
}
static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
int channel)
{
int band, i, j, k;
int32_t resp;
int32_t accum[DCA_SUBBANDS_32] = {0};
add_new_samples(c, in, DCA_SUBBANDS_32, channel);
/* Calculate the dot product of the signal with the (possibly inverted)
reference decoder's response to this vector:
(0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
so that -1.0 cancels 1.0 from the previous step */
for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
for (i = 0; i < c->start[channel]; k++, j++, i++)
accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
resp = 0;
/* TODO: implement FFT instead of this naive calculation */
for (band = 0; band < DCA_SUBBANDS_32; band++) {
for (j = 0; j < 32; j++)
resp += mul32(accum[j], band_delta_factor(band, j));
out[band] = (band & 2) ? (-resp) : resp;
}
}
static int32_t lfe_fir_64i[512];
static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
{
int i, j;
int channel = c->prim_channels;
int32_t accum = 0;
add_new_samples(c, in, LFE_INTERPOLATION, channel);
for (i = c->start[channel], j = 0; i < 512; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
for (i = 0; i < c->start[channel]; i++, j++)
accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
return accum;
}
static void init_lfe_fir(void)
{
static int initialized = 0;
int i;
if (initialized)
return;
for (i = 0; i < 512; i++)
lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
initialized = 1;
}
static void put_frame_header(DCAContext *c)
{
/* SYNC */
put_bits(&c->pb, 16, 0x7ffe);
put_bits(&c->pb, 16, 0x8001);
/* Frame type: normal */
put_bits(&c->pb, 1, 1);
/* Deficit sample count: none */
put_bits(&c->pb, 5, 31);
/* CRC is not present */
put_bits(&c->pb, 1, 0);
/* Number of PCM sample blocks */
put_bits(&c->pb, 7, PCM_SAMPLES-1);
/* Primary frame byte size */
put_bits(&c->pb, 14, c->frame_size-1);
/* Audio channel arrangement: L + R (stereo) */
put_bits(&c->pb, 6, c->num_channel);
/* Core audio sampling frequency */
put_bits(&c->pb, 4, c->sample_rate_code);
/* Transmission bit rate: 1411.2 kbps */
put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
/* Embedded down mix: disabled */
put_bits(&c->pb, 1, 0);
/* Embedded dynamic range flag: not present */
put_bits(&c->pb, 1, 0);
/* Embedded time stamp flag: not present */
put_bits(&c->pb, 1, 0);
/* Auxiliary data flag: not present */
put_bits(&c->pb, 1, 0);
/* HDCD source: no */
put_bits(&c->pb, 1, 0);
/* Extension audio ID: N/A */
put_bits(&c->pb, 3, 0);
/* Extended audio data: not present */
put_bits(&c->pb, 1, 0);
/* Audio sync word insertion flag: after each sub-frame */
put_bits(&c->pb, 1, 0);
/* Low frequency effects flag: not present or interpolation factor=64 */
put_bits(&c->pb, 2, c->lfe_state);
/* Predictor history switch flag: on */
put_bits(&c->pb, 1, 1);
/* No CRC */
/* Multirate interpolator switch: non-perfect reconstruction */
put_bits(&c->pb, 1, 0);
/* Encoder software revision: 7 */
put_bits(&c->pb, 4, 7);
/* Copy history: 0 */
put_bits(&c->pb, 2, 0);
/* Source PCM resolution: 16 bits, not DTS ES */
put_bits(&c->pb, 3, 0);
/* Front sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Surrounds sum/difference coding: no */
put_bits(&c->pb, 1, 0);
/* Dialog normalization: 0 dB */
put_bits(&c->pb, 4, 0);
}
static void put_primary_audio_header(DCAContext *c)
{
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
int ch, i;
/* Number of subframes */
put_bits(&c->pb, 4, SUBFRAMES - 1);
/* Number of primary audio channels */
put_bits(&c->pb, 3, c->prim_channels - 1);
/* Subband activity count */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
/* High frequency VQ start subband */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
/* Joint intensity coding index: 0, 0 */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 3, 0);
/* Transient mode codebook: A4, A4 (arbitrary) */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 2, 0);
/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Bit allocation quantizer select: linear 5-bit */
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, 3, 6);
/* Quantization index codebook select: dummy data
to avoid transmission of scale factor adjustment */
for (i = 1; i < 11; i++)
for (ch = 0; ch < c->prim_channels; ch++)
put_bits(&c->pb, bitlen[i], thr[i]);
/* Scale factor adjustment index: not transmitted */
}
/**
* 8-23 bits quantization
* @param sample
* @param bits
*/
static inline uint32_t quantize(int32_t sample, int bits)
{
av_assert0(sample < 1 << (bits - 1));
av_assert0(sample >= -(1 << (bits - 1)));
return sample & ((1 << bits) - 1);
}
static inline int find_scale_factor7(int64_t max_value, int bits)
{
int i = 0, j = 128, q;
max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
while (i < j) {
q = (i + j) >> 1;
if (max_value < scale_factor_quant7[q])
j = q;
else
i = q + 1;
}
av_assert1(i < 128);
return i;
}
static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
int scale_factor)
{
sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
