RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-28 02:12:28 +00:00
Code Issues Projects Releases Wiki Activity
8abfc327bd
ffmpeg/libavcodec/dpcm.c
Andreas Rheinhardt 20f9727018 avcodec/codec_internal: Add FFCodec, hide internal part of AVCodec 
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.

This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-03-21 01:33:09 +01:00

433 lines
15 KiB
C
Raw Blame History

/*
* Assorted DPCM codecs
* Copyright (c) 2003 The FFmpeg project
*
* 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
* Assorted DPCM (differential pulse code modulation) audio codecs
* by Mike Melanson (melanson@pcisys.net)
* Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
* for more information on the specific data formats, visit:
* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
* SOL DPCMs implemented by Konstantin Shishkov
*
* Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
* found in the Wing Commander IV computer game. These AVI files contain
* WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
* Clearly incorrect. To detect Xan DPCM, you will probably have to
* special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
* (Xan video) for its video codec. Alternately, such AVI files also contain
* the fourcc 'Axan' in the 'auds' chunk of the AVI header.
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "internal.h"
#include "mathops.h"
typedef struct DPCMContext {
int16_t array[256];
int sample[2]; ///< previous sample (for SOL_DPCM)
const int8_t *sol_table; ///< delta table for SOL_DPCM
} DPCMContext;
static const int32_t derf_steps[96] = {
0, 1, 2, 3, 4, 5, 6, 7,
8, 9, 10, 11, 12, 13, 14, 16,
17, 19, 21, 23, 25, 28, 31, 34,
37, 41, 45, 50, 55, 60, 66, 73,
80, 88, 97, 107, 118, 130, 143, 157,
173, 190, 209, 230, 253, 279, 307, 337,
371, 408, 449, 494, 544, 598, 658, 724,
796, 876, 963, 1060, 1166, 1282, 1411, 1552,
1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132,
7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289,
16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767,
};
static const int16_t interplay_delta_table[] = {
0, 1, 2, 3, 4, 5, 6, 7,
8, 9, 10, 11, 12, 13, 14, 15,
16, 17, 18, 19, 20, 21, 22, 23,
24, 25, 26, 27, 28, 29, 30, 31,
32, 33, 34, 35, 36, 37, 38, 39,
40, 41, 42, 43, 47, 51, 56, 61,
66, 72, 79, 86, 94, 102, 112, 122,
133, 145, 158, 173, 189, 206, 225, 245,
267, 292, 318, 348, 379, 414, 452, 493,
538, 587, 640, 699, 763, 832, 908, 991,
1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
-29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
-17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
-8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
-4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
-2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
-1081, -991, -908, -832, -763, -699, -640, -587,
-538, -493, -452, -414, -379, -348, -318, -292,
-267, -245, -225, -206, -189, -173, -158, -145,
-133, -122, -112, -102, -94, -86, -79, -72,
-66, -61, -56, -51, -47, -43, -42, -41,
-40, -39, -38, -37, -36, -35, -34, -33,
-32, -31, -30, -29, -28, -27, -26, -25,
-24, -23, -22, -21, -20, -19, -18, -17,
-16, -15, -14, -13, -12, -11, -10, -9,
-8, -7, -6, -5, -4, -3, -2, -1
};
static const int8_t sol_table_old[16] = {
0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
-0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
};
static const int8_t sol_table_new[16] = {
