mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-22 09:15:06 +02:00
602 lines
19 KiB
C
602 lines
19 KiB
C
/*
|
|
* 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++) {
|
|
real_channel = c->channel_order_tab[channel];
|
|
if (real_channel >= 0) {
|
|
/* 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 */
|
|
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)"),
|
|
};
|