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FFmpeg/libavcodec/vmdaudio.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

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/*
* Sierra VMD audio decoder
* Copyright (c) 2004 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
* Sierra VMD audio decoder
* by Vladimir "VAG" Gneushev (vagsoft at mail.ru)
* for more information on the Sierra VMD format, visit:
* http://www.pcisys.net/~melanson/codecs/
*
* The audio decoder, expects each encoded data
* chunk to be prepended with the appropriate 16-byte frame information
* record from the VMD file. It does not require the 0x330-byte VMD file
* header, but it does need the audio setup parameters passed in through
* normal libavcodec API means.
*/
#include <string.h>
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "internal.h"
#define BLOCK_TYPE_AUDIO 1
#define BLOCK_TYPE_INITIAL 2
#define BLOCK_TYPE_SILENCE 3
typedef struct VmdAudioContext {
int out_bps;
int chunk_size;
} VmdAudioContext;
static const uint16_t vmdaudio_table[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 vmdaudio_decode_init(AVCodecContext *avctx)
{
VmdAudioContext *s = avctx->priv_data;
int channels = avctx->ch_layout.nb_channels;
if (channels < 1 || channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
if (avctx->block_align < 1 || avctx->block_align % channels ||
avctx->block_align > INT_MAX - channels) {
av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
return AVERROR(EINVAL);
}
av_channel_layout_uninit(&avctx->ch_layout);
av_channel_layout_default(&avctx->ch_layout, channels);
if (avctx->bits_per_coded_sample == 16)
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
else
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
s->chunk_size = avctx->block_align + channels * (s->out_bps == 2);
av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
"block align = %d, sample rate = %d\n",
channels, avctx->bits_per_coded_sample, avctx->block_align,
avctx->sample_rate);
return 0;
}
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
int channels)
{
int ch;
const uint8_t *buf_end = buf + buf_size;
int predictor[2];
int st = channels - 1;
/* decode initial raw sample */
for (ch = 0; ch < channels; ch++) {
predictor[ch] = (int16_t)AV_RL16(buf);
buf += 2;
*out++ = predictor[ch];
}
/* decode DPCM samples */
ch = 0;
while (buf < buf_end) {
uint8_t b = *buf++;
if (b & 0x80)
predictor[ch] -= vmdaudio_table[b & 0x7F];
else
predictor[ch] += vmdaudio_table[b];
predictor[ch] = av_clip_int16(predictor[ch]);
*out++ = predictor[ch];
ch ^= st;
}
}
static int vmdaudio_decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
const uint8_t *buf_end;
int buf_size = avpkt->size;
VmdAudioContext *s = avctx->priv_data;
int block_type, silent_chunks, audio_chunks;
int ret;
uint8_t *output_samples_u8;
int16_t *output_samples_s16;
int channels = avctx->ch_layout.nb_channels;
if (buf_size < 16) {
av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
*got_frame_ptr = 0;
return buf_size;
}
block_type = buf[6];
if (block_type < BLOCK_TYPE_AUDIO || block_type > BLOCK_TYPE_SILENCE) {
av_log(avctx, AV_LOG_ERROR, "unknown block type: %d\n", block_type);
return AVERROR(EINVAL);
}
buf += 16;
buf_size -= 16;
/* get number of silent chunks */
silent_chunks = 0;
if (block_type == BLOCK_TYPE_INITIAL) {
uint32_t flags;
if (buf_size < 4) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
flags = AV_RB32(buf);
silent_chunks = av_popcount(flags);
buf += 4;
buf_size -= 4;
} else if (block_type == BLOCK_TYPE_SILENCE) {
silent_chunks = 1;
buf_size = 0; // should already be zero but set it just to be sure
}
/* ensure output buffer is large enough */
audio_chunks = buf_size / s->chunk_size;
/* drop incomplete chunks */
buf_size = audio_chunks * s->chunk_size;
if (silent_chunks + audio_chunks >= INT_MAX / avctx->block_align)
return AVERROR_INVALIDDATA;
/* get output buffer */
frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
avctx->ch_layout.nb_channels;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
output_samples_u8 = frame->data[0];
output_samples_s16 = (int16_t *)frame->data[0];
/* decode silent chunks */
if (silent_chunks > 0) {
int silent_size = avctx->block_align * silent_chunks;
av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels);
if (s->out_bps == 2) {
memset(output_samples_s16, 0x00, silent_size * 2);
output_samples_s16 += silent_size;
} else {
memset(output_samples_u8, 0x80, silent_size);
output_samples_u8 += silent_size;
}
}
/* decode audio chunks */
if (audio_chunks > 0) {
buf_end = buf + buf_size;
av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0);
while (buf_end - buf >= s->chunk_size) {
if (s->out_bps == 2) {
decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels);
output_samples_s16 += avctx->block_align;
} else {
memcpy(output_samples_u8, buf, s->chunk_size);
output_samples_u8 += avctx->block_align;
}
buf += s->chunk_size;
}
}
*got_frame_ptr = 1;
return avpkt->size;
}
const FFCodec ff_vmdaudio_decoder = {
.p.name = "vmdaudio",
.p.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_VMDAUDIO,
.priv_data_size = sizeof(VmdAudioContext),
.init = vmdaudio_decode_init,
FF_CODEC_DECODE_CB(vmdaudio_decode_frame),
.p.capabilities = AV_CODEC_CAP_DR1,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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