static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
if(sh_audio->format==0x31 || sh_audio->format==0x32){
unsigned char ibuf[65]; // 65 bytes / frame
if(demux_read_data(sh_audio->ds,ibuf,65)!=65) return -1; // EOF
XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short
return 2*320;
} else {
unsigned char ibuf[33]; // 33 bytes / frame
if(demux_read_data(sh_audio->ds,ibuf,33)!=33) return -1; // EOF
XA_GSM_Decoder(ibuf,(unsigned short *) buf); // decodes 33 byte -> 160 short
return 2*160;
}
}
int stream_fill_buffer(stream_t *s){
int len;
if (/*s->fd == NULL ||*/ s->eof) { s->buf_pos = s->buf_len = 0; return 0; }
switch(s->type){
case STREAMTYPE_STREAM:
#ifdef MPLAYER_NETWORK
if( s->streaming_ctrl!=NULL ) {
len=s->streaming_ctrl->streaming_read(s->fd,s->buffer,STREAM_BUFFER_SIZE, s->streaming_ctrl);break;
} else {
len=read(s->fd,s->buffer,STREAM_BUFFER_SIZE);break;
}
#else
len=read(s->fd,s->buffer,STREAM_BUFFER_SIZE);break;
#endif
case STREAMTYPE_DS:
len = demux_read_data((demux_stream_t*)s->priv,s->buffer,STREAM_BUFFER_SIZE);
break;
default:
len= s->fill_buffer ? s->fill_buffer(s,s->buffer,STREAM_BUFFER_SIZE) : 0;
}
if(len<=0){ s->eof=1; s->buf_pos=s->buf_len=0; return 0; }
s->buf_pos=0;
s->buf_len=len;
s->pos+=len;
// printf("[%d]",len);fflush(stdout);
return len;
}
int a52_fillbuff(sh_audio_t *sh_audio){
int length=0;
int flags=0;
int sample_rate=0;
int bit_rate=0;
sh_audio->a_in_buffer_len=0;
/* sync frame:*/
while(1){
while(sh_audio->a_in_buffer_len<8){
int c=demux_getc(sh_audio->ds);
if(c<0) return -1; /* EOF*/
sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
}
if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
if(length>=7 && length<=3840) break; /* we're done.*/
/* bad file => resync*/
if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
--sh_audio->a_in_buffer_len;
}
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
sh_audio->samplerate=sample_rate;
sh_audio->i_bps=bit_rate/8;
sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
if(sh_audio->format!=0x2000)
swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
return length;
}
static int read_frame(sh_audio_t *sh){
mad_decoder_t *this = sh->context;
int len;
while((len=demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size-sh->a_in_buffer_len))>0){
sh->a_in_buffer_len+=len;
while(1){
int ret;
mad_stream_buffer (&this->stream, sh->a_in_buffer, sh->a_in_buffer_len);
ret=mad_frame_decode (&this->frame, &this->stream);
if (this->stream.next_frame) {
int num_bytes =
(char*)sh->a_in_buffer+sh->a_in_buffer_len - (char*)this->stream.next_frame;
memmove(sh->a_in_buffer, this->stream.next_frame, num_bytes);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"libmad: %d bytes processed\n",sh->a_in_buffer_len-num_bytes);
sh->a_in_buffer_len = num_bytes;
}
if (ret == 0) return 1; // OK!!!
// error! try to resync!
if(this->stream.error==MAD_ERROR_BUFLEN) break;
}
}
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"Cannot sync MAD frame\n");
return 0;
}
/**
* \brief read data until the given 3-byte pattern is encountered, up to maxlen
* \param mem memory to read data into, may be NULL to discard data
* \param maxlen maximum number of bytes to read
* \param read number of bytes actually read
* \param pattern pattern to search for (lowest 8 bits are ignored)
* \return whether pattern was found
*/
int demux_pattern_3(demux_stream_t *ds, unsigned char *mem, int maxlen,
int *read, uint32_t pattern)
{
register uint32_t head = 0xffffff00;
register uint32_t pat = pattern & 0xffffff00;
int total_len = 0;
do {
register unsigned char *ds_buf = &ds->buffer[ds->buffer_size];
int len = ds->buffer_size - ds->buffer_pos;
register long pos = -len;
if (unlikely(pos >= 0)) { // buffer is empty
ds_fill_buffer(ds);
continue;
}
do {
head |= ds_buf[pos];
head <<= 8;
} while (++pos && head != pat);
len += pos;
if (total_len + len > maxlen)
len = maxlen - total_len;
len = demux_read_data(ds, mem ? &mem[total_len] : NULL, len);
total_len += len;
} while ((head != pat || total_len < 3) && total_len < maxlen && !ds->eof);
if (read)
*read = total_len;
return total_len >= 3 && head == pat;
}
int stream_fill_buffer(stream_t *s){
int len;
// we will retry even if we already reached EOF previously.
switch(s->type){
case STREAMTYPE_STREAM:
#ifdef CONFIG_NETWORKING
if( s->streaming_ctrl!=NULL && s->streaming_ctrl->streaming_read ) {
len=s->streaming_ctrl->streaming_read(s->fd,s->buffer,STREAM_BUFFER_SIZE, s->streaming_ctrl);
} else
#endif
if (s->fill_buffer)
len = s->fill_buffer(s, s->buffer, STREAM_BUFFER_SIZE);
else
len=read(s->fd,s->buffer,STREAM_BUFFER_SIZE);
break;
case STREAMTYPE_DS:
len = demux_read_data((demux_stream_t*)s->priv,s->buffer,STREAM_BUFFER_SIZE);
break;
default:
len= s->fill_buffer ? s->fill_buffer(s,s->buffer,STREAM_BUFFER_SIZE) : 0;
}
if(len<=0){ s->eof=1; return 0; }
// When reading succeeded we are obviously not at eof.
