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ffmpeg-mt
Commit d64d2504 authored by rtognimp's avatar rtognimp
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Cook compatibe decoder, patch by Benjamin Larsson 

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


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@4726 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b
parent 1d004364
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Showing with 1960 additions and 46 deletions
libavcodec/Makefile
View file @ d64d2504
......@@ -32,6 +32,9 @@ endif
ifeq ($(CONFIG_CINEPAK_DECODER),yes)
OBJS+= cinepak.o
endif
ifeq ($(CONFIG_COOK_DECODER),yes)
OBJS+= cook.o
endif
ifneq ($(CONFIG_CLJR_DECODER)$(CONFIG_CLJR_ENCODER),)
OBJS+= cljr.o
endif
......
libavcodec/allcodecs.c
View file @ d64d2504
......@@ -491,6 +491,9 @@ void avcodec_register_all(void)
#ifdef CONFIG_QDM2_DECODER
register_avcodec(&qdm2_decoder);
#endif //CONFIG_QDM2_DECODER
#ifdef CONFIG_COOK_DECODER
register_avcodec(&cook_decoder);
#endif //CONFIG_COOK_DECODER
#ifdef CONFIG_RAWVIDEO_DECODER
register_avcodec(&rawvideo_decoder);
#endif //CONFIG_RAWVIDEO_DECODER
......
libavcodec/avcodec.h
View file @ d64d2504
......@@ -21,8 +21,8 @@ extern "C" {
#define AV_STRINGIFY(s) AV_TOSTRING(s)
#define AV_TOSTRING(s) #s
#define LIBAVCODEC_VERSION_INT ((50<<16)+(1<<8)+0)
#define LIBAVCODEC_VERSION 50.1.0
#define LIBAVCODEC_VERSION_INT ((51<<16)+(0<<8)+0)
#define LIBAVCODEC_VERSION 51.0.0
#define LIBAVCODEC_BUILD LIBAVCODEC_VERSION_INT
#define LIBAVCODEC_IDENT "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
......@@ -187,6 +187,7 @@ enum CodecID {
CODEC_ID_WESTWOOD_SND1,
CODEC_ID_GSM,
CODEC_ID_QDM2,
CODEC_ID_COOK,
CODEC_ID_OGGTHEORA= 0x16000,
......@@ -2004,6 +2005,7 @@ extern AVCodec mp3_decoder;
extern AVCodec mp3adu_decoder;
extern AVCodec mp3on4_decoder;
extern AVCodec qdm2_decoder;
extern AVCodec cook_decoder;
extern AVCodec mace3_decoder;
extern AVCodec mace6_decoder;
extern AVCodec huffyuv_decoder;
......
libavcodec/cook.c 0 → 100644
View file @ d64d2504
This diff is collapsed.
libavcodec/cookdata.h 0 → 100644
View file @ d64d2504
This diff is collapsed.
libavcodec/ra288.c
View file @ d64d2504
......@@ -228,41 +228,19 @@ static int ra288_decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
uint8_t * buf, int buf_size)
{
if(avctx->extradata_size>=6)
{
//((short*)(avctx->extradata))[0]; /* subpacket size */
//((short*)(avctx->extradata))[1]; /* subpacket height */
//((short*)(avctx->extradata))[2]; /* subpacket flavour */
//((short*)(avctx->extradata))[3]; /* coded frame size */
//((short*)(avctx->extradata))[4]; /* codec's data length */
//((short*)(avctx->extradata))[5...] /* codec's data */
int bret;
void *datao;
int w=avctx->block_align; /* 228 */
int h=((short*)(avctx->extradata))[1]; /* 12 */
int cfs=((short*)(avctx->extradata))[3]; /* coded frame size 38 */
int i,j;
if(buf_size<w*h)
if (buf_size < avctx->block_align)
{
av_log(avctx, AV_LOG_ERROR, "ffra288: Error! Input buffer is too small [%d<%d]\n",buf_size,w*h);
av_log(avctx, AV_LOG_ERROR, "ffra288: Error! Input buffer is too small [%d<%d]\n",buf_size,avctx->block_align);
return 0;
}
datao = data;
bret = 0;
for (j = 0; j < h/2; j++)
for (i = 0; i < h; i++)
{
data=decode_block(avctx,&buf[j*cfs+cfs*i*h/2],(signed short *)data,cfs);
bret += cfs;
}
data = decode_block(avctx, buf, (signed short *)data, avctx->block_align);
*data_size = (char *)data - (char *)datao;
return bret;
}
else
{
av_log(avctx, AV_LOG_ERROR, "ffra288: Error: need extra data!!!\n");
return 0;
}
return avctx->block_align;
}
AVCodec ra_288_decoder =
......
libavformat/avformat.h
View file @ d64d2504
......@@ -5,8 +5,8 @@
extern "C" {
#endif
#define LIBAVFORMAT_VERSION_INT ((49<<16)+(2<<8)+0)
#define LIBAVFORMAT_VERSION 49.2.0
#define LIBAVFORMAT_VERSION_INT ((50<<16)+(0<<8)+0)
#define LIBAVFORMAT_VERSION 50.0.0
#define LIBAVFORMAT_BUILD LIBAVFORMAT_VERSION_INT
#define LIBAVFORMAT_IDENT "Lavf" AV_STRINGIFY(LIBAVFORMAT_VERSION)
......
libavformat/rm.c
View file @ d64d2504
......@@ -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;
}
......
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