X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fdeca52.c;h=0958a9139beb00b871a7c30e36b5ae7e02f5de22;hb=53540a2eb047671341ba18d82331c1352908279c;hp=94bf8a5405d7e2602e21fea4a096ffb1f91f3ec2;hpb=6e499d70fd7147fec222062679e24550a4881892;p=handbrake-jp%2Fhandbrake-jp-git.git
diff --git a/libhb/deca52.c b/libhb/deca52.c
index 94bf8a54..0958a913 100644
--- a/libhb/deca52.c
+++ b/libhb/deca52.c
@@ -1,12 +1,13 @@
/* $Id: deca52.c,v 1.14 2005/03/03 17:21:57 titer Exp $
This file is part of the HandBrake source code.
- Homepage: .
+ Homepage: .
It may be used under the terms of the GNU General Public License. */
#include "hb.h"
#include "a52dec/a52.h"
+#include "libavutil/crc.h"
struct hb_work_private_s
{
@@ -19,27 +20,25 @@ struct hb_work_private_s
int flags_out;
int rate;
int bitrate;
- float level;
-
+ int out_discrete_channels;
int error;
- int sync;
- int size;
-
- int64_t next_expected_pts;
-
- int64_t sequence;
-
+ int frames; // number of good frames decoded
+ int crc_errors; // number of frames with crc errors
+ int bytes_dropped; // total bytes dropped while resyncing
+ float level;
+ float dynamic_range_compression;
+ double next_expected_pts;
+ int64_t last_buf_pts;
+ hb_list_t *list;
+ const AVCRC *crc_table;
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 =
{
@@ -47,7 +46,9 @@ hb_work_object_t hb_deca52 =
"AC3 decoder",
deca52Init,
deca52Work,
- deca52Close
+ deca52Close,
+ 0,
+ deca52BSInfo
};
/***********************************************************************
@@ -56,33 +57,58 @@ hb_work_object_t hb_deca52 =
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;
+ pv->crc_table = av_crc_get_table( AV_CRC_16_ANSI );
pv->list = hb_list_init();
pv->state = a52_init( 0 );
- /* 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);
+ /* Decide what format we want out of a52dec
+ work.c has already done some of this deduction for us in do_job() */
- /* 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->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(audio->config.out.mixdown);
pv->level = 32768.0;
- pv->next_expected_pts = 0;
- pv->sequence = 0;
-
+ pv->dynamic_range_compression = audio->config.out.dynamic_range_compression;
+
return 0;
}
@@ -91,9 +117,15 @@ int deca52Init( hb_work_object_t * w, hb_job_t * job )
***********************************************************************
* 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;
+
+ if ( pv->crc_errors )
+ {
+ hb_log( "deca52: %d frames decoded, %d crc errors, %d bytes dropped",
+ pv->frames, pv->crc_errors, pv->bytes_dropped );
+ }
a52_free( pv->state );
hb_list_empty( &pv->list );
free( pv );
@@ -106,15 +138,25 @@ void deca52Close( hb_work_object_t * w )
* 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 && *buf_in )
+ 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;
+ }
+
+ if ( (*buf_in)->start < -1 && pv->next_expected_pts == 0 )
{
- pv->sequence = (*buf_in)->sequence;
+ // discard buffers that start before video time 0
+ *buf_out = NULL;
+ return HB_WORK_OK;
}
hb_list_add( pv->list, *buf_in );
@@ -123,8 +165,7 @@ int deca52Work( hb_work_object_t * w, hb_buffer_t ** buf_in,
/* If we got more than a frame, chain raw buffers */
*buf_out = buf = Decode( w );
while( buf )
- {
- buf->sequence = pv->sequence;
+ {
buf->next = Decode( w );
buf = buf->next;
}
@@ -135,88 +176,122 @@ int deca52Work( hb_work_object_t * w, hb_buffer_t ** buf_in,
/***********************************************************************
* 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;
+ int size = 0;
- /* Get a frame header if don't have one yet */
- if( !pv->sync )
+ // check that we're at the start of a valid frame and align to the
+ // start of a valid frame if we're not.