// This e.g. avoids issues with eof getting stuck when lavf seeks in MPEG-TS
s->eof=0;
s->buf_pos=0;
s->buf_len=len;
s->pos+=len;
// printf("[%d]",len);fflush(stdout);
if (s->capture_file)
stream_capture_do(s);
return len;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
DMO_AudioDecoder* ds_adec = sh_audio->context;
// int len=-1;
int size_in=0;
int size_out=0;
int srcsize=DMO_AudioDecoder_GetSrcSize(ds_adec, maxlen);
mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DMO says: srcsize=%d (buffsize=%d) out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen);
if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!!
if(sh_audio->a_in_buffer_len<srcsize){
sh_audio->a_in_buffer_len+=
demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len],
srcsize-sh_audio->a_in_buffer_len);
}
DMO_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len,
buf,maxlen, &size_in,&size_out);
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DMO: audio %d -> %d converted (in_buf_len=%d of %d) %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds));
if(size_in>=sh_audio->a_in_buffer_len){
sh_audio->a_in_buffer_len=0;
} else {
sh_audio->a_in_buffer_len-=size_in;
memmove(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len);
}
// len=size_out;
return size_out;
}
static int mpa_sync(sh_audio_t *sh, int no_frames, int *n, int *chans, int *srate, int *spf, int *mpa_layer, int *br)
{
int cnt = 0, x = 0, len, frames_count;
frames_count = 0;
do
{
while(cnt + 4 < sh->a_in_buffer_len)
{
x = mp_get_mp3_header(&(sh->a_in_buffer[cnt]), chans, srate, spf, mpa_layer, br);
if(x > 0)
{
frames_count++;
if(frames_count == no_frames)
{
*n = x;
return cnt;
}
}
cnt++;
}
len = demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],sh->a_in_buffer_size-sh->a_in_buffer_len);
if(len > 0)
sh->a_in_buffer_len += len;
} while(len > 0);
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"Cannot sync MPA frame: %d\r\n", len);
return -1;
}
int stream_read_internal(stream_t *s, void *buf, int len)
{
int orig_len = len;
// we will retry even if we already reached EOF previously.
switch(s->type){
case STREAMTYPE_STREAM:
#ifdef CONFIG_NETWORKING
if( s->streaming_ctrl!=NULL && s->streaming_ctrl->streaming_read ) {
len=s->streaming_ctrl->streaming_read(s->fd, buf, len, s->streaming_ctrl);
if (s->streaming_ctrl->status == streaming_stopped_e)
s->eof = 1;
} else
#endif
if (s->fill_buffer)
len = s->fill_buffer(s, buf, len);
else
len = read(s->fd, buf, len);
break;
case STREAMTYPE_DS:
len = demux_read_data((demux_stream_t*)s->priv, buf, len);
break;
default:
len= s->fill_buffer ? s->fill_buffer(s, buf, len) : 0;
}
if(len<=0){
off_t pos = s->pos;
// do not retry if this looks like proper eof
if (s->eof || (s->end_pos && pos == s->end_pos))
goto eof_out;
// dvdnav has some horrible hacks to "suspend" reads,
// we need to skip this code or seeks will hang.
if (s->type == STREAMTYPE_DVDNAV)
goto eof_out;
// just in case this is an error e.g. due to network
// timeout reset and retry
// Seeking is used as a hack to make network streams
// reopen the connection, ideally they would implement
// e.g. a STREAM_CTRL_RECONNECT to do this
s->eof=1;
stream_reset(s);
if (stream_seek_internal(s, pos) >= 0 || s->pos != pos) // seek failed
goto eof_out;
// make sure EOF is set to ensure no endless loops
s->eof=1;
return stream_read_internal(s, buf, orig_len);
eof_out:
s->eof=1;
return 0;
}
// When reading succeeded we are obviously not at eof.
// This e.g. avoids issues with eof getting stuck when lavf seeks in MPEG-TS
s->eof=0;
s->pos+=len;
return len;
}
static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
{
if(cmd==ADCTRL_SKIP_FRAME){
demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned len = sh_audio->channels*sh_audio->samplesize;
len = (minlen + len - 1) / len * len;
if (len > maxlen)
// if someone needs hundreds of channels adjust audio_out_minsize
// based on channels in preinit()
return -1;
len=demux_read_data(sh_audio->ds,buf,len);
return len;
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
int skip;
switch(cmd)
{
case ADCTRL_SKIP_FRAME:
skip=sh->i_bps/16;
skip=skip&(~3);
demux_read_data(sh->ds,NULL,skip);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
int stream_read_internal(stream_t *s, void *buf, int len)
{
int orig_len = len;
// we will retry even if we already reached EOF previously.
switch(s->type){
case STREAMTYPE_STREAM:
#ifdef CONFIG_NETWORKING
if( s->streaming_ctrl!=NULL && s->streaming_ctrl->streaming_read ) {
len=s->streaming_ctrl->streaming_read(s->fd, buf, len, s->streaming_ctrl);
if (s->streaming_ctrl->status == streaming_stopped_e &&
(!s->end_pos || s->pos == s->end_pos))
s->eof = 1;
} else
#endif
if (s->fill_buffer)
len = s->fill_buffer(s, buf, len);
else
len = read(s->fd, buf, len);
break;
case STREAMTYPE_DS:
len = demux_read_data((demux_stream_t*)s->priv, buf, len);
break;
default:
len= s->fill_buffer ? s->fill_buffer(s, buf, len) : 0;
}
if(len<=0){
// do not retry if this looks like proper eof
if (s->eof || (s->end_pos && s->pos == s->end_pos))
goto eof_out;
// just in case this is an error e.g. due to network
// timeout reset and retry
if (!stream_reconnect(s))
goto eof_out;
// make sure EOF is set to ensure no endless loops
s->eof=1;
return stream_read_internal(s, buf, orig_len);
eof_out:
s->eof=1;
return 0;
}
// When reading succeeded we are obviously not at eof.
