static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1;
while(len<minlen){
int len2=0;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) break; // error
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio(sh_audio->context,(int16_t*)buf,&len2,start,x);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
sh_audio->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
}
return len;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen) {
double pts;
context_t *ctx = sh->context;
int len, framelen, framesamples;
char *packet;
int i, err;
speex_decoder_ctl(ctx->dec_context, SPEEX_GET_FRAME_SIZE, &framesamples);
framelen = framesamples * ctx->hdr->nb_channels * sizeof(short);
if (maxlen < ctx->hdr->frames_per_packet * framelen) {
mp_msg(MSGT_DECAUDIO, MSGL_V, "maxlen too small in decode_audio\n");
return -1;
}
len = ds_get_packet_pts(sh->ds, (unsigned char **)&packet, &pts);
if (len <= 0) return -1;
if (sh->pts == MP_NOPTS_VALUE)
sh->pts = 0;
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
speex_bits_read_from(&ctx->bits, packet, len);
i = ctx->hdr->frames_per_packet;
do {
err = speex_decode_int(ctx->dec_context, &ctx->bits, (short *)buf);
if (err == -2)
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Error decoding file.\n");
if (ctx->hdr->nb_channels == 2)
speex_decode_stereo_int((short *)buf, framesamples, &ctx->stereo);
buf = &buf[framelen];
} while (--i > 0);
sh->pts_bytes += ctx->hdr->frames_per_packet * framelen;
return ctx->hdr->frames_per_packet * framelen;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1;
while(len<minlen){
AVPacket pkt;
int len2=maxlen;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) {
start = NULL;
x = 0;
ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x <= 0)
break; // error
} else {
int in_size = x;
int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
sh_audio->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
// LATM may need many packets to find mux info
if (y == AVERROR(EAGAIN))
continue;
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(!sh_audio->parser && y<x)
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int samplesize = av_get_bytes_per_sample(((AVCodecContext *)
sh_audio->context)->sample_fmt);
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels,
len2 / samplesize, samplesize);
}
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
maxlen -= len2;
sh_audio->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
if (setup_format(sh_audio, sh_audio->context))
break;
}
return len;
}
static int decode_audio(sh_audio_t *sh, unsigned char *buf,
int minlen, int maxlen)
{
struct spdifContext *spdif_ctx = sh->context;
AVFormatContext *lavf_ctx = spdif_ctx->lavf_ctx;
AVPacket pkt;
double pts;
int ret, in_size, consumed, x;
unsigned char *start = NULL;
consumed = spdif_ctx->out_buffer_len = 0;
spdif_ctx->out_buffer_size = maxlen;
spdif_ctx->out_buffer = buf;
while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
&& spdif_ctx->out_buffer_len < minlen) {
if (sh->ds->eof)
break;
x = ds_get_packet_pts(sh->ds, &start, &pts);
if (x <= 0) {
x = 0;
ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x == 0)
continue; // END_NOT_FOUND
in_size = x;
} else {
in_size = x;
consumed = ds_parse(sh->ds, &start, &x, pts, 0);
if (x == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] in_size[%d] consumed[%d] x[%d].\n",
start, in_size, consumed, x);
continue; // END_NOT_FOUND
}
sh->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
mp_msg(MSGT_DECAUDIO,MSGL_V,
"start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
start, pkt.size, in_size, consumed, x);
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
if (ret < 0)
break;
}
sh->pts_bytes += spdif_ctx->out_buffer_len;
return spdif_ctx->out_buffer_len;
}
static int control(sh_audio_t *sh, int cmd, void *arg)
{
unsigned char *start;
double pts;
switch (cmd) {
case ADCTRL_RESYNC_STREAM:
case ADCTRL_SKIP_FRAME:
ds_get_packet_pts(sh->ds, &start, &pts);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int len = 0;
int samples;
#ifdef CONFIG_TREMOR
ogg_int32_t **pcm;
#else
float scale;
float **pcm;
#endif
struct ov_struct_st *ov = sh->context;
while(len < minlen) {
while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))<=0){
ogg_packet op;
double pts;
memset(&op,0,sizeof(op)); //op.b_o_s = op.e_o_s = 0;
op.bytes = ds_get_packet_pts(sh->ds,&op.packet, &pts);
if(op.bytes<=0) break;
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */
vorbis_synthesis_blockin(&ov->vd,&ov->vb);
}
if(samples<=0) break; // error/EOF
while(samples>0){
int i,j;
int clipflag=0;
int convsize=(maxlen-len)/(2*ov->vi.channels); // max size!
