static int a52_fillbuff(a52_buffer_t * buf)
{
int length = 0;
int flags = 0;
int sample_rate = 0;
int bit_rate = 0;
buf->a_in_buffer_len = 0;
while (1) {
while (buf->a_in_buffer_len < 7) {
int c = read_byte();
if (c < 0)
return -1;
buf->a_in_buffer[buf->a_in_buffer_len++] = c;
}
length =
a52_syncinfo(buf->a_in_buffer, &flags, &sample_rate,
&bit_rate);
//a52_flags |= flags;
if (length >= 7 && length <= 3840)
break;
memmove(buf->a_in_buffer, buf->a_in_buffer + 1, 6);
--buf->a_in_buffer_len;
}
buf->samplerate = sample_rate;
buf->bps = bit_rate / 8;
a52_read((buf->a_in_buffer + 7), (length - 7));
return length;
}
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;
}
/* Return 1 if it isn't an AC3 header */
int mpeg3_ac3_check(unsigned char *header)
{
int flags, samplerate, bitrate;
return !a52_syncinfo(header,
&flags,
&samplerate,
&bitrate);
}
static GstFlowReturn
gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
gint * _offset, gint * len)
{
GstA52Dec *a52dec;
const guint8 *data;
gint av, size;
gint length = 0, flags, sample_rate, bit_rate;
GstFlowReturn result = GST_FLOW_EOS;
a52dec = GST_A52DEC (bdec);
size = av = gst_adapter_available (adapter);
data = (const guint8 *) gst_adapter_map (adapter, av);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* shift window to re-find sync */
data++;
size--;
} else if (length <= size) {
GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
result = GST_FLOW_OK;
break;
} else {
GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
length, size);
break;
}
}
gst_adapter_unmap (adapter);
*_offset = av - size;
*len = length;
return result;
}
/* returns: number of available channels*/
static int a52_printinfo(sh_audio_t *sh_audio){
int flags, sample_rate, bit_rate;
char* mode="unknown";
int channels=0;
a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
switch(flags&A52_CHANNEL_MASK){
case A52_CHANNEL: mode="channel"; channels=2; break;
case A52_MONO: mode="mono"; channels=1; break;
case A52_STEREO: mode="stereo"; channels=2; break;
case A52_3F: mode="3f";channels=3;break;
case A52_2F1R: mode="2f+1r";channels=3;break;
case A52_3F1R: mode="3f+1r";channels=4;break;
case A52_2F2R: mode="2f+2r";channels=4;break;
case A52_3F2R: mode="3f+2r";channels=5;break;
case A52_CHANNEL1: mode="channel1"; channels=2; break;
case A52_CHANNEL2: mode="channel2"; channels=2; break;
case A52_DOLBY: mode="dolby"; channels=2; break;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
channels, (flags&A52_LFE)?1:0,
mode, (flags&A52_LFE)?"+lfe":"",
sample_rate, bit_rate*0.001f);
return (flags&A52_LFE) ? (channels+1) : channels;
}
static void a52_decode_data(uint8_t *start, uint8_t *end)
{
static uint8_t *bufptr = buf;
static uint8_t *bufpos = buf + 7;
/*
* sample_rate and flags are static because this routine could
* exit between the a52_syncinfo() and the ao_setup(), and we want
* to have the same values when we get back !
*/
static int sample_rate;
static int flags;
int bit_rate;
int len;
while (1) {
len = end - start;
if (!len)
break;
if (len > bufpos - bufptr)
len = bufpos - bufptr;
memcpy(bufptr, start, len);
bufptr += len;
start += len;
if (bufptr == bufpos) {
if (bufpos == buf + 7) {
int length;
length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
if (!length) {
//DEBUGF("skip\n");
for (bufptr = buf; bufptr < buf + 6; bufptr++)
bufptr[0] = bufptr[1];
continue;
}
bufpos = buf + length;
} else {
/* Unity gain is 1 << 26, and we want to end up on 28 bits
of precision instead of the default 30.