// This e.g. avoids issues with eof getting stuck when lavf seeks in MPEG-TS
s->eof=0;
s->pos+=len;
return len;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
sh_audio->ds->ss_mul) !=
sh_audio->ds->ss_mul)
return -1; /* EOF */
if (maxlen < 2 * 4 * sh_audio->wf->nBlockAlign * 2 / 3) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "dk3adpcm: maxlen too small in decode_audio\n");
return -1;
}
return 2 * dk3_adpcm_decode_block(
(unsigned short*)buf, sh_audio->a_in_buffer,
sh_audio->ds->ss_mul);
}
static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
{
int skip;
switch(cmd)
{
case ADCTRL_SKIP_FRAME:
skip=sh_audio->wf->nBlockAlign;
if(skip<16){
skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7);
if(skip<16) skip=16;
}
demux_read_data(sh_audio->ds,NULL,skip);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
int len;
int l=demux_read_data(sh_audio->ds,buf,minlen/2);
unsigned short *d=(unsigned short *) buf;
unsigned char *s=buf;
len=2*l;
if(sh_audio->format==6 || sh_audio->format==0x77616C61){
/* aLaw */
while(l>0){ --l; d[l]=alaw2short[s[l]]; }
} else {
/* uLaw */
while(l>0){ --l; d[l]=ulaw2short[s[l]]; }
}
return len;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned len = sh_audio->channels*sh_audio->samplesize;
len = (minlen + len - 1) / len * len;
if (len > maxlen)
// if someone needs hundreds of channels adjust audio_out_minsize
// based on channels in preinit()
return -1;
len=demux_read_data(sh_audio->ds,buf,len);
if (len > 0 && sh_audio->channels >= 5) {
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
sh_audio->channels,
len / sh_audio->samplesize, sh_audio->samplesize);
}
return len;
}
static int aac_sync(sh_audio_t *sh)
{
int pos = 0;
if(!sh->codecdata_len) {
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
}
}
return pos;
}
static int aac_sync(sh_audio_t *sh)
{
int pos = 0;
// do not probe LATM, faad does that
if(!sh->codecdata_len && sh->format != mmioFOURCC('M', 'P', '4', 'L')) {
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
}
}
return pos;
}
static int bread(char *data, /* Output: Output data array */
int size, /* Input: Length of each data */
int nbits, /* Input: Number of bits to write */
sh_audio_t *sh) /* Input: File pointer */
{
/*--- Variables ---*/
int ibits, iptr, idata, ibufadr, ibufbit, icl;
unsigned char mask, tmpdat;
int retval;
vqf_priv_t *priv=sh->context;
/*--- Main operation ---*/
retval = 0;
mask = 0x1;
for ( ibits=0; ibits<nbits; ibits++ ){
if ( priv->readable == 0 ){ /* when the file data buffer is empty */
priv->nbuf = demux_read_data(sh->ds, priv->buf, BBUFSIZ);
priv->nbuf *= 8;
priv->readable = 1;
}
iptr = priv->ptr; /* current file data buffer pointer */
if ( iptr >= priv->nbuf ) /* If data file is empty then return */
return retval;
ibufadr = iptr/BYTE_BIT; /* current file data buffer address */
ibufbit = iptr%BYTE_BIT; /* current file data buffer bit */
/* tmpdat = stream->buf[ibufadr] >> (BYTE_BIT-ibufbit-1); */
tmpdat = (unsigned char)priv->buf[ibufadr];
tmpdat >>= (BYTE_BIT-ibufbit-1);
/* current data bit */
idata = ibits*size; /* output data address */
data[idata] = (char)(tmpdat & mask); /* set output data */
for (icl=1; icl<size; icl++)
data[idata+icl] = 0; /* clear the rest output data buffer */
priv->ptr += 1; /* update data buffer pointer */
if (priv->ptr == BBUFLEN){
priv->ptr = 0;
priv->readable = 0;
}
++retval;
}
return retval;
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int len, start, tot;
int chans, srate, spf, mpa_layer, br;
int tot2;
tot = tot2 = 0;
while(tot2 < maxlen)
{
start = mpa_sync(sh, 1, &len, &chans, &srate, &spf, &mpa_layer, &br);
if(start < 0 || tot2 + spf * 2 * chans > maxlen)
break;
if(start + len > sh->a_in_buffer_len)
{
int l;
l = FFMIN(sh->a_in_buffer_size - sh->a_in_buffer_len, start + len);
l = demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], l);
if(! l)
break;
sh->a_in_buffer_len += l;
continue;
}
memcpy(&buf[tot], &(sh->a_in_buffer[start]), len);
tot += len;
sh->a_in_buffer_len -= start + len;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[start + len]), sh->a_in_buffer_len);
tot2 += spf * 2 * chans;
/* HACK: seems to fix most A/V sync issues */
break;
}
memset(&buf[tot], 0, tot2-tot);
return tot2;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
ACMSTREAMHEADER ash;
HRESULT hr;
DWORD srcsize=0;
DWORD len=minlen;
acm_context_t *priv = sh_audio->context;
acmStreamSize(priv->handle, len, &srcsize, ACM_STREAMSIZEF_DESTINATION);
mp_msg(MSGT_WIN32,MSGL_DBG3,"acm says: srcsize=%ld (buffsize=%d) out_size=%ld\n",srcsize,sh_audio->a_in_buffer_size,len);
if(srcsize<sh_audio->wf->nBlockAlign){
srcsize=sh_audio->wf->nBlockAlign;
acmStreamSize(priv->handle, srcsize, &len, ACM_STREAMSIZEF_SOURCE);
if(len>maxlen) len=maxlen;
}
// if(srcsize==0) srcsize=((WAVEFORMATEX *)&sh_audio->o_wf_ext)->nBlockAlign;
if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!!
