8svx/iff: fix decoding of compressed stereo 8svx files.

Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.

Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da2a
pull/2/head
Justin Ruggles 13 years ago
parent fda459cee7
commit 1993c6849c
  1. 101
      libavcodec/8svx.c
  2. 45
      libavformat/iff.c

@ -32,8 +32,14 @@
/** decoder context */ /** decoder context */
typedef struct EightSvxContext { typedef struct EightSvxContext {
uint8_t fib_acc; uint8_t fib_acc[2];
const int8_t *table; const int8_t *table;
/* buffer used to store the whole first packet.
data is only sent as one large packet */
uint8_t *data[2];
int data_size;
int data_idx;
} EightSvxContext; } EightSvxContext;
static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1,
@ -41,6 +47,8 @@ static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1,
static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1,
0, 1, 2, 4, 8, 16, 32, 64 }; 0, 1, 2, 4, 8, 16, 32, 64 };
#define MAX_FRAME_SIZE 32768
/** /**
* Delta decode the compressed values in src, and put the resulting * Delta decode the compressed values in src, and put the resulting
* decoded samples in dst. * decoded samples in dst.
@ -48,16 +56,18 @@ static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1,
* @param[in,out] state starting value. it is saved for use in the next call. * @param[in,out] state starting value. it is saved for use in the next call.
*/ */
static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size, static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
uint8_t *state, const int8_t *table) uint8_t *state, const int8_t *table, int channels)
{ {
uint8_t val = *state; uint8_t val = *state;
while (src_size--) { while (src_size--) {
uint8_t d = *src++; uint8_t d = *src++;
val = av_clip_uint8(val + table[d & 0xF]); val = av_clip_uint8(val + table[d & 0xF]);
*dst++ = val; *dst = val;
dst += channels;
val = av_clip_uint8(val + table[d >> 4]); val = av_clip_uint8(val + table[d >> 4]);
*dst++ = val; *dst = val;
dst += channels;
} }
*state = val; *state = val;
@ -67,33 +77,69 @@ static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt) AVPacket *avpkt)
{ {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
EightSvxContext *esc = avctx->priv_data; EightSvxContext *esc = avctx->priv_data;
int buf_size;
uint8_t *out_data = data; uint8_t *out_data = data;
int consumed = buf_size; int out_data_size;
/* for the first packet, copy data to buffer */
if (avpkt->data) {
int chan_size = (avpkt->size / avctx->channels) - 2;
if(avctx->frame_number == 0) { if (avpkt->size < 2) {
if (buf_size < 2) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n"); av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR(EINVAL); return AVERROR(EINVAL);
} }
esc->fib_acc = (int8_t)buf[1] + 128; if (esc->data[0]) {
buf_size -= 2; av_log(avctx, AV_LOG_ERROR, "unexpected data after first packet\n");
buf += 2; return AVERROR(EINVAL);
}
esc->fib_acc[0] = avpkt->data[1] + 128;
if (avctx->channels == 2)
esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;
esc->data_idx = 0;
esc->data_size = chan_size;
if (!(esc->data[0] = av_malloc(chan_size)))
return AVERROR(ENOMEM);
if (avctx->channels == 2) {
if (!(esc->data[1] = av_malloc(chan_size))) {
av_freep(&esc->data[0]);
return AVERROR(ENOMEM);
}
}
memcpy(esc->data[0], &avpkt->data[2], chan_size);
if (avctx->channels == 2)
memcpy(esc->data[1], &avpkt->data[2+chan_size+2], chan_size);
}
if (!esc->data[0]) {
av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
return AVERROR(EINVAL);
} }
if (*data_size < buf_size * 2) { /* decode next piece of data from the buffer */
buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
if (buf_size <= 0) {
*data_size = 0;
return avpkt->size;
}
out_data_size = buf_size * 2 * avctx->channels;
if (*data_size < out_data_size) {
av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n",
*data_size); *data_size);
return AVERROR(EINVAL); return AVERROR(EINVAL);
} }
delta_decode(out_data, &esc->data[0][esc->data_idx], buf_size,
&esc->fib_acc[0], esc->table, avctx->channels);
if (avctx->channels == 2) {
delta_decode(&out_data[1], &esc->data[1][esc->data_idx], buf_size,
&esc->fib_acc[1], esc->table, avctx->channels);
}
esc->data_idx += buf_size;
*data_size = out_data_size;
delta_decode(out_data, buf, buf_size, &esc->fib_acc, esc->table); return avpkt->size;
*data_size = buf_size * 2;
return consumed;
} }
/** initialize 8svx decoder */ /** initialize 8svx decoder */
@ -101,6 +147,11 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{ {
EightSvxContext *esc = avctx->priv_data; EightSvxContext *esc = avctx->priv_data;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
return AVERROR(EINVAL);
}
switch(avctx->codec->id) { switch(avctx->codec->id) {
case CODEC_ID_8SVX_FIB: case CODEC_ID_8SVX_FIB:
esc->table = fibonacci; esc->table = fibonacci;
@ -115,13 +166,25 @@ static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
return 0; return 0;
} }
static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
av_freep(&esc->data[0]);
av_freep(&esc->data[1]);
return 0;
}
AVCodec ff_eightsvx_fib_decoder = { AVCodec ff_eightsvx_fib_decoder = {
.name = "8svx_fib", .name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO, .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_8SVX_FIB, .id = CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext), .priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init, .init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame, .decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
}; };
@ -131,6 +194,8 @@ AVCodec ff_eightsvx_exp_decoder = {
.id = CODEC_ID_8SVX_EXP, .id = CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext), .priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init, .init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame, .decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DELAY,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
}; };