*/
level_t level = 1 << 24;
sample_t bias = 0;
int i;
/* This is the configuration for the downmixing: */
flags = A52_STEREO | A52_ADJUST_LEVEL;
if (a52_frame(state, buf, &flags, &level, bias))
goto error;
a52_dynrng(state, NULL, NULL);
frequency = sample_rate;
/* An A52 frame consists of 6 blocks of 256 samples
So we decode and output them one block at a time */
for (i = 0; i < 6; i++) {
if (a52_block(state))
goto error;
output_audio(a52_samples(state));
samplesdone += 256;
}
ci->set_elapsed(samplesdone/(frequency/1000));
bufptr = buf;
bufpos = buf + 7;
continue;
error:
//logf("Error decoding A52 stream\n");
bufptr = buf;
bufpos = buf + 7;
}
}
}
}
static GF_Err AC3_ProcessData(GF_MediaDecoder *ifcg,
char *inBuffer, u32 inBufferLength,
u16 ES_ID,
char *outBuffer, u32 *outBufferLength,
u8 PaddingBits, u32 mmlevel)
{
short *out_samples;
int i, len, bit_rate;
sample_t level;
A52CTX();
/*check not using scalabilty*/
if (ctx->ES_ID != ES_ID) return GF_BAD_PARAM;
/*if late or seeking don't decode*/
switch (mmlevel) {
case GF_CODEC_LEVEL_SEEK:
case GF_CODEC_LEVEL_DROP:
*outBufferLength = 0;
return GF_OK;
default:
break;
}
if (ctx->out_size > *outBufferLength) {
*outBufferLength = ctx->out_size;
return GF_BUFFER_TOO_SMALL;
}
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[A52] Decoding AU\n"));
len = a52_syncinfo(inBuffer, &ctx->flags, &ctx->sample_rate, &bit_rate);
if (!len) return GF_NON_COMPLIANT_BITSTREAM;
/*init decoder*/
if (!ctx->out_size) {
ctx->num_channels = ac3_channels[ctx->flags & 7];
if (ctx->flags & A52_LFE) ctx->num_channels++;
ctx->flags |= A52_ADJUST_LEVEL;
ctx->out_size = ctx->num_channels * sizeof(short) * 1536;
*outBufferLength = ctx->out_size;
return GF_BUFFER_TOO_SMALL;
}
level = 1;
if ( a52_frame(ctx->codec, inBuffer, &ctx->flags, &level, 384)) {
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[A52] Error decoding AU\n" ));
*outBufferLength = 0;
return GF_NON_COMPLIANT_BITSTREAM;
}
out_samples = (short*)outBuffer;
for (i=0; i<6; i++) {
if (a52_block(ctx->codec))
return GF_NON_COMPLIANT_BITSTREAM;
float_to_int(ctx->samples, out_samples + i * 256 * ctx->num_channels, ctx->num_channels);
}
*outBufferLength = 6 * ctx->num_channels * 256 * sizeof(short);
return GF_OK;
}
uint8_t ADM_AudiocodecAC3::run(uint8_t *inptr, uint32_t nbIn, float *outptr, uint32_t *nbOut)
{
uint32_t avail;
uint32_t length;
int flags = 0, samprate = 0, bitrate = 0;
uint8_t chan = _wavHeader->channels;
*nbOut=0;
// Ready to decode
while(nbIn)
{
if(nbIn<7)
{
if(nbIn)
printf("[a52]: no data to decode avail %u\n",nbIn);
break;
}
length = a52_syncinfo(inptr, &flags, &samprate, &bitrate);
if(length==0)
{
printf("[a52] No startcode found\n");
break;
}
if(length>nbIn)
{
// not enough data
break;
}
CHANNEL_TYPE *p_ch_type = channelMapping;
if (flags & A52_LFE) {
*(p_ch_type++) = CHTYP_LFE;
}
switch (flags & A52_CHANNEL_MASK) {
case A52_CHANNEL:
case A52_MONO:
*(p_ch_type++) = CHTYP_MONO;
break;
case A52_STEREO:
case A52_DOLBY:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
