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FFmpeg/libavcodec/dcadec.c

418 lines
13 KiB
C

/*
* Copyright (C) 2016 foo86
*
* 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/opt.h"
#include "libavutil/channel_layout.h"
#include "dcadec.h"
#include "dcamath.h"
#include "dca_syncwords.h"
#include "profiles.h"
#define MIN_PACKET_SIZE 16
#define MAX_PACKET_SIZE 0x104000
int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask)
{
static const uint8_t dca2wav_norm[28] = {
2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12,
13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5,
};
static const uint8_t dca2wav_wide[28] = {
2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12,
13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5,
};
int dca_ch, wav_ch, nchannels = 0;
if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) {
for (dca_ch = 0; dca_ch < DCA_SPEAKER_COUNT; dca_ch++)
if (dca_mask & (1U << dca_ch))
ch_remap[nchannels++] = dca_ch;
avctx->channel_layout = dca_mask;
} else {
int wav_mask = 0;
int wav_map[18];
const uint8_t *dca2wav;
if (dca_mask == DCA_SPEAKER_LAYOUT_7POINT0_WIDE ||
dca_mask == DCA_SPEAKER_LAYOUT_7POINT1_WIDE)
dca2wav = dca2wav_wide;
else
dca2wav = dca2wav_norm;
for (dca_ch = 0; dca_ch < 28; dca_ch++) {
if (dca_mask & (1 << dca_ch)) {
wav_ch = dca2wav[dca_ch];
if (!(wav_mask & (1 << wav_ch))) {
wav_map[wav_ch] = dca_ch;
wav_mask |= 1 << wav_ch;
}
}
}
for (wav_ch = 0; wav_ch < 18; wav_ch++)
if (wav_mask & (1 << wav_ch))
ch_remap[nchannels++] = wav_map[wav_ch];
avctx->channel_layout = wav_mask;
}
avctx->channels = nchannels;
return nchannels;
}
static uint16_t crc16(const uint8_t *data, int size)
{
static const uint16_t crctab[16] = {
0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7,
0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef,
};
uint16_t res = 0xffff;
int i;
for (i = 0; i < size; i++) {
res = (res << 4) ^ crctab[(data[i] >> 4) ^ (res >> 12)];
res = (res << 4) ^ crctab[(data[i] & 15) ^ (res >> 12)];
}
return res;
}
int ff_dca_check_crc(GetBitContext *s, int p1, int p2)
{
if (((p1 | p2) & 7) || p1 < 0 || p2 > s->size_in_bits || p2 - p1 < 16)
return -1;
if (crc16(s->buffer + p1 / 8, (p2 - p1) / 8))
return -1;
return 0;
}
void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples,
int *coeff_l, int nsamples, int ch_mask)
{
int pos, spkr, max_spkr = av_log2(ch_mask);
int *coeff_r = coeff_l + av_popcount(ch_mask);
av_assert0(DCA_HAS_STEREO(ch_mask));
// Scale left and right channels
pos = (ch_mask & DCA_SPEAKER_MASK_C);
dcadsp->dmix_scale(samples[DCA_SPEAKER_L], coeff_l[pos ], nsamples);
dcadsp->dmix_scale(samples[DCA_SPEAKER_R], coeff_r[pos + 1], nsamples);
// Downmix remaining channels
for (spkr = 0; spkr <= max_spkr; spkr++) {
if (!(ch_mask & (1U << spkr)))
continue;
if (*coeff_l && spkr != DCA_SPEAKER_L)
dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr],
*coeff_l, nsamples);
if (*coeff_r && spkr != DCA_SPEAKER_R)
dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr],
*coeff_r, nsamples);
coeff_l++;
coeff_r++;
}
}
void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples,
int *coeff_l, int nsamples, int ch_mask)
{
int pos, spkr, max_spkr = av_log2(ch_mask);
int *coeff_r = coeff_l + av_popcount(ch_mask);
const float scale = 1.0f / (1 << 15);
av_assert0(DCA_HAS_STEREO(ch_mask));
// Scale left and right channels
pos = (ch_mask & DCA_SPEAKER_MASK_C);
fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_L], samples[DCA_SPEAKER_L],
coeff_l[pos ] * scale, nsamples);
fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_R], samples[DCA_SPEAKER_R],
coeff_r[pos + 1] * scale, nsamples);
// Downmix remaining channels
for (spkr = 0; spkr <= max_spkr; spkr++) {
if (!(ch_mask & (1U << spkr)))
continue;
if (*coeff_l && spkr != DCA_SPEAKER_L)
fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr],
*coeff_l * scale, nsamples);
if (*coeff_r && spkr != DCA_SPEAKER_R)
fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr],
*coeff_r * scale, nsamples);
coeff_l++;
coeff_r++;
}
}
static int convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size)
{
switch (AV_RB32(src)) {
case DCA_SYNCWORD_CORE_BE:
case DCA_SYNCWORD_SUBSTREAM:
memcpy(dst, src, src_size);
return src_size;
case DCA_SYNCWORD_CORE_LE:
case DCA_SYNCWORD_CORE_14B_BE:
case DCA_SYNCWORD_CORE_14B_LE:
return avpriv_dca_convert_bitstream(src, src_size, dst, max_size);
default:
return AVERROR_INVALIDDATA;
}
}
static int dcadec_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
DCAContext *s = avctx->priv_data;
AVFrame *frame = data;
uint8_t *input = avpkt->data;
int input_size = avpkt->size;
int i, ret, prev_packet = s->packet;
if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) {
av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n");
return AVERROR_INVALIDDATA;
}
av_fast_malloc(&s->buffer, &s->buffer_size,
FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_SIZE);
if (!s->buffer)
return AVERROR(ENOMEM);