if(sh_audio->a_in_buffer_len<srcsize){
sh_audio->a_in_buffer_len+=
demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len],
srcsize-sh_audio->a_in_buffer_len);
}
mp_msg(MSGT_WIN32,MSGL_DBG3,"acm convert %d -> %ld bytes\n",sh_audio->a_in_buffer_len,len);
memset(&ash, 0, sizeof(ash));
ash.cbStruct=sizeof(ash);
ash.fdwStatus=0;
ash.dwUser=0;
ash.pbSrc=sh_audio->a_in_buffer;
ash.cbSrcLength=sh_audio->a_in_buffer_len;
ash.pbDst=buf;
ash.cbDstLength=len;
hr=acmStreamPrepareHeader(priv->handle,&ash,0);
if(hr){
mp_msg(MSGT_WIN32,MSGL_V,"ACM_Decoder: acmStreamPrepareHeader error %d\n",(int)hr);
return -1;
}
hr=acmStreamConvert(priv->handle,&ash,0);
if(hr){
mp_msg(MSGT_WIN32,MSGL_DBG2,"ACM_Decoder: acmStreamConvert error %d\n",(int)hr);
switch(hr)
{
case ACMERR_NOTPOSSIBLE:
case ACMERR_UNPREPARED:
mp_msg(MSGT_WIN32, MSGL_DBG2, "ACM_Decoder: acmStreamConvert error: probarly not initialized!\n");
}
// return -1;
}
mp_msg(MSGT_WIN32,MSGL_DBG2,"acm converted %ld -> %ld\n",ash.cbSrcLengthUsed,ash.cbDstLengthUsed);
if(ash.cbSrcLengthUsed>=sh_audio->a_in_buffer_len){
sh_audio->a_in_buffer_len=0;
} else {
sh_audio->a_in_buffer_len-=ash.cbSrcLengthUsed;
memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[ash.cbSrcLengthUsed],sh_audio->a_in_buffer_len);
}
len=ash.cbDstLengthUsed;
hr=acmStreamUnprepareHeader(priv->handle,&ash,0);
if(hr){
mp_msg(MSGT_WIN32,MSGL_V,"ACM_Decoder: acmStreamUnprepareHeader error %d\n",(int)hr);
}
return len;
}
static void demux_seek_avi(demuxer_t *demuxer, float rel_seek_secs,
float audio_delay, int flags)
{
avi_priv_t *priv=demuxer->priv;
demux_stream_t *d_audio=demuxer->audio;
demux_stream_t *d_video=demuxer->video;
sh_audio_t *sh_audio=d_audio->sh;
sh_video_t *sh_video=d_video->sh;
float skip_audio_secs=0;
//FIXME: OFF_T - Didn't check AVI case yet (avi files can't be >2G anyway?)
//================= seek in AVI ==========================
int rel_seek_frames=rel_seek_secs*sh_video->fps;
int video_chunk_pos=d_video->pos;
int i;
if(flags&SEEK_ABSOLUTE){
// seek absolute
video_chunk_pos=0;
}
if(flags&SEEK_FACTOR){
rel_seek_frames=rel_seek_secs*priv->numberofframes;
}
priv->skip_video_frames=0;
priv->avi_audio_pts=0;
// ------------ STEP 1: find nearest video keyframe chunk ------------
// find nearest video keyframe chunk pos:
if(rel_seek_frames>0){
// seek forward
while(video_chunk_pos<priv->idx_size-1){
int id=((AVIINDEXENTRY *)priv->idx)[video_chunk_pos].ckid;
if(avi_stream_id(id)==d_video->id){ // video frame
if((--rel_seek_frames)<0 && ((AVIINDEXENTRY *)priv->idx)[video_chunk_pos].dwFlags&AVIIF_KEYFRAME) break;
}
++video_chunk_pos;
}
} else {
// seek backward
while(video_chunk_pos>0){
int id=((AVIINDEXENTRY *)priv->idx)[video_chunk_pos].ckid;
if(avi_stream_id(id)==d_video->id){ // video frame
if((++rel_seek_frames)>0 && ((AVIINDEXENTRY *)priv->idx)[video_chunk_pos].dwFlags&AVIIF_KEYFRAME) break;
}
--video_chunk_pos;
}
}
priv->idx_pos_a=priv->idx_pos_v=priv->idx_pos=video_chunk_pos;
// re-calc video pts:
d_video->pack_no=0;
for(i=0;i<video_chunk_pos;i++){
int id=((AVIINDEXENTRY *)priv->idx)[i].ckid;
if(avi_stream_id(id)==d_video->id) ++d_video->pack_no;
}
priv->video_pack_no=
sh_video->num_frames=sh_video->num_frames_decoded=d_video->pack_no;
priv->avi_video_pts=d_video->pack_no*(float)sh_video->video.dwScale/(float)sh_video->video.dwRate;
d_video->pos=video_chunk_pos;
mp_msg(MSGT_SEEK,MSGL_DBG2,"V_SEEK: pack=%d pts=%5.3f chunk=%d \n",d_video->pack_no,priv->avi_video_pts,video_chunk_pos);
// ------------ STEP 2: seek audio, find the right chunk & pos ------------
d_audio->pack_no=0;
priv->audio_block_no=0;
d_audio->dpos=0;
if(sh_audio){
int i;
int len=0;
int skip_audio_bytes=0;
int curr_audio_pos=-1;
int audio_chunk_pos=-1;
int chunk_max=(demuxer->type==DEMUXER_TYPE_AVI)?video_chunk_pos:priv->idx_size;
if(sh_audio->audio.dwSampleSize){
// constant rate audio stream
/* immediate seeking to audio position, including when streams are delayed */
curr_audio_pos=(priv->avi_video_pts + audio_delay)*(float)sh_audio->audio.dwRate/(float)sh_audio->audio.dwScale;
curr_audio_pos*=sh_audio->audio.dwSampleSize;
// find audio chunk pos:
for(i=0;i<chunk_max;i++){
int id=((AVIINDEXENTRY *)priv->idx)[i].ckid;
if(avi_stream_id(id)==d_audio->id){
len=((AVIINDEXENTRY *)priv->idx)[i].dwChunkLength;
if(d_audio->dpos<=curr_audio_pos && curr_audio_pos<(d_audio->dpos+len)){
break;
}
++d_audio->pack_no;
priv->audio_block_no+=priv->audio_block_size ?