break;
case A52_3F:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_CENTER;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
break;
case A52_2F1R:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
*(p_ch_type++) = CHTYP_REAR_CENTER;
break;
case A52_3F1R:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_CENTER;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
*(p_ch_type++) = CHTYP_REAR_CENTER;
break;
case A52_2F2R:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
*(p_ch_type++) = CHTYP_REAR_LEFT;
*(p_ch_type++) = CHTYP_REAR_RIGHT;
break;
case A52_3F2R:
*(p_ch_type++) = CHTYP_FRONT_LEFT;
*(p_ch_type++) = CHTYP_FRONT_CENTER;
*(p_ch_type++) = CHTYP_FRONT_RIGHT;
*(p_ch_type++) = CHTYP_REAR_LEFT;
*(p_ch_type++) = CHTYP_REAR_RIGHT;
break;
default:
ADM_assert(0);
}
sample_t level = 1, bias = 0;
if (a52_frame(AC3_HANDLE, inptr, &flags, &level, bias))
{
printf("\n A52_frame failed!");
inptr+=length;
nbIn-=length;
*nbOut += 256 * chan * 6;
break;
};
inptr+=length;
nbIn-=length;
*nbOut += 256 * chan * 6;
float *cur;
for (int i = 0; i < 6; i++) {
if (a52_block(AC3_HANDLE)) {
printf("\n A52_block failed! on fblock :%lu", i);
// in that case we silent out the chunk
memset(outptr, 0, 256 * chan * sizeof(float));
} else {
for (int k = 0; k < chan; k++) {
sample_t *sample=(sample_t *)ac3_sample;
sample += 256 * k;
cur = outptr + k;
for (int j = 0; j < 256; j++) {
*cur = *sample++;
cur+=chan;
}
}
}
outptr += chan * 256;
}
}
return 1;
}
/* Decode AC3 header */
int mpeg3_ac3_header(mpeg3_ac3_t *audio, unsigned char *header)
{
int result = 0;
audio->flags = 0;
//printf("mpeg3_ac3_header %02x%02x%02x%02x%02x%02x%02x%02x\n", header[0], header[1], header[2], header[3], header[4], header[5], header[6], header[7]);
result = a52_syncinfo(header,
&audio->flags,
&audio->samplerate,
&audio->bitrate);
if(result)
{
//printf("%d\n", result);
audio->framesize = result;
audio->channels = 0;
if(audio->flags & A52_LFE)
audio->channels++;
/*
* printf("mpeg3_ac3_header %08x %08x\n",
* audio->flags & A52_LFE,
* audio->flags & A52_CHANNEL_MASK);
*/
switch(audio->flags & A52_CHANNEL_MASK)
{
case A52_CHANNEL:
audio->channels++;
break;
case A52_MONO:
audio->channels++;
break;
case A52_STEREO:
audio->channels += 2;
break;
case A52_3F:
audio->channels += 3;
break;
case A52_2F1R:
audio->channels += 3;
break;
case A52_3F1R:
audio->channels += 4;
break;
case A52_2F2R:
audio->channels += 4;
break;
case A52_3F2R:
audio->channels += 5;
break;
case A52_DOLBY:
audio->channels += 2;
break;
default:
printf("mpeg3_ac3_header: unknown channel code: %x\n", audio->flags & A52_CHANNEL_MASK);
break;
}
}
//printf("mpeg3_ac3_header 1 %d\n", audio->channels);
return result;
}
HRESULT TaudioCodecLiba52::decode(TbyteBuffer &src)
{
unsigned char *p=src.size() ? &src[0] : NULL;
unsigned char *base=p;
unsigned char *end=p+src.size();
while (end-p>7) {
int size=0,flags,sample_rate,bit_rate;
if ((size=a52_syncinfo(p,&flags,&sample_rate,&bit_rate))>0) {
bool enoughData=p+size<=end;
