/* $Id: deca52.c,v 1.14 2005/03/03 17:21:57 titer Exp $
This file is part of the HandBrake source code.
- Homepage: <http://handbrake.m0k.org/>.
+ Homepage: <http://handbrake.fr/>.
It may be used under the terms of the GNU General Public License. */
#include "hb.h"
int rate;
int bitrate;
float level;
-
+ float dynamic_range_compression;
+
int error;
int sync;
int size;
+ int64_t next_expected_pts;
+
+ int64_t sequence;
+
uint8_t frame[3840];
hb_list_t * list;
-
+
int out_discrete_channels;
-
+
};
-int deca52Init( hb_work_object_t *, hb_job_t * );
-int deca52Work( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
-void deca52Close( hb_work_object_t * );
+static int deca52Init( hb_work_object_t *, hb_job_t * );
+static int deca52Work( hb_work_object_t *, hb_buffer_t **, hb_buffer_t ** );
+static void deca52Close( hb_work_object_t * );
+static int deca52BSInfo( hb_work_object_t * , const hb_buffer_t *,
+ hb_work_info_t * );
hb_work_object_t hb_deca52 =
{
"AC3 decoder",
deca52Init,
deca52Work,
- deca52Close
+ deca52Close,
+ 0,
+ deca52BSInfo
};
/***********************************************************************
static hb_buffer_t * Decode( hb_work_object_t * w );
/***********************************************************************
+ * dynrng_call
+ ***********************************************************************
+ * Boosts soft audio -- taken from gbooker's work in A52Decoder, comment and all..
+ * Two cases
+ * 1) The user requested a compression of 1 or less, return the typical power rule
+ * 2) The user requested a compression of more than 1 (decompression):
+ * If the stream's requested compression is less than 1.0 (loud sound), return the normal compression
+ * If the stream's requested compression is more than 1.0 (soft sound), use power rule (which will make
+ * it louder in this case).
+ *
+ **********************************************************************/
+static sample_t dynrng_call (sample_t c, void *data)
+{
+ float *level = (float *)data;
+ float levelToUse = (float)*level;
+ if(c > 1.0 || levelToUse <= 1.0)
+ {
+ return powf(c, levelToUse);
+ }
+ else
+ return c;
+}
+
+/***********************************************************************
* hb_work_deca52_init
***********************************************************************
* Allocate the work object, initialize liba52
**********************************************************************/
-int deca52Init( hb_work_object_t * w, hb_job_t * job )
+static int deca52Init( hb_work_object_t * w, hb_job_t * job )
{
hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
+ hb_audio_t * audio = w->audio;
w->private_data = pv;
pv->job = job;
/* Decide what format we want out of a52dec
work.c has already done some of this deduction for us in do_job() */
-
- pv->flags_out = HB_AMIXDOWN_GET_A52_FORMAT(w->amixdown);
+
+ pv->flags_out = HB_AMIXDOWN_GET_A52_FORMAT(audio->config.out.mixdown);
/* pass the number of channels used into the private work data */
/* will only be actually used if we're not doing AC3 passthru */
- pv->out_discrete_channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(w->amixdown);
+ pv->out_discrete_channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
pv->level = 32768.0;
+ pv->dynamic_range_compression = audio->config.out.dynamic_range_compression;
+
+ pv->next_expected_pts = 0;
+ pv->sequence = 0;
return 0;
}
***********************************************************************
* Free memory
**********************************************************************/
-void deca52Close( hb_work_object_t * w )
+static void deca52Close( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
a52_free( pv->state );
* Add the given buffer to the data we already have, and decode as much
* as we can
**********************************************************************/
-int deca52Work( hb_work_object_t * w, hb_buffer_t ** buf_in,
+static int deca52Work( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * buf;