0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
};
static const int16_t sol_table_16[128] = {
0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
};
static av_cold int dpcm_decode_init(AVCodecContext *avctx)
{
DPCMContext *s = avctx->priv_data;
int i;
if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
s->sample[0] = s->sample[1] = 0;
switch(avctx->codec->id) {
case AV_CODEC_ID_ROQ_DPCM:
/* initialize square table */
for (i = 0; i < 128; i++) {
int16_t square = i * i;
s->array[i ] = square;
s->array[i + 128] = -square;
}
break;
case AV_CODEC_ID_SOL_DPCM:
switch(avctx->codec_tag){
case 1:
s->sol_table = sol_table_old;
s->sample[0] = s->sample[1] = 0x80;
break;
case 2:
s->sol_table = sol_table_new;
s->sample[0] = s->sample[1] = 0x80;
break;
case 3:
break;
default:
av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
return -1;
}
break;
case AV_CODEC_ID_SDX2_DPCM:
for (i = -128; i < 128; i++) {
int16_t square = i * i * 2;
s->array[i+128] = i < 0 ? -square: square;
}
break;
case AV_CODEC_ID_GREMLIN_DPCM: {
int delta = 0;
int code = 64;
int step = 45;
s->array[0] = 0;
for (i = 0; i < 127; i++) {
delta += (code >> 5);
code += step;
step += 2;
s->array[i*2 + 1] = delta;
s->array[i*2 + 2] = -delta;
}
s->array[255] = delta + (code >> 5);
}
break;
default:
break;
}
if (avctx->codec->id == AV_CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
static int dpcm_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
int buf_size = avpkt->size;
DPCMContext *s = avctx->priv_data;
AVFrame *frame = data;
int out = 0, ret;
int predictor[2];
int ch = 0;
int stereo = avctx->ch_layout.nb_channels - 1;
int16_t *output_samples, *samples_end;
GetByteContext gb;
if (stereo && (buf_size & 1))
buf_size--;
bytestream2_init(&gb, avpkt->data, buf_size);
/* calculate output size */
switch(avctx->codec->id) {
case AV_CODEC_ID_ROQ_DPCM:
out = buf_size - 8;
break;
case AV_CODEC_ID_INTERPLAY_DPCM:
out = buf_size - 6 - avctx->ch_layout.nb_channels;
break;
case AV_CODEC_ID_XAN_DPCM:
out = buf_size - 2 * avctx->ch_layout.nb_channels;
break;
case AV_CODEC_ID_SOL_DPCM:
if (avctx->codec_tag != 3)
out = buf_size * 2;
else
out = buf_size;
break;
case AV_CODEC_ID_DERF_DPCM:
case AV_CODEC_ID_GREMLIN_DPCM:
case AV_CODEC_ID_SDX2_DPCM:
out = buf_size;
break;
}
if (out <= 0) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
if (out % avctx->ch_layout.nb_channels) {
av_log(avctx, AV_LOG_WARNING, "channels have differing number of samples\n");
}
/* get output buffer */
frame->nb_samples = (out + avctx->ch_layout.nb_channels - 1) / avctx->ch_layout.nb_channels;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
output_samples = (int16_t *)frame->data[0];
samples_end = output_samples + out;
switch(avctx->codec->id) {
case AV_CODEC_ID_ROQ_DPCM:
bytestream2_skipu(&gb, 6);
if (stereo) {
predictor[1] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
predictor[0] = sign_extend(bytestream2_get_byteu(&gb) << 8, 16);
} else {
predictor[0] = sign_extend(bytestream2_get_le16u(&gb), 16);
}
/* decode the samples */
while (output_samples < samples_end) {
predictor[ch] += s->array[bytestream2_get_byteu(&gb)];
predictor[ch] = av_clip_int16(predictor[ch]);
*output_samples++ = predictor[ch];
/* toggle channel */
ch ^= stereo;
}
break;
case AV_CODEC_ID_INTERPLAY_DPCM:
bytestream2_skipu(&gb, 6); /* skip over the stream mask and stream length */
for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
*output_samples++ = predictor[ch];
}
ch = 0;