int bout=((samples<convsize)?samples:convsize);
if(bout<=0) break; // no buffer space
/* convert floats to 16 bit signed ints (host order) and
interleave */
#ifdef CONFIG_TREMOR
if (ov->rg_scale_int == 64) {
for(i=0;i<ov->vi.channels;i++){
ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
ogg_int16_t *ptr=convbuffer+i;
ogg_int32_t *mono=pcm[i];
for(j=0;j<bout;j++){
int val=mono[j]>>9;
/* might as well guard against clipping */
if(val>32767){
val=32767;
clipflag=1;
}
if(val<-32768){
val=-32768;
clipflag=1;
}
*ptr=val;
ptr+=ov->vi.channels;
}
}
} else
#endif /* CONFIG_TREMOR */
{
#ifndef CONFIG_TREMOR
scale = 32767.f * ov->rg_scale;
#endif
for(i=0;i<ov->vi.channels;i++){
ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
ogg_int16_t *ptr=convbuffer+i;
#ifdef CONFIG_TREMOR
ogg_int32_t *mono=pcm[i];
for(j=0;j<bout;j++){
int val=(mono[j]*ov->rg_scale_int)>>(9+6);
#else
float *mono=pcm[i];
for(j=0;j<bout;j++){
int val=mono[j]*scale;
/* might as well guard against clipping */
if(val>32767){
val=32767;
clipflag=1;
}
if(val<-32768){
val=-32768;
clipflag=1;
}
#endif /* CONFIG_TREMOR */
*ptr=val;
ptr+=ov->vi.channels;
}
}
}
if(clipflag)
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence));
len+=2*ov->vi.channels*bout;
sh->pts_bytes += 2*ov->vi.channels*bout;
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
samples-=bout;
vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how
many samples we
actually consumed */
} //while(samples>0)
// if (!samples) break; // why? how?
}
if (len > 0 && ov->vi.channels >= 5) {
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_VORBIS_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
ov->vi.channels, len / sh->samplesize,
sh->samplesize);
}
return len;
}
static int init(sh_audio_t *sh, const char *decoder)
{
int x, in_size, srate, bps, *dtshd_rate;
unsigned char *start;
double pts;
AVFormatContext *lavf_ctx = NULL;
AVStream *stream = NULL;
const AVOption *opt = NULL;
struct spdifContext *spdif_ctx = NULL;
spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
if (!spdif_ctx)
goto fail;
spdif_ctx->lavf_ctx = avformat_alloc_context();
if (!spdif_ctx->lavf_ctx)
goto fail;
sh->context = spdif_ctx;
lavf_ctx = spdif_ctx->lavf_ctx;
lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
if (!lavf_ctx->oformat)
goto fail;
lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
if (!lavf_ctx->priv_data)
goto fail;
lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
read_packet, write_packet, seek);
if (!lavf_ctx->pb)
goto fail;
stream = avformat_new_stream(lavf_ctx, 0);
if (!stream)
goto fail;
lavf_ctx->duration = AV_NOPTS_VALUE;
lavf_ctx->start_time = AV_NOPTS_VALUE;
lavf_ctx->streams[0]->codec->codec_id = mp_codec_to_av_codec_id(decoder);
lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This codec is not supported by spdifenc.\n");
goto fail;
}
// get sample_rate & bitrate from parser
x = ds_get_packet_pts(sh->ds, &start, &pts);
in_size = x;
if (x <= 0) {
pts = MP_NOPTS_VALUE;
x = 0;
}
ds_parse(sh->ds, &start, &x, pts, 0);
srate = 48000; //fake value
bps = 768000/8; //fake value
if (x && sh->avctx) { // we have parser and large enough buffer
if (sh->avctx->sample_rate < 44100) {
mp_msg(MSGT_DECAUDIO,MSGL_INFO,
"This stream sample_rate[%d Hz] may be broken. "
"Force reset 48000Hz.\n",
sh->avctx->sample_rate);
srate = 48000; //fake value
} else
srate = sh->avctx->sample_rate;
bps = sh->avctx->bit_rate/8;
}
sh->ds->buffer_pos -= in_size;
switch (lavf_ctx->streams[0]->codec->codec_id) {
case AV_CODEC_ID_AAC:
spdif_ctx->iec61937_packet_size = 16384;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_AC3:
spdif_ctx->iec61937_packet_size = 6144;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_DTS:
if(sh->opts->dtshd) {
opt = av_opt_find(&lavf_ctx->oformat->priv_class,
"dtshd_rate", NULL, 0, 0);
if (!opt)
goto fail;
dtshd_rate = (int*)(((uint8_t*)lavf_ctx->priv_data) +
opt->offset);
*dtshd_rate = 192000*4;
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000; // DTS core require 48000
sh->channels = 2*4;
sh->i_bps = bps;
} else {
spdif_ctx->iec61937_packet_size = 32768;
sh->sample_format = AF_FORMAT_AC3_LE;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
}
break;
case AV_CODEC_ID_EAC3:
spdif_ctx->iec61937_packet_size = 24576;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_MP3:
spdif_ctx->iec61937_packet_size = 4608;
sh->sample_format = AF_FORMAT_MPEG2;
sh->samplerate = srate;
sh->channels = 2;
sh->i_bps = bps;
break;
case AV_CODEC_ID_TRUEHD:
spdif_ctx->iec61937_packet_size = 61440;
sh->sample_format = AF_FORMAT_IEC61937_LE;
sh->samplerate = 192000;
sh->channels = 8;
sh->i_bps = bps;
break;
default:
break;
}
return 1;
fail:
uninit(sh);
return 0;
}
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;
}
/* This tries to extract a requested amount of decoded data.
* Even when you request 0 bytes, it will feed enough input so that
* the decoder _could_ have delivered something.
* Returns byte count >= 0, -1 on error.
*
* Thoughts on exact pts keeping:
* We have to assume that MPEG frames are cut in pieces by packet boundaries.
* Also, it might be possible that the first packet does not contain enough
* data to ensure initial stream sync... or re-sync on erroneous streams.
* So we need something robust to relate the decoded byte count to the correct
* time stamp. This is tricky, though. From the outside, you cannot tell if,
* after having fed two packets until the first output arrives, one should
* start counting from the first packet's pts or the second packet's.
* So, let's just count from the last fed package's pts. If the packets are
* exactly cut to MPEG frames, this will cause one frame mismatch in the
* beginning (when mpg123 peeks ahead for the following header), but will
* be corrected with the third frame already. One might add special code to
* not increment the base pts past the first packet's after a resync before
* the first decoded bytes arrived. */
static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
{
int ret = MPG123_OK;
int got = 0;
struct ad_mpg123_context *con = sh->context;
/* There will be one MPG123_NEW_FORMAT message on first open.
* This will be handled in init(). */
do {
size_t got_now = 0;
/* Feed the decoder. This will only fire from the second round on. */
if (ret == MPG123_NEED_MORE) {
int incount;
double pts;
unsigned char *inbuf;
/* Feed more input data. */
incount = ds_get_packet_pts(sh->ds, &inbuf, &pts);
if (incount <= 0)
break; /* Apparently that's it. EOF. */
/* Next bytes from that presentation time. */
if (pts != MP_NOPTS_VALUE) {
sh->pts = pts;
sh->pts_bytes = 0;
}
#ifdef AD_MPG123_FRAMEWISE
/* Have to use mpg123_feed() to avoid decoding here. */
ret = mpg123_feed(con->handle, inbuf, incount);
#else
/* Do not use mpg123_feed(), added in later libmpg123 versions. */
ret = mpg123_decode(con->handle, inbuf, incount, NULL, 0, NULL);
#endif
if (ret == MPG123_ERR)
break;
}
/* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
* Should not happen in our usage, but it is a valid return code. */
else if (ret == MPG123_ERR || ret == MPG123_DONE)
break;
/* Try to decode a bit. This is the return value that counts
* for the loop condition. */
#ifdef AD_MPG123_FRAMEWISE