+ if ( (*buf_in)->size <= 0 )
+ {
+ /* EOF on input stream - send it downstream & say that we're done */
+ *buf_out = *buf_in;
+ *buf_in = NULL;
+ return HB_WORK_DONE;
+ }
+
+ pv->sequence = (*buf_in)->sequence;
+
hb_list_add( pv->list, *buf_in );
*buf_in = NULL;
*buf_out = buf = Decode( w );
while( buf )
{
+ buf->sequence = pv->sequence;
buf->next = Decode( w );
buf = buf->next;
}
/***********************************************************************
* Decode
***********************************************************************
- *
+ *
**********************************************************************/
static hb_buffer_t * Decode( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * buf;
+ hb_audio_t * audio = w->audio;
int i, j, k;
uint64_t pts, pos;
/* Get the whole frame */
hb_list_getbytes( pv->list, pv->frame, pv->size, &pts, &pos );
+ if (pts == -1)
+ {
+ pts = pv->next_expected_pts;
+ }
/* AC3 passthrough: don't decode the AC3 frame */
- if( pv->job->acodec & HB_ACODEC_AC3 )
+ if( audio->config.out.codec == HB_ACODEC_AC3 )
{
buf = hb_buffer_init( pv->size );
memcpy( buf->data, pv->frame, pv->size );
buf->start = pts + ( pos / pv->size ) * 6 * 256 * 90000 / pv->rate;
buf->stop = buf->start + 6 * 256 * 90000 / pv->rate;
+ pv->next_expected_pts = buf->stop;
pv->sync = 0;
return buf;
}
/* Feed liba52 */
a52_frame( pv->state, pv->frame, &pv->flags_out, &pv->level, 0 );
+ if ( pv->dynamic_range_compression > 1.0 )
+ {
+ a52_dynrng( pv->state, dynrng_call, &pv->dynamic_range_compression);
+ }
+
/* 6 blocks per frame, 256 samples per block, channelsused channels */
buf = hb_buffer_init( 6 * 256 * pv->out_discrete_channels * sizeof( float ) );
buf->start = pts + ( pos / pv->size ) * 6 * 256 * 90000 / pv->rate;
buf->stop = buf->start + 6 * 256 * 90000 / pv->rate;
+ /*
+ * To track AC3 PTS add this back in again.
+ *hb_log("AC3: pts is %lld, buf->start %lld buf->stop %lld", pts, buf->start, buf->stop);
+ */
+
+ pv->next_expected_pts = buf->stop;
+
for( i = 0; i < 6; i++ )
{
sample_t * samples_in;
return buf;
}
+static int deca52BSInfo( hb_work_object_t *w, const hb_buffer_t *b,
+ hb_work_info_t *info )
+{
+ int i, rate, bitrate, flags;
+
+ memset( info, 0, sizeof(*info) );
+
+ /* since AC3 frames don't line up with MPEG ES frames scan the
+ * entire frame for an AC3 sync pattern. */
+ for ( i = 0; i < b->size - 7; ++i )
+ {
+ if( a52_syncinfo( &b->data[i], &flags, &rate, &bitrate ) != 0 )
+ {
+ break;
+ }
+ }
+ if ( i >= b->size - 7 )
+ {
+ /* didn't find AC3 sync */
+ return 0;
+ }
+
+ info->name = "AC-3";
+ info->rate = rate;
+ info->rate_base = 1;
+ info->bitrate = bitrate;
+ info->flags = flags;
+ if ( (flags & A52_CHANNEL_MASK) == A52_DOLBY )
+ {
+ info->flags |= AUDIO_F_DOLBY;
+ }
+
+ switch( flags & A52_CHANNEL_MASK )
+ {
+ /* mono sources */
+ case A52_MONO:
+ case A52_CHANNEL1:
+ case A52_CHANNEL2:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_MONO;
+ break;
+ /* stereo input */
+ case A52_CHANNEL:
+ case A52_STEREO:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_STEREO;
+ break;
+ /* dolby (DPL1 aka Dolby Surround = 4.0 matrix-encoded) input */
+ case A52_DOLBY:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_DOLBY;
+ break;
+ /* 3F/2R input */
+ case A52_3F2R:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_3F2R;
+ break;
+ /* 3F/1R input */
+ case A52_3F1R:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_3F1R;
+ break;
+ /* other inputs */
+ case A52_3F:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_3F;
+ break;
+ case A52_2F1R:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_2F1R;
+ break;
+ case A52_2F2R:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_2F2R;
+ break;
+ /* unknown */
+ default:
+ info->channel_layout = HB_INPUT_CH_LAYOUT_STEREO;
+ }
+
+ if (flags & A52_LFE)
+ {
+ info->channel_layout |= HB_INPUT_CH_LAYOUT_HAS_LFE;
+ }
+
+ return 1;
+}
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