+ // we have to check the header & crc so we need at least
+ // 7 (the header size) + 128 (the minimum frame size) bytes
+ while( hb_list_bytes( pv->list ) >= 7+128 )
{
- while( hb_list_bytes( pv->list ) >= 7 )
+ /* check if this is a valid header */
+ hb_list_seebytes( pv->list, pv->frame, 7 );
+ size = a52_syncinfo( pv->frame, &pv->flags_in, &pv->rate, &pv->bitrate );
+ if ( size > 0 )
{
- /* We have 7 bytes, check if this is a correct header */
- hb_list_seebytes( pv->list, pv->frame, 7 );
- pv->size = a52_syncinfo( pv->frame, &pv->flags_in, &pv->rate,
- &pv->bitrate );
- if( pv->size )
+ // header looks valid - check the crc1
+ if( size > hb_list_bytes( pv->list ) )
+ {
+ // don't have all the frame's data yet
+ return NULL;
+ }
+ int crc1size = (size >> 1) + (size >> 3);
+ hb_list_seebytes( pv->list, pv->frame, crc1size );
+ if ( av_crc( pv->crc_table, 0, pv->frame + 2, crc1size - 2 ) == 0 )
{
- /* It is. W00t. */
+ // crc1 is ok - say we have valid frame sync
if( pv->error )
{
- hb_log( "a52_syncinfo ok" );
+ hb_log( "output track %d: ac3 in sync after skipping %d bytes",
+ audio->config.out.track, pv->error );
+ pv->bytes_dropped += pv->error;
+ pv->error = 0;
}
- pv->error = 0;
- pv->sync = 1;
break;
}
-
- /* It is not */
- if( !pv->error )
- {
- hb_log( "a52_syncinfo failed" );
- pv->error = 1;
- }
-
- /* Try one byte later */
- hb_list_getbytes( pv->list, pv->frame, 1, NULL, NULL );
}
+ // no sync - discard one byte then try again
+ hb_list_getbytes( pv->list, pv->frame, 1, NULL, NULL );
+ ++pv->error;
}
- if( !pv->sync ||
- hb_list_bytes( pv->list ) < pv->size )
+ // we exit the above loop either in error state (we didn't find sync
+ // or don't have enough data yet to validate sync) or in sync. If we're
+ // not in sync we need more data so just return.
+ if( pv->error || size <= 0 || hb_list_bytes( pv->list ) < size )
{
/* Need more data */
return NULL;
}
- /* Get the whole frame */
- hb_list_getbytes( pv->list, pv->frame, pv->size, &pts, &pos );
+ // Get the whole frame and check its CRC. If the CRC is wrong
+ // discard the frame - we'll resync on the next call.
+
+ uint64_t ipts;
+ hb_list_getbytes( pv->list, pv->frame, size, &ipts, NULL );
+ if ( av_crc( pv->crc_table, 0, pv->frame + 2, size - 2 ) != 0 )
+ {
+ ++pv->crc_errors;
+ return NULL;
+ }
+ ++pv->frames;
+ if ( ipts != pv->last_buf_pts )
+ {
+ pv->last_buf_pts = ipts;
+ }
+ else
+ {
+ // spec says that the PTS is the start time of the first frame
+ // that starts in the PES frame so we only use the PTS once then
+ // get the following frames' PTS from the frame length.
+ ipts = -1;
+ }
+
+ double pts = ( ipts != -1 ) ? ipts : pv->next_expected_pts;
+ double frame_dur = (6. * 256. * 90000.) / pv->rate;
/* AC3 passthrough: don't decode the AC3 frame */
- if( pv->job->acodec & HB_ACODEC_AC3 )
+ if( audio->config.out.codec == HB_ACODEC_AC3_PASS )
{
- 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->sync = 0;
+ buf = hb_buffer_init( size );
+ memcpy( buf->data, pv->frame, size );
+ buf->start = pts;
+ pts += frame_dur;
+ buf->stop = pts;
+ pv->next_expected_pts = pts;
return buf;
}
/* Feed liba52 */
a52_frame( pv->state, pv->frame, &pv->flags_out, &pv->level, 0 );
- /* 6 blocks per frame, 256 samples per block, channelsused channels */
- buf = hb_buffer_init( 6 * 256 * pv->out_discrete_channels * sizeof( float ) );
- if (pts == -1)
+ /* If a user specifies strong dynamic range compression (>1), adjust it.
+ If a user specifies default dynamic range compression (1), leave it alone.