((len+priv->audio_block_size-1)/priv->audio_block_size) : 1;
d_audio->dpos+=len;
}
}
audio_chunk_pos=i;
skip_audio_bytes=curr_audio_pos-d_audio->dpos;
mp_msg(MSGT_SEEK,MSGL_V,"SEEK: i=%d (max:%d) dpos=%d (wanted:%d) \n",
i,chunk_max,(int)d_audio->dpos,curr_audio_pos);
} else {
// VBR audio
/* immediate seeking to audio position, including when streams are delayed */
int chunks=(priv->avi_video_pts + audio_delay)*(float)sh_audio->audio.dwRate/(float)sh_audio->audio.dwScale;
audio_chunk_pos=0;
// find audio chunk pos:
for(i=0;i<priv->idx_size && chunks>0;i++){
int id=((AVIINDEXENTRY *)priv->idx)[i].ckid;
if(avi_stream_id(id)==d_audio->id){
len=((AVIINDEXENTRY *)priv->idx)[i].dwChunkLength;
if(i>chunk_max){
skip_audio_bytes+=len;
} else {
++d_audio->pack_no;
priv->audio_block_no+=priv->audio_block_size ?
((len+priv->audio_block_size-1)/priv->audio_block_size) : 1;
d_audio->dpos+=len;
audio_chunk_pos=i;
}
if(priv->audio_block_size)
chunks-=(len+priv->audio_block_size-1)/priv->audio_block_size;
}
}
}
// Now we have:
// audio_chunk_pos = chunk no in index table (it's <=chunk_max)
// skip_audio_bytes = bytes to be skipped after chunk seek
// d-audio->pack_no = chunk_no in stream at audio_chunk_pos
// d_audio->dpos = bytepos in stream at audio_chunk_pos
// let's seek!
// update stream position:
d_audio->pos=audio_chunk_pos;
if(demuxer->type==DEMUXER_TYPE_AVI){
// interleaved stream:
if(audio_chunk_pos<video_chunk_pos){
// calc priv->skip_video_frames & adjust video pts counter:
for(i=audio_chunk_pos;i<video_chunk_pos;i++){
int id=((AVIINDEXENTRY *)priv->idx)[i].ckid;
if(avi_stream_id(id)==d_video->id) ++priv->skip_video_frames;
}
// requires for correct audio pts calculation (demuxer):
priv->avi_video_pts-=priv->skip_video_frames*(float)sh_video->video.dwScale/(float)sh_video->video.dwRate;
priv->avi_audio_pts=priv->avi_video_pts;
// set index position:
priv->idx_pos_a=priv->idx_pos_v=priv->idx_pos=audio_chunk_pos;
}
} else {
// non-interleaved stream:
priv->idx_pos_a=audio_chunk_pos;
priv->idx_pos_v=video_chunk_pos;
priv->idx_pos=(audio_chunk_pos<video_chunk_pos)?audio_chunk_pos:video_chunk_pos;
}
mp_msg(MSGT_SEEK,MSGL_V,"SEEK: idx=%d (a:%d v:%d) v.skip=%d a.skip=%d/%4.3f \n",
(int)priv->idx_pos,audio_chunk_pos,video_chunk_pos,
(int)priv->skip_video_frames,skip_audio_bytes,skip_audio_secs);
if(skip_audio_bytes){
demux_read_data(d_audio,NULL,skip_audio_bytes);
}
}
d_video->pts=priv->avi_video_pts; // OSD
}
static int init(sh_audio_t *sh)
{
unsigned long faac_samplerate;
unsigned char faac_channels;
int faac_init, pos = 0;
faac_hdec = faacDecOpen();
// If we don't get the ES descriptor, try manual config
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = malloc(sh->codecdata_len);
memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len) {
faacDecConfigurationPtr faac_conf;
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if(sh->samplerate)
faac_conf->defSampleRate = sh->samplerate;
/* XXX: FAAD support FLOAT output, how do we handle
* that (FAAD_FMT_FLOAT)? ::atmos
*/
if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
switch(sh->samplesize){
case 1: // 8Bit
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
default:
sh->samplesize=2;
case 2: // 16Bit
faac_conf->outputFormat = FAAD_FMT_16BIT;
break;
case 3: // 24Bit
faac_conf->outputFormat = FAAD_FMT_24BIT;
break;
case 4: // 32Bit
faac_conf->outputFormat = FAAD_FMT_32BIT;
break;
}
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
faacDecSetConfiguration(faac_hdec, faac_conf);
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
#if CONFIG_FAAD_INTERNAL
/* init the codec, look for LATM */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,1);
if (faac_init < 0 && sh->a_in_buffer_len >= 3 && sh->format == mmioFOURCC('M', 'P', '4', 'L')) {
// working LATM not found at first try, look further on in stream
int i;
for (i = 0; i < 5; i++) {
pos = sh->a_in_buffer_len-3;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), 3);
sh->a_in_buffer_len = 3;
sh->a_in_buffer_len += demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,1);
if (faac_init >= 0) break;
}
}
#else
/* external faad does not have latm lookup support */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
#endif
if (faac_init < 0) {
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
sh->a_in_buffer_size - sh->a_in_buffer_len);
pos = 0;
}
/* init the codec */
#if CONFIG_FAAD_INTERNAL
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,0);
#else
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
#endif
}
sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
// XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
} else { // We have ES DS in codecdata
faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if (audio_output_channels <= 2) {
faac_conf->downMatrix = 1;
faacDecSetConfiguration(faac_hdec, faac_conf);
}
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_init = faacDecInit2(faac_hdec, sh->codecdata,
sh->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_init < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
// XXX: free a_in_buffer here or in uninit?