if (enoughData) {
if (codecId==CODEC_ID_SPDIF_AC3) {
bpssum+=(lastbps=bit_rate/1000);
numframes++;
HRESULT hr=deciA->deliverSampleSPDIF(p,size,bit_rate,sample_rate,true);
if (hr!=S_OK) {
return hr;
}
} else {
flags|=A52_ADJUST_LEVEL;
liba52::sample_t level=1,bias=0;
if (a52_frame(state,p,&flags,&level,bias)==0) {
bpssum+=(lastbps=bit_rate/1000);
numframes++;
// Dynamic range compression
if (deci->getParam2(IDFF_audio_decoder_DRC)) {
liba52::sample_t drcLevel = ((liba52::sample_t)deci->getParam2(IDFF_audio_decoder_DRC_Level) / 100);
a52_dynrngsetlevel(state, drcLevel);
} else {
a52_dynrngsetlevel(state, 0.0);
}
int scmapidx=std::min(flags&A52_CHANNEL_MASK,int(countof(scmaps)/2));
const Tscmap &scmap=scmaps[scmapidx+((flags&A52_LFE)?(countof(scmaps)/2):0)];
float *dst0,*dst;
dst0=dst=(float*)getDst(6*256*scmap.nchannels*sizeof(float));
int i=0;
for(; i<6 && a52_block(state)==0; i++) {
liba52::sample_t* samples=a52_samples(state);
for (int j=0; j<256; j++,samples++)
for (int ch=0; ch<scmap.nchannels; ch++) {
*dst++=float(*(samples+256*scmap.ch[ch])/level);
}
}
if (i==6) {
fmt.sf=TsampleFormat::SF_FLOAT32;
fmt.freq=sample_rate;
fmt.setChannels(scmap.nchannels,scmap.channelMask);
HRESULT hr=sinkA->deliverDecodedSample(dst0,6*256,fmt);
if (hr!=S_OK) {
return hr;
}
}
}
}
p+=size;
}
memmove(base,p,end-p);
end=base+(end-p);
p=base;
if (!enoughData) {
break;
}
} else {
p++;
}
}
src.resize(end-p);
return S_OK;
}
static GstFlowReturn
gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
GstA52Dec *a52dec;
gint channels, i;
gboolean need_reneg = FALSE;
gint chans;
gint length = 0, flags, sample_rate, bit_rate;
GstMapInfo map;
GstFlowReturn result = GST_FLOW_OK;
GstBuffer *outbuf;
const gint num_blocks = 6;
a52dec = GST_A52DEC (bdec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
/* parsed stuff already, so this should work out fine */
gst_buffer_map (buffer, &map, GST_MAP_READ);
g_assert (map.size >= 7);
/* re-obtain some sync header info,
* should be same as during _parse and could also be cached there,
* but anyway ... */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
length = a52_syncinfo (map.data, &flags, &sample_rate, &bit_rate);
g_assert (length == map.size);
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
if (a52dec->sample_rate != sample_rate) {
GST_DEBUG_OBJECT (a52dec, "sample rate changed");
need_reneg = TRUE;
a52dec->sample_rate = sample_rate;
}
if (flags) {
if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) {
GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update");
a52dec->flag_update = TRUE;
}
a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
}
if (bit_rate != a52dec->bit_rate) {
a52dec->bit_rate = bit_rate;
gst_a52dec_update_streaminfo (a52dec);
}
/* If we haven't had an explicit number of channels chosen through properties
* at this point, choose what to downmix to now, based on what the peer will
* accept - this allows a52dec to do downmixing in preference to a
* downstream element such as audioconvert.