put_bits(&c->pb, bits, quantize((int) sample, bits));
}
static void put_subframe(DCAContext *c,
int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
int subframe)
{
int i, sub, ss, ch, max_value;
int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
/* Subsubframes count */
put_bits(&c->pb, 2, SUBSUBFRAMES -1);
/* Partial subsubframe sample count: dummy */
put_bits(&c->pb, 3, 0);
/* Prediction mode: no ADPCM, in each channel and subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 1, 0);
/* Prediction VQ addres: not transmitted */
/* Bit allocation index */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 5, QUANTIZER_BITS+3);
if (SUBSUBFRAMES > 1) {
/* Transition mode: none for each channel and subband */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 1, 0); /* codebook A4 */
}
/* Determine scale_factor */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++) {
max_value = 0;
for (i = 0; i < 8 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
}
if (c->lfe_channel) {
max_value = 0;
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
max_value = FFMAX(max_value, FFABS(lfe_data[i]));
c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
}
/* Scale factors: the same for each channel and subband,
encoded according to Table D.1.2 */
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
/* Joint subband scale factor codebook select: not transmitted */
/* Scale factors for joint subband coding: not transmitted */
/* Stereo down-mix coefficients: not transmitted */
/* Dynamic range coefficient: not transmitted */
/* Stde information CRC check word: not transmitted */
/* VQ encoded high frequency subbands: not transmitted */
/* LFE data */
if (c->lfe_channel) {
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
put_bits(&c->pb, 8, c->lfe_scale_factor);
}
/* Audio data (subsubframes) */
for (ss = 0; ss < SUBSUBFRAMES ; ss++)
for (ch = 0; ch < c->prim_channels; ch++)
for (sub = 0; sub < DCA_SUBBANDS; sub++)
for (i = 0; i < 8; i++)
put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
/* DSYNC */
put_bits(&c->pb, 16, 0xffff);
}
static void put_frame(DCAContext *c,
int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
uint8_t *frame)
{
int i;
init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
put_primary_audio_header(c);
for (i = 0; i < SUBFRAMES; i++)
put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
flush_put_bits(&c->pb);
c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
put_frame_header(c);
flush_put_bits(&c->pb);
}
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int i, k, channel;
DCAContext *c = avctx->priv_data;
const int16_t *samples;
int ret, real_channel = 0;
if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
return ret;
samples = (const int16_t *)frame->data[0];
for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
for (channel = 0; channel < c->prim_channels + 1; channel++) {
/* Get 32 PCM samples */
for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
}
/* Put subband samples into the proper place */
real_channel = c->channel_order_tab[channel];
if (real_channel >= 0) {
qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
}
}
}
if (c->lfe_channel) {
for (i = 0; i < PCM_SAMPLES / 2; i++) {
for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
c->lfe_data[i] = lfe_downsample(c, c->pcm);
}
}
put_frame(c, c->subband, avpkt->data);
avpkt->size = c->frame_size;
*got_packet_ptr = 1;
return 0;
}
static int encode_init(AVCodecContext *avctx)
{
DCAContext *c = avctx->priv_data;
int i;
uint64_t layout = avctx->channel_layout;
c->prim_channels = avctx->channels;
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
if (!layout) {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
"encoder will guess the layout, but it "
"might be incorrect.\n");
layout = av_get_default_channel_layout(avctx->channels);
}
switch (layout) {
case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
default:
av_log(avctx, AV_LOG_ERROR,
"Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
return AVERROR_PATCHWELCOME;
}
if (c->lfe_channel) {
init_lfe_fir();
c->prim_channels--;
c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
c->lfe_state = LFE_PRESENT;
c->lfe_offset = dca_lfe_index[c->a_mode];
} else {
c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
c->lfe_state = LFE_MISSING;
}
for (i = 0; i < 16; i++) {
if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
break;
}
if (i == 16) {
av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
for (i = 0; i < 16; i++)
av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
av_log(avctx, AV_LOG_ERROR, "supported.\n");
return -1;
}
c->sample_rate_code = i;
avctx->frame_size = 32 * PCM_SAMPLES;
if (!cos_table[127])
qmf_init();
return 0;
}
AVCodec ff_dca_encoder = {
.name = "dca",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = encode_init,
.encode2 = encode_frame,
.capabilities = CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
};