return 0;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
// 8 channels is aac channel order #7.
sh->channels = faac_channels == 7 ? 8 : faac_channels;
if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
sh->samplerate = faac_samplerate;
sh->samplesize=2;
//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
}
return 1;
}
static int init(sh_audio_t *sh)
{
unsigned long faac_samplerate;
unsigned char faac_channels;
int faac_init, pos = 0;
faac_hdec = faacDecOpen();
// If we don't get the ES descriptor, try manual config
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = (char*)(sh->wf+1);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len) {
#if 1
faacDecConfigurationPtr faac_conf;
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if(sh->samplerate)
faac_conf->defSampleRate = sh->samplerate;
/* XXX: FAAD support FLOAT output, how do we handle
* that (FAAD_FMT_FLOAT)? ::atmos
*/
if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
switch(sh->samplesize){
case 1: // 8Bit
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
default:
sh->samplesize=2;
case 2: // 16Bit
faac_conf->outputFormat = FAAD_FMT_16BIT;
break;
case 3: // 24Bit
faac_conf->outputFormat = FAAD_FMT_24BIT;
break;
case 4: // 32Bit
faac_conf->outputFormat = FAAD_FMT_32BIT;
break;
}
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
faacDecSetConfiguration(faac_hdec, faac_conf);
#endif
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
sh->a_in_buffer_size - sh->a_in_buffer_len);
pos = 0;
}
/* init the codec */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
// XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
} else { // We have ES DS in codecdata
faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if (audio_output_channels <= 2) {
faac_conf->downMatrix = 1;
faacDecSetConfiguration(faac_hdec, faac_conf);
}
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_init = faacDecInit2(faac_hdec, sh->codecdata,
sh->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_init < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
// XXX: free a_in_buffer here or in uninit?
return 0;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
sh->channels = faac_channels;
if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
/* re-map channels */
switch (sh->channels) {
default:
case 1: /* no action needed */
case 2: /* no action needed */
case 3: /* no suitable default behavior? */
case 4: /* no suitable default behavior? */
break;
case 5: /* mplayer treats this like 6-channel */
case 6:
sh->chan_map = af_set_channel_map(6, "04" "10" "21" "32" "43" "55");
break;
case 7: /* not supported by mplayer? */
break;
}
sh->samplerate = faac_samplerate;
sh->samplesize=2;
//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
}
return 1;
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int j = 0, len = 0, last_dec_len = 1, errors = 0;
void *faac_sample_buffer;
while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {
/* update buffer for raw aac streams: */
if(!sh->codecdata_len)
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
#ifdef DUMP_AAC_COMPRESSED
{int i;
for (i = 0; i < 16; i++)
printf ("%02X ", sh->a_in_buffer[i]);
printf ("\n");}
#endif
if(!sh->codecdata_len){
// raw aac stream:
do {
faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
/* update buffer index after faacDecDecode */
if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
sh->a_in_buffer_len=0;
} else {
sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
}
if(faac_finfo.error > 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
faacDecGetErrorMessage(faac_finfo.error));
if (sh->a_in_buffer_len <= 0) {
errors = MAX_FAAD_ERRORS;
break;
}
sh->a_in_buffer_len--;
memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
aac_sync(sh);
errors++;
} else
break;
} while(errors < MAX_FAAD_ERRORS);
} else {
// packetized (.mp4) aac stream:
unsigned char* bufptr=NULL;
double pts;
int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
if(buflen<=0) break;
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
}
//for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
if(faac_finfo.error > 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
faacDecGetErrorMessage(faac_finfo.error));
} else if (faac_finfo.samples == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
} else {
/* XXX: samples already multiplied by channels! */
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
sh->samplesize*faac_finfo.samples);
memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
last_dec_len = sh->samplesize*faac_finfo.samples;
len += last_dec_len;
sh->pts_bytes += last_dec_len;
//printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed);
}
}
return len;