for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++)
ret = convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size);
if (ret < 0)
return ret;
input = s->buffer;
input_size = ret;
s->packet = 0;
// Parse backward compatible core sub-stream
if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) {
int frame_size;
if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0) {
s->core_residual_valid = 0;
return ret;
}
s->packet |= DCA_PACKET_CORE;
// EXXS data must be aligned on 4-byte boundary
frame_size = FFALIGN(s->core.frame_size, 4);
if (input_size - 4 > frame_size) {
input += frame_size;
input_size -= frame_size;
}
}
if (!s->core_only) {
DCAExssAsset *asset = NULL;
// Parse extension sub-stream (EXSS)
if (AV_RB32(input) == DCA_SYNCWORD_SUBSTREAM) {
if ((ret = ff_dca_exss_parse(&s->exss, input, input_size)) < 0) {
if (avctx->err_recognition & AV_EF_EXPLODE)
return ret;
} else {
s->packet |= DCA_PACKET_EXSS;
asset = &s->exss.assets[0];
}
}
// Parse XLL component in EXSS
if (asset && (asset->extension_mask & DCA_EXSS_XLL)) {
if ((ret = ff_dca_xll_parse(&s->xll, input, asset)) < 0) {
// Conceal XLL synchronization error
if (ret == AVERROR(EAGAIN)
&& (prev_packet & DCA_PACKET_XLL)
&& (s->packet & DCA_PACKET_CORE))
s->packet |= DCA_PACKET_XLL | DCA_PACKET_RECOVERY;
else if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
return ret;
} else {
s->packet |= DCA_PACKET_XLL;
}
}
// Parse core extensions in EXSS or backward compatible core sub-stream
if ((s->packet & DCA_PACKET_CORE)
&& (ret = ff_dca_core_parse_exss(&s->core, input, asset)) < 0)
return ret;
}
// Filter the frame
if (s->packet & DCA_PACKET_XLL) {
if (s->packet & DCA_PACKET_CORE) {
int x96_synth = -1;
// Enable X96 synthesis if needed
if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000)
x96_synth = 1;
if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) {
s->core_residual_valid = 0;
return ret;
}
// Force lossy downmixed output on the first core frame filtered.
// This prevents audible clicks when seeking and is consistent with
// what reference decoder does when there are multiple channel sets.
if (!s->core_residual_valid) {
if (s->xll.nreschsets > 0 && s->xll.nchsets > 1)
s->packet |= DCA_PACKET_RECOVERY;
s->core_residual_valid = 1;
}
}
if ((ret = ff_dca_xll_filter_frame(&s->xll, frame)) < 0) {
// Fall back to core unless hard error
if (!(s->packet & DCA_PACKET_CORE))
return ret;
if (ret != AVERROR_INVALIDDATA || (avctx->err_recognition & AV_EF_EXPLODE))
return ret;
if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) {
s->core_residual_valid = 0;
return ret;
}
}
} else if (s->packet & DCA_PACKET_CORE) {
if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) {
s->core_residual_valid = 0;
return ret;
}
s->core_residual_valid = !!(s->core.filter_mode & DCA_FILTER_MODE_FIXED);
} else {
return AVERROR_INVALIDDATA;
}
*got_frame_ptr = 1;
return avpkt->size;
}
static av_cold void dcadec_flush(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_dca_core_flush(&s->core);
ff_dca_xll_flush(&s->xll);
s->core_residual_valid = 0;
}
static av_cold int dcadec_close(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
ff_dca_core_close(&s->core);
ff_dca_xll_close(&s->xll);
av_freep(&s->buffer);
s->buffer_size = 0;
return 0;
}
static av_cold int dcadec_init(AVCodecContext *avctx)
{
DCAContext *s = avctx->priv_data;
s->avctx = avctx;
s->core.avctx = avctx;
s->exss.avctx = avctx;
s->xll.avctx = avctx;
if (ff_dca_core_init(&s->core) < 0)
return AVERROR(ENOMEM);
ff_dcadsp_init(&s->dcadsp);
s->core.dcadsp = &s->dcadsp;
s->xll.dcadsp = &s->dcadsp;
switch (avctx->request_channel_layout & ~AV_CH_LAYOUT_NATIVE) {
case 0:
s->request_channel_layout = 0;
break;
case AV_CH_LAYOUT_STEREO:
case AV_CH_LAYOUT_STEREO_DOWNMIX:
s->request_channel_layout = DCA_SPEAKER_LAYOUT_STEREO;
break;
case AV_CH_LAYOUT_5POINT0:
s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT0;
break;
case AV_CH_LAYOUT_5POINT1:
s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT1;
break;
default:
av_log(avctx, AV_LOG_WARNING, "Invalid request_channel_layout\n");
break;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
avctx->bits_per_raw_sample = 24;
return 0;
}
#define OFFSET(x) offsetof(DCAContext, x)
#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption dcadec_options[] = {
{ "core_only", "Decode core only without extensions", OFFSET(core_only), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, PARAM },
{ NULL }
};
static const AVClass dcadec_class = {
.class_name = "DCA decoder",
.item_name = av_default_item_name,
.option = dcadec_options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DECODER,
};
AVCodec ff_dca_decoder = {
.name = "dca",
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = dcadec_init,
.decode = dcadec_decode_frame,
.close = dcadec_close,
.flush = dcadec_flush,
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
.priv_class = &dcadec_class,
.profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};