ADM_AudiocodecAC3::ADM_AudiocodecAC3( uint32_t fourcc, WAVHeader *info,uint32_t extraLength,uint8_t *extraData)
: ADM_Audiocodec(fourcc)
{
int flags=0;
ADM_assert(fourcc==WAV_AC3);
ac3_handle=NULL;
ac3_sample=NULL;
#ifdef ADM_CPU_X86
#define CHK(x,y) if(CpuCaps::has##x()) flags|=MM_ACCEL_X86_##y;
CHK(MMX,MMX);
CHK(3DNOW,3DNOW);
CHK(MMXEXT,MMXEXT);
#endif
ac3_handle=(void *)a52_init(flags);
if(!ac3_handle)
{
printf("Cannot init a52\n");
ADM_assert(0);
}
ac3_sample=(sample_t *)a52_samples(AC3_HANDLE);
if(!ac3_sample)
{
printf("Cannot init a52 sample\n");
ADM_assert(0);
}
_downmix=0;
_wavHeader = info;
ADM_assert(_wavHeader);
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
sample_t level=a52_level, bias=384;
int flags=a52_flags|A52_ADJUST_LEVEL;
int i,len=-1;
if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
bias = 0;
if(!sh_audio->a_in_buffer_len)
if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
sh_audio->a_in_buffer_len=0;
if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
return len;
}
/* handle dynrng */
if (a52_drc_action != DRC_NO_ACTION) {
if (a52_drc_action == DRC_NO_COMPRESSION)
a52_dynrng(a52_state, NULL, NULL);
else
a52_dynrng(a52_state, dynrng_call, NULL);
}
len=0;
for (i = 0; i < 6; i++) {
if (a52_block (a52_state)){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
break;
}
len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]);
}
assert(len <= maxlen);
return len;
}
static GF_Err AC3_AttachStream(GF_BaseDecoder *ifcg, GF_ESD *esd)
{
A52CTX();
if (ctx->ES_ID && ctx->ES_ID!=esd->ESID) return GF_NOT_SUPPORTED;
GF_LOG(GF_LOG_DEBUG, GF_LOG_CODEC, ("[A52] Attaching stream %d\n", esd->ESID));
if (ctx->codec) a52_free(ctx->codec);
ctx->codec = a52_init(MM_ACCEL_DJBFFT);
if (!ctx->codec) {
GF_LOG(GF_LOG_ERROR, GF_LOG_CODEC, ("[A52] Error initializing decoder\n"));
return GF_IO_ERR;
}
ctx->samples = a52_samples(ctx->codec);
if (!ctx->samples) {
a52_free(ctx->codec);
GF_LOG(GF_LOG_ERROR, GF_LOG_CODEC, ("[A52] Error initializing decoder\n"));
return GF_IO_ERR;
}
ctx->num_channels = 0;
ctx->sample_rate = 0;
ctx->out_size = 0;
ctx->num_samples = 1536;
ctx->ES_ID = esd->ESID;
return GF_OK;
}
static int a52_decode_frame(unsigned char *buf, int maxlen,
struct frame_fmt *fmt)
{
sample_t level = 1, bias = 384;
int flags;
int i, len = -1;
if (!a52_buffer.a_in_buffer_len)
if (a52_fillbuff(&a52_buffer) < 0)
return len;
a52_buffer.a_in_buffer_len = 0;
flags = a52_flags | A52_ADJUST_LEVEL;
level *= gain;
if (a52_frame(a52_state, a52_buffer.a_in_buffer, &flags, &level, bias)) {
printf("a52_decode_frame a52: error decoding frame\n");
return len;
}
len = 0;
for (i = 0; i < 6; i++) {
if (a52_block(a52_state)) {
printf("a52: error at resampling\n");
break;
}
len +=
2 * a52_resample(a52_samples(a52_state),
(int16_t *) & buf[len]);
}
assert(len <= maxlen);
return len;
}
static GstStateChangeReturn
gst_a52dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstA52Dec *a52dec = GST_A52DEC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:{
GstA52DecClass *klass;
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
a52dec->state = a52_init (klass->a52_cpuflags);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->time = 0;
a52dec->sent_segment = FALSE;
a52dec->flag_update = TRUE;
gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
a52dec->samples = NULL;