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen){
int len = 0, last_dec_len = 1, err = 0, in_len;
int outOfData = 0, errors=0;;
uint8_t *inBuf;
while(len < minlen && last_dec_len > 0 ) {
/* update buffer for raw aac streams: */
if(!sh->codecdata_len){
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
//#define DUMP_AAC_COMPRESSED
#ifdef DUMP_AAC_COMPRESSED
{int i;
inBuf = sh->a_in_buffer;
for (i = 0; i < 16; i++)
printf ("%02x ", inBuf[i]);
printf ("\n");}
#endif
inBuf = sh->a_in_buffer;
in_len = sh->a_in_buffer_len;
// raw aac stream:
err = AACDecode(hAACDecoder, &(inBuf), &(in_len), buf+len);
if(in_len > 0) {
memmove(sh->a_in_buffer,inBuf,in_len);
}
sh->a_in_buffer_len = in_len;
if (err == ERR_AAC_NONE)
{
AACGetLastFrameInfo(hAACDecoder, frameInfo);
reorder_ch_pcm(buf+len, frameInfo->outputSamps, frameInfo->nChans);
last_dec_len = sh->samplesize*frameInfo->outputSamps*sh->channels/frameInfo->nChans;
len += last_dec_len;
sh->pts_bytes += last_dec_len;
if(errors)
errors=0;
}else if(err == ERR_AAC_INDATA_UNDERFLOW){
errors++;
}else if(sh->a_in_buffer_len > 0){
sh->a_in_buffer_len--;
memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
aac_sync(sh);
errors++;
}
if(errors > 10){
break;
}
}else{
/* update buffer for raw aac streams: */
// packetized (.mp4) aac stream:
unsigned char* bufptr=NULL;
double pts;
int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
if(buflen<=0) break;
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
err = AACDecode(hAACDecoder, &bufptr, &buflen, buf+len);
if (err == ERR_AAC_NONE)
{
AACGetLastFrameInfo(hAACDecoder, frameInfo);
reorder_ch_pcm(buf+len, frameInfo->outputSamps, frameInfo->nChans);
last_dec_len = sh->samplesize*frameInfo->outputSamps*sh->channels/frameInfo->nChans;
len += last_dec_len;
sh->pts_bytes += last_dec_len;
}
}
}
return len;
}
static int init(sh_audio_t *sh){
int pos = 0;
int32_t result = HXR_OK;
uint16_t *temp;
uint8_t *inBuf;
int in_len;
int32_t NumChannels, SampleRate;
hAACDecoder = (HAACDecoder *)AACInitDecoder();
ga_config_data configData;
int i;
if (!hAACDecoder) {
printf(" *** Error initializing decoder ***\n");
return 0;
}
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = (char*)(sh->wf+1);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"realaac: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len){
//return 0;
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
inBuf = sh->a_in_buffer;
in_len = sh->a_in_buffer_len;
temp = (uint16_t *)malloc(sizeof(uint16_t) * sh->audio_out_minsize);
/* decode ADTS frame from opaque data to get config data */
result = AACDecode(hAACDecoder, &inBuf, &in_len, temp);
free(temp);
if (result != HXR_OK)
{
return 0;
}
/* get the config data struct from the decoder */
AACGetLastFrameInfo(hAACDecoder, frameInfo);
NumChannels = frameInfo->nChans;
SampleRate = frameInfo->sampRateCore;
}else{
struct BITSTREAM *pBitstream ;
pBitstream = (struct BITSTREAM *)malloc(sizeof(struct BITSTREAM)) ;
if (!pBitstream)
return 0 ;
pBitstream->cacheBitsLeft = 0 ;
pBitstream->buffer = sh->codecdata ;
pBitstream->nBits = sh->codecdata_len*8 ;
pBitstream->pkptr = pBitstream->buffer + (0>>3) ;
pBitstream->cacheBitsLeft = 0 ;
pBitstream->cache = 0 ;
pBitstream->inc = 1 ;
result = ga_config_get_data(pBitstream, &configData);
if(pBitstream){
free(pBitstream);
}
if (result != HXR_OK) /* config data error */
{
return 0;
}
NumChannels = frameInfo->nChans=configData.numChannels;
SampleRate = frameInfo->sampRateCore=configData.samplingFrequency;
/* see MPEG4 spec for index of each object type */
if (configData.audioObjectType == 2)
frameInfo->profile = AAC_PROFILE_LC;
else if (configData.audioObjectType == 1)
frameInfo->profile = AAC_PROFILE_MP;
else if (configData.audioObjectType == 3)
frameInfo->profile = AAC_PROFILE_SSR;
else
return 0; /* don't know - assume LC */
/* set decoder to handle raw data blocks */
AACSetRawBlockParams(hAACDecoder, 0, frameInfo);
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"realaac: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"realaac: Negotiated samplerate: %ldHz channels: %d\n", SampleRate, NumChannels);
sh->channels = NumChannels;
if (audio_output_channels <= 2) sh->channels = NumChannels > 1 ? 2 : 1;
sh->samplerate = SampleRate;
sh->samplesize=2;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"realaac: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"realaac: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
return 1; // return values: 1=OK 0=ERROR
}
// MP3 decoder buffer callback:
int mplayer_audio_read(char *buf,int size){
return demux_read_data(dec_audio_sh->ds,buf,size);
}
static int preinit(sh_audio_t *sh){
// let's check if the driver is available, return 0 if not.