while (output_samples < samples_end) {
predictor[ch] += interplay_delta_table[bytestream2_get_byteu(&gb)];
predictor[ch] = av_clip_int16(predictor[ch]);
*output_samples++ = predictor[ch];
/* toggle channel */
ch ^= stereo;
}
break;
case AV_CODEC_ID_XAN_DPCM:
{
int shift[2] = { 4, 4 };
for (ch = 0; ch < avctx->ch_layout.nb_channels; ch++)
predictor[ch] = sign_extend(bytestream2_get_le16u(&gb), 16);
ch = 0;
while (output_samples < samples_end) {
int diff = bytestream2_get_byteu(&gb);
int n = diff & 3;
if (n == 3)
shift[ch]++;
else
shift[ch] -= (2 * n);
diff = sign_extend((diff &~ 3) << 8, 16);
/* saturate the shifter to 0..31 */
shift[ch] = av_clip_uintp2(shift[ch], 5);
diff >>= shift[ch];
predictor[ch] += diff;
predictor[ch] = av_clip_int16(predictor[ch]);
*output_samples++ = predictor[ch];
/* toggle channel */
ch ^= stereo;
}
break;
}
case AV_CODEC_ID_SOL_DPCM:
if (avctx->codec_tag != 3) {
uint8_t *output_samples_u8 = frame->data[0],
*samples_end_u8 = output_samples_u8 + out;
while (output_samples_u8 < samples_end_u8) {
int n = bytestream2_get_byteu(&gb);
s->sample[0] += s->sol_table[n >> 4];
s->sample[0] = av_clip_uint8(s->sample[0]);
*output_samples_u8++ = s->sample[0];
s->sample[stereo] += s->sol_table[n & 0x0F];
s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
*output_samples_u8++ = s->sample[stereo];
}
} else {
while (output_samples < samples_end) {
int n = bytestream2_get_byteu(&gb);
if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
else s->sample[ch] += sol_table_16[n & 0x7F];
s->sample[ch] = av_clip_int16(s->sample[ch]);
*output_samples++ = s->sample[ch];
/* toggle channel */
ch ^= stereo;
}
}
break;
case AV_CODEC_ID_SDX2_DPCM:
while (output_samples < samples_end) {
int8_t n = bytestream2_get_byteu(&gb);
if (!(n & 1))
s->sample[ch] = 0;
s->sample[ch] += s->array[n + 128];
s->sample[ch] = av_clip_int16(s->sample[ch]);
*output_samples++ = s->sample[ch];
ch ^= stereo;
}
break;
case AV_CODEC_ID_GREMLIN_DPCM: {
int idx = 0;
while (output_samples < samples_end) {
uint8_t n = bytestream2_get_byteu(&gb);
*output_samples++ = s->sample[idx] += (unsigned)s->array[n];
idx ^= 1;
}
}
break;
case AV_CODEC_ID_DERF_DPCM: {
int idx = 0;
while (output_samples < samples_end) {
uint8_t n = bytestream2_get_byteu(&gb);
int index = FFMIN(n & 0x7f, 95);
s->sample[idx] += (n & 0x80 ? -1: 1) * derf_steps[index];
s->sample[idx] = av_clip_int16(s->sample[idx]);
*output_samples++ = s->sample[idx];
idx ^= stereo;
}
}
break;
}
*got_frame_ptr = 1;
return avpkt->size;
}
#define DPCM_DECODER(id_, name_, long_name_) \
const FFCodec ff_ ## name_ ## _decoder = { \
.p.name = #name_, \
.p.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
.p.type = AVMEDIA_TYPE_AUDIO, \
.p.id = id_, \
.p.capabilities = AV_CODEC_CAP_DR1, \
.priv_data_size = sizeof(DPCMContext), \
.init = dpcm_decode_init, \
.decode = dpcm_decode_frame, \
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
}
DPCM_DECODER(AV_CODEC_ID_DERF_DPCM, derf_dpcm, "DPCM Xilam DERF");
DPCM_DECODER(AV_CODEC_ID_GREMLIN_DPCM, gremlin_dpcm, "DPCM Gremlin");
DPCM_DECODER(AV_CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
DPCM_DECODER(AV_CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
DPCM_DECODER(AV_CODEC_ID_SDX2_DPCM, sdx2_dpcm, "DPCM Squareroot-Delta-Exact");
DPCM_DECODER(AV_CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
DPCM_DECODER(AV_CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");
Reference in New Issue View Git Blame Copy Permalink