+ If a user specifies no dynamic range compression (0), call a null function. */
+ if( pv->dynamic_range_compression > 1.0 )
+ {
+ a52_dynrng( pv->state, dynrng_call, &pv->dynamic_range_compression );
+ }
+ else if( !pv->dynamic_range_compression )
{
- pts = pv->next_expected_pts;
+ a52_dynrng( pv->state, NULL, NULL );
}
- 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;
-
+ /* 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;
+ pts += frame_dur;
+ buf->stop = pts;
+ pv->next_expected_pts = pts;
+
for( i = 0; i < 6; i++ )
{
sample_t * samples_in;
@@ -229,15 +304,165 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
/* Interleave */
for( j = 0; j < 256; j++ )
{
- for ( k = 0; k < pv->out_discrete_channels; k++ )
- {
- samples_out[(pv->out_discrete_channels*j)+k] = samples_in[(256*k)+j];
- }
+ for ( k = 0; k < pv->out_discrete_channels; k++ )
+ {
+ samples_out[(pv->out_discrete_channels*j)+k] = samples_in[(256*k)+j];
+ }
}
}
-
- pv->sync = 0;
return buf;
}
+static int find_sync( const uint8_t *buf, int len )
+{
+ int i;
+
+ // since AC3 frames don't line up with MPEG ES frames scan the
+ // frame for an AC3 sync pattern.
+ for ( i = 0; i < len - 16; ++i )
+ {
+ int rate, bitrate, flags;
+ int size = a52_syncinfo( (uint8_t *)buf + i, &flags, &rate, &bitrate );
+ if( size > 0 )
+ {
+ // we have a plausible sync header - see if crc1 checks
+ int crc1size = (size >> 1) + (size >> 3);
+ if ( i + crc1size > len )
+ {
+ // don't have enough data to check crc1
+ break;
+ }
+ if ( av_crc( av_crc_get_table( AV_CRC_16_ANSI ), 0,
+ buf + i + 2, crc1size - 2 ) == 0 )
+ {
+ // crc checks - we've got sync
+ return i;
+ }
+ }
+ }
+ return -1;
+}
+
+static int deca52BSInfo( hb_work_object_t *w, const hb_buffer_t *b,
+ hb_work_info_t *info )
+{
+ memset( info, 0, sizeof(*info) );
+
+ // We don't know if the way that AC3 frames are fragmented into whatever
+ // packetization the container uses will give us enough bytes per fragment
+ // to check the CRC (we need at least 5/8 of the the frame). So we
+ // copy the fragment we got into an accumulation buffer in the audio object
+ // then look for sync over all the frags we've accumulated so far.
+ uint8_t *buf = w->audio->priv.config.a52.buf;
+ int len = w->audio->priv.config.a52.len, blen = b->size;
+ if ( len + blen > sizeof(w->audio->priv.config.a52.buf) )
+ {
+ // we don't have enough empty space in the accumulation buffer to
+ // hold the new frag - make room for it by discarding the oldest data.
+ if ( blen >= sizeof(w->audio->priv.config.a52.buf) )
+ {
+ // the frag is bigger than our accumulation buffer - copy all
+ // that will fit (the excess doesn't matter since the buffer
+ // is many times the size of a max length ac3 frame).
+ blen = sizeof(w->audio->priv.config.a52.buf);
+ len = 0;
+ }
+ else
+ {
+ // discard enough bytes from the front of the buffer to make
+ // room for the new stuff
+ int newlen = sizeof(w->audio->priv.config.a52.buf) - blen;
+ memcpy( buf, buf + len - newlen, newlen );
+ len = newlen;
+ }
+ }
+ // add the new frag to the buffer
+ memcpy( buf+len, b->data, blen );
+ len += blen;
+
+ int i;
+ if ( ( i = find_sync( buf, len ) ) < 0 )
+ {
+ // didn't find sync - wait for more data
+ w->audio->priv.config.a52.len = len;
+ return 0;
+ }
+
+ // got sync - extract and canoncalize the bitstream parameters
+ int rate = 0, bitrate = 0, flags = 0;
+ uint8_t raw = buf[i + 5];
+ a52_syncinfo( buf + i, &flags, &rate, &bitrate );
+
+ if ( rate == 0 || bitrate == 0 )
+ {
+ // invalid AC-3 parameters - toss what we have so we'll start over
+ // with the next buf otherwise we'll keep syncing on this junk.
+ w->audio->priv.config.a52.len = 0;
+ return 0;
+ }
+
+ // bsid | bsmod | acmod | cmixlv | surmixlv | dsurmod | lfeon | dialnorm | compre
+ // 5 3 3 2 2 2 1 5 1
+ // byte1 | byte2 | byte3
+
+ info->name = "AC-3";
+ info->rate = rate;
+ info->rate_base = 1;
+ info->bitrate = bitrate;
+ info->flags = flags;
+ info->version = raw >> 3; /* bsid is the first 5 bits */
+ info->mode = raw & 0x7; /* bsmod is the following 3 bits */
+
+ 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;
+}