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Cook compatibe decoder, patch by Benjamin Larsson

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Add cook demucing, change rm demuxer so that it reorders audio packets
before sending them to the decoder, and send minimum decodeable sized
packets; pass only real codec extradata fo the decoder
Fix 28_8 decoder for the new demuxer strategy

Originally committed as revision 4726 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Benjamin Larsson
2005-12-09 16:08:18 +00:00
committed by Roberto Togni
parent 60d76256cb
commit e0f7e32970
8 changed files with 1960 additions and 46 deletions
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111
libavformat/rm.c
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@@ -42,6 +42,14 @@ typedef struct {
int old_format;
int current_stream;
int remaining_len;
/// Audio descrambling matrix parameters
uint8_t *audiobuf; ///< place to store reordered audio data
int64_t audiotimestamp; ///< Audio packet timestamp
int sub_packet_cnt; // Subpacket counter, used while reading
int sub_packet_size, sub_packet_h, coded_framesize; ///< Descrambling parameters from container
int audio_stream_num; ///< Stream number for audio packets
int audio_pkt_cnt; ///< Output packet counter
int audio_framesize; /// Audio frame size from container
} RMContext;
#ifdef CONFIG_MUXERS
@@ -478,6 +486,7 @@ static void get_str8(ByteIOContext *pb, char *buf, int buf_size)
static void rm_read_audio_stream_info(AVFormatContext *s, AVStream *st,
int read_all)
{
RMContext *rm = s->priv_data;
ByteIOContext *pb = &s->pb;
char buf[128];
uint32_t version;
@@ -500,39 +509,60 @@ static void rm_read_audio_stream_info(AVFormatContext *s, AVStream *st,
st->codec->codec_type = CODEC_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_RA_144;
} else {
int flavor, sub_packet_h, coded_framesize;
int flavor, sub_packet_h, coded_framesize, sub_packet_size;
/* old version (4) */
get_be32(pb); /* .ra4 */
get_be32(pb); /* data size */
get_be16(pb); /* version2 */
get_be32(pb); /* header size */
flavor= get_be16(pb); /* add codec info / flavor */
coded_framesize= get_be32(pb); /* coded frame size */
rm->coded_framesize = coded_framesize = get_be32(pb); /* coded frame size */
get_be32(pb); /* ??? */
get_be32(pb); /* ??? */
get_be32(pb); /* ??? */
sub_packet_h= get_be16(pb); /* 1 */
rm->sub_packet_h = sub_packet_h = get_be16(pb); /* 1 */
st->codec->block_align= get_be16(pb); /* frame size */
get_be16(pb); /* sub packet size */
rm->sub_packet_size = sub_packet_size = get_be16(pb); /* sub packet size */
get_be16(pb); /* ??? */
if (((version >> 16) & 0xff) == 5) {
get_be16(pb); get_be16(pb); get_be16(pb); }
st->codec->sample_rate = get_be16(pb);
get_be32(pb);
st->codec->channels = get_be16(pb);
if (((version >> 16) & 0xff) == 5) {
get_be32(pb);
buf[0] = get_byte(pb);
buf[1] = get_byte(pb);
buf[2] = get_byte(pb);
buf[3] = get_byte(pb);
buf[4] = 0;
} else {
get_str8(pb, buf, sizeof(buf)); /* desc */
get_str8(pb, buf, sizeof(buf)); /* desc */
}
st->codec->codec_type = CODEC_TYPE_AUDIO;
if (!strcmp(buf, "dnet")) {
st->codec->codec_id = CODEC_ID_AC3;
} else if (!strcmp(buf, "28_8")) {
st->codec->codec_id = CODEC_ID_RA_288;
st->codec->extradata_size= 10;
st->codec->extradata_size= 0;
