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
This commit is contained in:
Justin Ruggles
2011-09-07 15:28:07 -04:00
parent fda459cee7
commit 1993c6849c
2 changed files with 88 additions and 58 deletions

View File

@ -59,8 +59,6 @@
#define RIGHT 4
#define STEREO 6
#define PACKET_SIZE 1024
typedef enum {
COMP_NONE,
COMP_FIB,
@ -76,22 +74,9 @@ typedef struct {
uint64_t body_pos;
uint32_t body_size;
uint32_t sent_bytes;
uint32_t audio_frame_count;
} 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 */
static int get_metadata(AVFormatContext *s,
const char *const tag,
@ -278,40 +263,20 @@ static int iff_read_packet(AVFormatContext *s,
{
IffDemuxContext *iff = s->priv_data;
AVIOContext *pb = s->pb;
AVStream *st = s->streams[0];
int ret;
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);
if(av_new_packet(pkt, PACKET_SIZE) < 0) {
av_log(s, AV_LOG_ERROR, "cannot allocate packet\n");
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);
}
ret = av_get_packet(pb, pkt, iff->body_size);
if (ret < 0)
return ret;
if(iff->sent_bytes == 0)
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;
if(st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
pkt->pts = iff->audio_frame_count;
iff->audio_frame_count += ret / st->codec->channels;
}
return ret;
}