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vmdaudio: fix decoding of 16-bit audio format. 

The initial sample of each block is raw 16-bit PCM, not DPCM.
Fixes decoding of all samples in:
http://streams.videolan.org/samples/game-formats/sierra-vmd/Lighthouse/
This commit is contained in:
Justin Ruggles 2011-09-11 20:17:54 -04:00
parent bb416bd68c
commit 4568c2bf97
1 changed files with 80 additions and 40 deletions
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120
libavcodec/vmdav.c
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@ -417,9 +417,8 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
#define BLOCK_TYPE_SILENCE 3
typedef struct VmdAudioContext {
AVCodecContext *avctx;
int out_bps;
int predictors[2];
int chunk_size;
} VmdAudioContext;
static const uint16_t vmdaudio_table[128] = {
@ -442,13 +441,23 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
{
VmdAudioContext *s = avctx->priv_data;
s->avctx = avctx;
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);
}
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,
@ -457,52 +466,47 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
return 0;
}
static void vmdaudio_decode_audio(VmdAudioContext *s, unsigned char *data,
const uint8_t *buf, int buf_size, int stereo)
static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
int channels)
{
int i;
int chan = 0;
int16_t *out = (int16_t*)data;
int ch;
const uint8_t *buf_end = buf + buf_size;
int predictor[2];
int st = channels - 1;
for(i = 0; i < buf_size; i++) {
if(buf[i] & 0x80)
s->predictors[chan] -= vmdaudio_table[buf[i] & 0x7F];
/* 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
s->predictors[chan] += vmdaudio_table[buf[i]];
s->predictors[chan] = av_clip_int16(s->predictors[chan]);
out[i] = s->predictors[chan];
chan ^= stereo;
predictor[ch] += vmdaudio_table[b];
predictor[ch] = av_clip_int16(predictor[ch]);
*out++ = predictor[ch];
ch ^= st;
}
}
static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
const uint8_t *buf, int silent_chunks, int data_size)
{
int silent_size = s->avctx->block_align * silent_chunks * s->out_bps;
if (silent_chunks) {
memset(data, s->out_bps == 2 ? 0x00 : 0x80, silent_size);
data += silent_size;
}
if (s->avctx->bits_per_coded_sample == 16)
vmdaudio_decode_audio(s, data, buf, data_size, s->avctx->channels == 2);
else {
/* just copy the data */
memcpy(data, buf, data_size);
}
return silent_size + data_size * s->out_bps;
}
static int vmdaudio_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
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;
unsigned char *output_samples = (unsigned char *)data;
int block_type, silent_chunks, audio_chunks;
int nb_samples, out_size;
uint8_t *output_samples_u8 = data;
int16_t *output_samples_s16 = data;
if (buf_size < 16) {
av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
@ -518,10 +522,16 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
buf += 16;
buf_size -= 16;
/* get number of silent chunks */
silent_chunks = 0;
if (block_type == BLOCK_TYPE_INITIAL) {
uint32_t flags = AV_RB32(buf);
silent_chunks = av_popcount(flags);
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) {
@ -530,11 +540,41 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
}
/* ensure output buffer is large enough */
if (*data_size < (avctx->block_align*silent_chunks + buf_size) * s->out_bps)
audio_chunks = buf_size / s->chunk_size;
nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels;
out_size = nb_samples * avctx->channels * s->out_bps;
if (*data_size < out_size)
return -1;
*data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);
/* decode silent chunks */
if (silent_chunks > 0) {
int silent_size = avctx->block_align * silent_chunks;
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;
while (buf < 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;
}
}
*data_size = out_size;
return avpkt->size;
}