rm->audio_framesize = st->codec->block_align;
st->codec->block_align = coded_framesize;
rm->audiobuf = av_malloc(rm->audio_framesize * sub_packet_h);
} else if (!strcmp(buf, "cook")) {
int codecdata_length, i;
get_be16(pb); get_byte(pb);
if (((version >> 16) & 0xff) == 5)
get_byte(pb);
codecdata_length = get_be32(pb);
st->codec->codec_id = CODEC_ID_COOK;
st->codec->extradata_size= codecdata_length;
st->codec->extradata= av_mallocz(st->codec->extradata_size);
/* this is completly braindead and broken, the idiot who added this codec and endianness
specific reordering to mplayer and libavcodec/ra288.c should be drowned in a see of cola */
//FIXME pass the unpermutated extradata
((uint16_t*)st->codec->extradata)[1]= sub_packet_h;
((uint16_t*)st->codec->extradata)[2]= flavor;
((uint16_t*)st->codec->extradata)[3]= coded_framesize;
for(i = 0; i < codecdata_length; i++)
((uint8_t*)st->codec->extradata)[i] = get_byte(pb);
rm->audio_framesize = st->codec->block_align;
st->codec->block_align = rm->sub_packet_size;
rm->audiobuf = av_malloc(rm->audio_framesize * sub_packet_h);
} else {
st->codec->codec_id = CODEC_ID_NONE;
pstrcpy(st->codec->codec_name, sizeof(st->codec->codec_name),
@@ -819,6 +849,16 @@ static int rm_read_packet(AVFormatContext *s, AVPacket *pkt)
}
pkt->size = len;
st = s->streams[0];
} else if (rm->audio_pkt_cnt) {
// If there are queued audio packet return them first
st = s->streams[rm->audio_stream_num];
av_new_packet(pkt, st->codec->block_align);
memcpy(pkt->data, rm->audiobuf + st->codec->block_align *
(rm->sub_packet_h * rm->audio_framesize / st->codec->block_align - rm->audio_pkt_cnt),
st->codec->block_align);
rm->audio_pkt_cnt--;
pkt->flags = 0;
pkt->stream_index = rm->audio_stream_num;
} else {
int seq=1;
resync:
@@ -850,15 +890,57 @@ resync:
if(len2 && len2<len)
len=len2;
rm->remaining_len-= len;
av_get_packet(pb, pkt, len);
}
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if ((st->codec->codec_id == CODEC_ID_RA_288) ||
(st->codec->codec_id == CODEC_ID_COOK)) {
int x;
int sps = rm->sub_packet_size;
int cfs = rm->coded_framesize;
int h = rm->sub_packet_h;
int y = rm->sub_packet_cnt;
int w = rm->audio_framesize;
if (flags & 2)
y = rm->sub_packet_cnt = 0;
if (!y)
rm->audiotimestamp = timestamp;
switch(st->codec->codec_id) {
case CODEC_ID_RA_288:
for (x = 0; x < h/2; x++)
get_buffer(pb, rm->audiobuf+x*2*w+y*cfs, cfs);
break;
case CODEC_ID_COOK:
for (x = 0; x < w/sps; x++)
get_buffer(pb, rm->audiobuf+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
break;
}
if (++(rm->sub_packet_cnt) < h)
goto resync;
else {
rm->sub_packet_cnt = 0;
rm->audio_stream_num = i;
rm->audio_pkt_cnt = h * w / st->codec->block_align - 1;
// Release first audio packet
av_new_packet(pkt, st->codec->block_align);
memcpy(pkt->data, rm->audiobuf, st->codec->block_align);
timestamp = rm->audiotimestamp;
flags = 2; // Mark first packet as keyframe
}
} else
av_get_packet(pb, pkt, len);
}
if( (st->discard >= AVDISCARD_NONKEY && !(flags&2))
|| st->discard >= AVDISCARD_ALL){
url_fskip(pb, len);
av_free_packet(pkt);
goto resync;
}
av_get_packet(pb, pkt, len);
pkt->stream_index = i;
#if 0
@@ -896,6 +978,9 @@ resync:
static int rm_read_close(AVFormatContext *s)
{
RMContext *rm = s->priv_data;
av_free(rm->audiobuf);
return 0;
}