// (you should do that if you use external lib(s) which is optional)
unsigned int result;
char *path;
path = malloc(strlen(codec_path) + strlen(sh->codec->dll) + 2);
if (!path) return 0;
sprintf(path, "%s/%s", codec_path, sh->codec->dll);
/* first try to load linux dlls, if failed and we're supporting win32 dlls,
then try to load the windows ones */
#ifdef HAVE_LIBDL
if (strstr(sh->codec->dll,".dll") || !load_syms_linux(path))
#endif
#ifdef CONFIG_WIN32DLL
if (!load_syms_windows(sh->codec->dll))
#endif
{
mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_MissingDLLcodec, sh->codec->dll);
mp_msg(MSGT_DECVIDEO, MSGL_HINT, "Read the RealAudio section of the DOCS!\n");
free(path);
return 0;
}
#ifdef CONFIG_WIN32DLL
if((raSetDLLAccessPath && dll_type == 0) || (wraSetDLLAccessPath && dll_type == 1)){
#else
if(raSetDLLAccessPath){
#endif
// used by 'SIPR'
path = realloc(path, strlen(codec_path) + 13);
sprintf(path, "DT_Codecs=%s", codec_path);
if(path[strlen(path)-1]!='/'){
path[strlen(path)+1]=0;
path[strlen(path)]='/';
}
path[strlen(path)+1]=0;
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
{
int i;
for (i=0; i < strlen(path); i++)
if (path[i] == '/') path[i] = '\\';
wraSetDLLAccessPath(path);
}
else
#endif
raSetDLLAccessPath(path);
}
#ifdef CONFIG_WIN32DLL
if (dll_type == 1){
if (wraOpenCodec2) {
sprintf(path, "%s\\", codec_path);
result = wraOpenCodec2(&sh->context, path);
} else
result=wraOpenCodec(&sh->context);
} else
#endif
if (raOpenCodec2) {
sprintf(path, "%s/", codec_path);
result = raOpenCodec2(&sh->context, path);
} else
result=raOpenCodec(&sh->context);
if(result){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"Decoder open failed, error code: 0x%X\n",result);
return 0;
}
// printf("opencodec ok (result: %x)\n", result);
free(path); /* after this it isn't used anymore */
sh->samplerate=sh->wf->nSamplesPerSec;
sh->samplesize=sh->wf->wBitsPerSample/8;
sh->channels=sh->wf->nChannels;
{
ra_init_t init_data={
sh->wf->nSamplesPerSec,
sh->wf->wBitsPerSample,
sh->wf->nChannels,
100, // quality
sh->wf->nBlockAlign, // subpacket size
sh->wf->nBlockAlign, // coded frame size
sh->wf->cbSize, // codec data length
(char*)(sh->wf+1) // extras
};
#ifdef CONFIG_WIN32DLL
wra_init_t winit_data={
sh->wf->nSamplesPerSec,
sh->wf->wBitsPerSample,
sh->wf->nChannels,
100, // quality
sh->wf->nBlockAlign, // subpacket size
sh->wf->nBlockAlign, // coded frame size
sh->wf->cbSize, // codec data length
(char*)(sh->wf+1) // extras
};
#endif
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
result=wraInitDecoder(sh->context,&winit_data);
else
#endif
result=raInitDecoder(sh->context,&init_data);
if(result){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"Decoder init failed, error code: 0x%X\n",result);
return 0;
}
// printf("initdecoder ok (result: %x)\n", result);
}
#ifdef CONFIG_WIN32DLL
if((raSetPwd && dll_type == 0) || (wraSetPwd && dll_type == 1)){
#else
if(raSetPwd){
#endif
// used by 'SIPR'
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
wraSetPwd(sh->context,"Ardubancel Quazanga");
else
#endif
raSetPwd(sh->context,"Ardubancel Quazanga"); // set password... lol.
}
if (sh->format == mmioFOURCC('s','i','p','r')) {
short flavor;
if (sh->wf->nAvgBytesPerSec > 1531)
flavor = 3;
else if (sh->wf->nAvgBytesPerSec > 937)
flavor = 1;
else if (sh->wf->nAvgBytesPerSec > 719)
flavor = 0;
else
flavor = 2;
mp_msg(MSGT_DECAUDIO,MSGL_V,"Got sipr flavor %d from bitrate %d\n",flavor, sh->wf->nAvgBytesPerSec);
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
result=wraSetFlavor(sh->context,flavor);
else
#endif
result=raSetFlavor(sh->context,flavor);
if(result){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"Decoder flavor setup failed, error code: 0x%X\n",result);
return 0;
}
} // sipr flavor
sh->i_bps=sh->wf->nAvgBytesPerSec;
sh->audio_out_minsize=128000; // no idea how to get... :(
sh->audio_in_minsize = sh->wf->nBlockAlign;
return 1; // return values: 1=OK 0=ERROR
}
static int init(sh_audio_t *sh_audio){
// initialize the decoder, set tables etc...
// you can store HANDLE or private struct pointer at sh->context
// you can access WAVEFORMATEX header at sh->wf
// set sample format/rate parameters if you didn't do it in preinit() yet.
return 1; // return values: 1=OK 0=ERROR
}
static void uninit(sh_audio_t *sh){
// uninit the decoder etc...
// again: you don't have to free() a_in_buffer here! it's done by the core.
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
{
if (wraFreeDecoder) wraFreeDecoder(sh->context);
if (wraCloseCodec) wraCloseCodec(sh->context);
}
#endif
if (raFreeDecoder) raFreeDecoder(sh->context);
if (raCloseCodec) raCloseCodec(sh->context);
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
{
if (rv_handle) FreeLibrary(rv_handle);
} else
#endif
// this dlclose() causes some memory corruption, and crashes soon (in caller):
// if (rv_handle) dlclose(rv_handle);
rv_handle = NULL;
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen){
int result;
int len=-1;
if(sh->a_in_buffer_len<=0){
// fill the buffer!
if (sh->ds->eof)
return 0;
demux_read_data(sh->ds, sh->a_in_buffer, sh->wf->nBlockAlign);
sh->a_in_buffer_size=
sh->a_in_buffer_len=sh->wf->nBlockAlign;
}
#ifdef CONFIG_WIN32DLL
if (dll_type == 1)
result=wraDecode(sh->context, sh->a_in_buffer+sh->a_in_buffer_size-sh->a_in_buffer_len, sh->wf->nBlockAlign,
buf, &len, -1);
else
#endif
result=raDecode(sh->context, sh->a_in_buffer+sh->a_in_buffer_size-sh->a_in_buffer_len, sh->wf->nBlockAlign,
buf, &len, -1);
sh->a_in_buffer_len-=sh->wf->nBlockAlign;
// printf("radecode: %d bytes, res=0x%X \n",len,result);
return len; // return value: number of _bytes_ written to output buffer,
// or -1 for EOF (or uncorrectable error)
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...){
// various optional functions you MAY implement:
switch(cmd){
case ADCTRL_RESYNC_STREAM:
// it is called once after seeking, to resync.
// Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
return CONTROL_TRUE;
case ADCTRL_SKIP_FRAME:
// it is called to skip (jump over) small amount (1/10 sec or 1 frame)
// of audio data - used to sync audio to video after seeking
// if you don't return CONTROL_TRUE, it will defaults to:
// ds_fill_buffer(sh_audio->ds); // skip 1 demux packet
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}