if (!buf) { /* fake call just for feeding to get format */
ret = mpg123_getformat(con->handle, NULL, NULL, NULL);
} else { /* This is the decoding. One frame at a time. */
ret = mpg123_replace_buffer(con->handle, buf, count);
if (ret == MPG123_OK)
ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
}
#else
ret = mpg123_decode(con->handle, NULL, 0, buf + got, count - got,
&got_now);
#endif
got += got_now;
sh->pts_bytes += got_now;
#ifdef AD_MPG123_FRAMEWISE
} while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
#else
} while (ret == MPG123_NEED_MORE || got < count);
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int len = 0;
union smc *sc = (union smc *)sm_com; // shared memory
int temp ;
int buflen = sh->a_in_buffer_len;
int force = 0;
while(len < minlen ) {
int y;
int len2=0;
unsigned char* bufptr=NULL;
double pts;
int ret;
if (buflen<=0){
buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
// printf("\nbuflen=%d\n",buflen);
if(buflen<=0){
sh->a_in_buffer_len = 0;
return -1; //EOF
}
// if (buflen>0){
//fill buffer
// |............csd...................|
// phead buflen ptail
csd_avail_num = 0; //total available csd num
phead = bufptr;
ptail = phead + buflen ;
force = 1;
if (pts != MP_NOPTS_VALUE)
{
sh->pts = pts;
sh->pts_bytes = 0;
}
// }
}
if (phead<=ptail ){
//fill buffer [csd_buf_size]
pfillin_comin = sm_in ;
csd_avail_num = 0 ;
while ( csd_avail_num < csd_buf_size && phead<ptail){
*pfillin_comin++ = *phead++;
csd_avail_num++; //get sm_in[csd_avail_num]
}
}
//force flag :inform decoder to start subframe
sc->pdecod.paremeter1 = force; //send to server
ret = ipc_decode(sc, APE_ID, csd_avail_num, id);
if (!ret)
return 0 ; //IPC Error
else if(ret < 0)
return -1;
if (sc->sctrl.codec_id == APE_ID) //decode success!
len2 = sc->sctrl.data_len;
//copy res data
y = sc->sctrl.paremeter1 ;//y: consume byetes
res_num = csd_avail_num - (sc->sctrl.paremeter2 &0xfffffffc) ;
phead -= res_num ; //shift phead abide by residue
if(len2>0)
memcpy( buf, sm_out , len2); // sm_out -->wav out
force = 0;
if(y<0)
{
mp_msg(MSGT_DECAUDIO,MSGL_V,"APE: error code %d\n", y);
buflen = 0;
break;
}
//buflen -= y; //total-consumed
temp = buflen ;
temp-=y;
if(0<temp<=-3)//because of res allignment,it maybe minus
{
temp=0;
// if(temp<0) printf("\n%d\n",temp);
}
else if (temp <=-4) //X
{
printf("error decode");
break ;
}
buflen = temp ;
len+=len2;
buf+=len2;
sh->pts_bytes += len2;
}
sh->a_in_buffer_len = buflen;
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 decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
int j = 0, len = -1;
union smc *sc = (union smc *)sm_com; // shared memory
while(len < minlen ) {
int y;
int len2=0;
unsigned char* bufptr=NULL;
double pts;
int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
if(buflen<=0)
break; //EOF
if (pts != MP_NOPTS_VALUE)
{
sh->pts = pts;
sh->pts_bytes = 0;
}
memcpy(sm_in,bufptr, buflen);
#if 1
int ret;
ret = ipc_decode(sc, AMRNB_ID, buflen, id);
if(!ret)
return 0;
else if(ret < 0)
return -1;
#else
if (!ipc_decode(sc, AMRNB_ID, buflen, id))
return 0 ; //IPC Error
#endif
if (sc->sctrl.codec_id == AMRNB_ID) //decode success!
len2 = sc->sctrl.data_len;
y = sc->sctrl.paremeter1 ;
// sh->channels = sc->sctrl.paremeter2;
// sh->samplerate = sc->sctrl.paremeter3;
memcpy( buf, sm_out , len2); // sm_out -->16 bit wav
if(y<0)
{
mp_msg(MSGT_DECAUDIO,MSGL_V,"AMR_audio: error\n");
break;
}
if(y<buflen)
sh->ds->buffer_pos+=y-buflen; // put back data (HACK!)
if(len2>0){
// first decoding
if(len<0)
len=len2;
else
len+=len2;
buf+=len2;
sh->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
}
return len;
}