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
clear_queued (a52dec);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
a52_free (a52dec->state);
a52dec->state = NULL;
break;
default:
break;
}
return ret;
}
mpeg3_ac3_t* mpeg3_new_ac3()
{
mpeg3_ac3_t *result = calloc(1, sizeof(mpeg3_ac3_t));
result->stream = mpeg3bits_new_stream(0, 0);
#ifdef A52_INIT_NEEDS_ARG
result->state = a52_init(0);
#else
result->state = a52_init();
#endif
result->output = a52_samples(result->state);
return result;
}
static gboolean
gst_a52dec_start (GstAudioDecoder * dec)
{
GstA52Dec *a52dec = GST_A52DEC (dec);
GstA52DecClass *klass;
GST_DEBUG_OBJECT (dec, "start");
klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
#if defined(A52_ACCEL_DETECT)
a52dec->state = a52_init ();
/* This line is just to avoid being accused of not using klass */
a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
#else
a52dec->state = a52_init (klass->a52_cpuflags);
#endif
if (!a52dec->state) {
GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
("failed to initialize a52 state"));
return FALSE;
}
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->flag_update = TRUE;
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_estimate_rate (dec, TRUE);
return TRUE;
}
/*****************************************************************************
* DoWork: decode an ATSC A/52 frame.
*****************************************************************************/
static void DoWork( filter_t * p_filter,
aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
{
filter_sys_t *p_sys = p_filter->p_sys;
#ifdef LIBA52_FIXED
sample_t i_sample_level = (1 << 24);
#else
sample_t i_sample_level = 1;
#endif
int i_flags = p_sys->i_flags;
int i_bytes_per_block = 256 * p_sys->i_nb_channels
* sizeof(sample_t);
int i;
/* Do the actual decoding now. */
a52_frame( p_sys->p_liba52, p_in_buf->p_buffer,
&i_flags, &i_sample_level, 0 );
if ( (i_flags & A52_CHANNEL_MASK) != (p_sys->i_flags & A52_CHANNEL_MASK)
&& !p_sys->b_dontwarn )
{
msg_Warn( p_filter,
"liba52 couldn't do the requested downmix 0x%x->0x%x",
p_sys->i_flags & A52_CHANNEL_MASK,
i_flags & A52_CHANNEL_MASK );
p_sys->b_dontwarn = 1;
}
if( !p_sys->b_dynrng )
{
a52_dynrng( p_sys->p_liba52, NULL, NULL );
}
for ( i = 0; i < 6; i++ )
{
sample_t * p_samples;
if( a52_block( p_sys->p_liba52 ) )
{
msg_Warn( p_filter, "a52_block failed for block %d", i );
}
p_samples = a52_samples( p_sys->p_liba52 );
if ( ((p_sys->i_flags & A52_CHANNEL_MASK) == A52_CHANNEL1
|| (p_sys->i_flags & A52_CHANNEL_MASK) == A52_CHANNEL2
|| (p_sys->i_flags & A52_CHANNEL_MASK) == A52_MONO)
&& (p_filter->fmt_out.audio.i_physical_channels
& (AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT)) )
{
Duplicate( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples );
}
else if ( p_filter->fmt_out.audio.i_original_channels
& AOUT_CHAN_REVERSESTEREO )
{
Exchange( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples );
}
else
{
/* Interleave the *$£%ù samples. */
Interleave( (sample_t *)(p_out_buf->p_buffer + i * i_bytes_per_block),
p_samples, p_sys->i_nb_channels, p_sys->pi_chan_table);
}
}
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_buffer = i_bytes_per_block * 6;
}
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;
}
}
}
}
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;
}