@ -59,8 +59,6 @@
#define RIGHT 4 #define RIGHT 4
#define STEREO 6 #define STEREO 6
#define PACKET_SIZE 1024
typedef enum { typedef enum {
COMP_NONE, COMP_NONE,
COMP_FIB, COMP_FIB,
@ -76,22 +74,9 @@ typedef struct {
uint64_t body_pos; uint64_t body_pos;
uint32_t body_size; uint32_t body_size;
uint32_t sent_bytes; uint32_t sent_bytes;
uint32_t audio_frame_count;
} IffDemuxContext; } IffDemuxContext;
static void interleave_stereo(const uint8_t *src, uint8_t *dest, int size)
{
uint8_t *end = dest + size;
size = size>>1;
while(dest < end) {
*dest++ = *src;
*dest++ = *(src+size);
src++;
}
}
/* Metadata string read */ /* Metadata string read */
static int get_metadata(AVFormatContext *s, static int get_metadata(AVFormatContext *s,
const char *const tag, const char *const tag,
@ -278,40 +263,20 @@ static int iff_read_packet(AVFormatContext *s,
{ {
IffDemuxContext *iff = s->priv_data; IffDemuxContext *iff = s->priv_data;
AVIOContext *pb = s->pb; AVIOContext *pb = s->pb;
AVStream *st = s->streams[0];
int ret; int ret;
if(iff->sent_bytes >= iff->body_size) if(iff->sent_bytes >= iff->body_size)
return AVERROR(EIO); return AVERROR_EOF;
if(st->codec->channels == 2) {
uint8_t sample_buffer[PACKET_SIZE];
ret = avio_read(pb, sample_buffer, PACKET_SIZE); ret = av_get_packet(pb, pkt, iff->body_size);
if(av_new_packet(pkt, PACKET_SIZE) < 0) { if (ret < 0)
av_log(s, AV_LOG_ERROR, "cannot allocate packet\n"); return ret;
return AVERROR(ENOMEM);
}
interleave_stereo(sample_buffer, pkt->data, PACKET_SIZE);
} else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
ret = av_get_packet(pb, pkt, iff->body_size);
} else {
ret = av_get_packet(pb, pkt, PACKET_SIZE);
}
if(iff->sent_bytes == 0) if(iff->sent_bytes == 0)
pkt->flags |= AV_PKT_FLAG_KEY; pkt->flags |= AV_PKT_FLAG_KEY;
iff->sent_bytes = iff->body_size;
if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
iff->sent_bytes += PACKET_SIZE;
} else {
iff->sent_bytes = iff->body_size;
}
pkt->stream_index = 0; pkt->stream_index = 0;
if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
pkt->pts = iff->audio_frame_count;
iff->audio_frame_count += ret / st->codec->channels;
}
return ret; return ret;
} }

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