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/*
* Sierra VMD audio decoder
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.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;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
return AVERROR(EINVAL);
}
if (avctx->block_align < 1) {
av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
return AVERROR(EINVAL);
}
avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
AV_CH_LAYOUT_STEREO;
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 + avctx->channels * (s->out_bps == 2);
av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
"block align = %d, sample rate = %d\n",
avctx->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, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
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;
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;
/* get output buffer */
frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
avctx->channels;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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 = FFMIN(avctx->block_align * silent_chunks,
frame->nb_samples * avctx->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 & ~(avctx->channels > 1));
while (buf + s->chunk_size <= buf_end) {
if (s->out_bps == 2) {
decode_audio_s16(output_samples_s16, buf, s->chunk_size,
avctx->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;
}
AVCodec ff_vmdaudio_decoder = {
.name = "vmdaudio",
.long_name = NULL_IF_CONFIG_SMALL("Sierra VMD audio"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_VMDAUDIO,
.priv_data_size = sizeof(VmdAudioContext),
.init = vmdaudio_decode_init,
.decode = vmdaudio_decode_frame,
.capabilities = CODEC_CAP_DR1,
};
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