*/
if (a52dec->request_channels != A52_CHANNEL) {
flags = a52dec->request_channels;
} else if (a52dec->flag_update) {
GstCaps *caps;
a52dec->flag_update = FALSE;
caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
gint orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6;
gint fixed_channels = 0;
const int a52_channels[6] = {
A52_MONO,
A52_STEREO,
A52_STEREO | A52_LFE,
A52_2F2R,
A52_2F2R | A52_LFE,
A52_3F2R | A52_LFE,
};
/* Prefer the original number of channels, but fixate to something
* preferred (first in the caps) downstream if possible.
*/
gst_structure_fixate_field_nearest_int (structure, "channels",
orig_channels);
if (gst_structure_get_int (structure, "channels", &fixed_channels)
&& fixed_channels <= 6) {
if (fixed_channels < orig_channels)
flags = a52_channels[fixed_channels - 1];
} else {
flags = a52_channels[5];
}
gst_caps_unref (copy);
} else if (flags)
flags = a52dec->stream_channels;
else
flags = A52_3F2R | A52_LFE;
if (caps)
gst_caps_unref (caps);
} else {
flags = a52dec->using_channels;
}
/* process */
flags |= A52_ADJUST_LEVEL;
a52dec->level = 1;
if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) {
gst_buffer_unmap (buffer, &map);
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
("a52_frame error"), result);
goto exit;
}
gst_buffer_unmap (buffer, &map);
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
if (a52dec->using_channels != channels) {
need_reneg = TRUE;
a52dec->using_channels = channels;
}
/* negotiate if required */
if (need_reneg) {
GST_DEBUG_OBJECT (a52dec,
"a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
if (!gst_a52dec_reneg (a52dec))
goto failed_negotiation;
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans)
goto invalid_flags;
/* handle decoded data;
* each frame has 6 blocks, one block is 256 samples, ea */
outbuf =
gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
{
guint8 *ptr = map.data;
for (i = 0; i < num_blocks; i++) {
if (a52_block (a52dec->state)) {
/* also marks discont */
GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
("error decoding block %d", i), result);
if (result != GST_FLOW_OK) {
gst_buffer_unmap (outbuf, &map);
goto exit;
}
} else {
gint n, c;
gint *reorder_map = a52dec->channel_reorder_map;
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) ptr)[n * chans + reorder_map[c]] =
a52dec->samples[c * 256 + n];
}
}
}
ptr += 256 * chans * (SAMPLE_WIDTH / 8);
}
}
gst_buffer_unmap (outbuf, &map);
result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
exit:
return result;
/* ERRORS */
failed_negotiation:
{
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
return GST_FLOW_ERROR;
}
invalid_flags:
{
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
("Invalid channel flags: %d", flags));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec;
guint8 *data;
guint size;
gint length = 0, flags, sample_rate, bit_rate;
GstFlowReturn result = GST_FLOW_OK;
a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
if (!a52dec->sent_segment) {
GstSegment segment;
/* Create a basic segment. Usually, we'll get a new-segment sent by
* another element that will know more information (a demuxer). If we're
* just looking at a raw AC3 stream, we won't - so we need to send one
* here, but we don't know much info, so just send a minimal TIME
* new-segment event
*/
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_push_event (a52dec->srcpad, gst_event_new_new_segment (FALSE,
segment.rate, segment.format, segment.start,
segment.duration, segment.start));
a52dec->sent_segment = TRUE;
}
/* merge with cache, if any. Also make sure timestamps match */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (a52dec,
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
if (a52dec->cache) {
buf = gst_buffer_join (a52dec->cache, buf);
a52dec->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* no sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: %d", length);
if (flags != a52dec->prev_flags)
a52dec->flag_update = TRUE;
a52dec->prev_flags = flags;
result = gst_a52dec_handle_frame (a52dec, data,
length, flags, sample_rate, bit_rate);
if (result != GST_FLOW_OK) {
size = 0;
break;
}
size -= length;
data += length;
} else {
/* not enough data */
GST_LOG ("Not enough data available");
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
a52dec->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
return result;
}
