libhb: fix problem with incorrect addition of audio silence

sync needs the precise duration of a frame of audio and has
no way to compute it for passthru audio.  So add a duration
member to hb_buffer_t and set it in the audio decoders.

This should eventually supersede hb_buffer_t member "stop"
since it provides redundant information.


git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4618 b64f7644-9d1e-0410-96f1-a4d463321fa5

6 files changed, 21 insertions, 3 deletions

diff --git a/libhb/deca52.c b/libhb/deca52.c
index 1f05cc829..09eb88101 100644
--- a/libhb/deca52.c
+++ b/libhb/deca52.c
@@ -265,6 +265,7 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
         buf = hb_buffer_init( size );
         memcpy( buf->data, pv->frame, size );
         buf->s.start = pts;
+        buf->s.duration = frame_dur;
         pts += frame_dur;
         buf->s.stop  = pts;
         pv->next_expected_pts = pts;
@@ -289,6 +290,7 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
     /* 6 blocks per frame, 256 samples per block, channelsused channels */
     buf        = hb_buffer_init( 6 * 256 * pv->out_discrete_channels * sizeof( float ) );
     buf->s.start = pts;
+    buf->s.duration = frame_dur;
     pts += frame_dur;
     buf->s.stop  = pts;
     pv->next_expected_pts = pts;
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c
index bb7c4c3c3..45239a29c 100644
--- a/libhb/decavcodec.c
+++ b/libhb/decavcodec.c
@@ -1523,7 +1523,8 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
         {
             int isamp = av_get_bytes_per_sample( context->sample_fmt );
             nsamples = out_size / isamp;
-            double pts_next = pv->pts_next + nsamples * pv->duration;
+            double duration = nsamples * pv->duration;
+            double pts_next = pv->pts_next + duration;
 
             // DTS-HD can be passed through to mkv
             if( audio->config.out.codec & HB_ACODEC_PASS_FLAG )
@@ -1536,6 +1537,7 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
                 buf = hb_buffer_init( avp.size );
                 memcpy( buf->data, avp.data, avp.size );
                 buf->s.start = pv->pts_next;
+                buf->s.duration = duration;
                 buf->s.stop  = pts_next;
                 hb_list_add( pv->list, buf );
                 pv->pts_next = pts_next;
@@ -1576,6 +1578,7 @@ static void decodeAudio( hb_audio_t * audio, hb_work_private_t *pv, uint8_t *dat
             hb_buffer_t * buf;
             buf = downmixAudio( audio, pv, buffer, context->channels, nsamples );
             buf->s.start = pv->pts_next;
+            buf->s.duration = duration;
             buf->s.stop = pts_next;
             hb_list_add( pv->list, buf );
 
diff --git a/libhb/decdca.c b/libhb/decdca.c
index 5bc118007..49fb3ecae 100644
--- a/libhb/decdca.c
+++ b/libhb/decdca.c
@@ -235,6 +235,7 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
         buf = hb_buffer_init( pv->size );
         memcpy( buf->data, pv->frame, pv->size );
         buf->s.start = pts;
+        buf->s.duration = frame_dur;
         pv->next_pts = pts + frame_dur;
         buf->s.stop  = pv->next_pts;
         pv->sync = 0;
@@ -249,10 +250,12 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
 
     /* num_blocks blocks per frame, 256 samples per block, channelsused channels */
     int nsamp = num_blocks * 256;
+    frame_dur = (double)nsamp / (double)pv->rate * 90000.;
     buf = hb_buffer_init( nsamp * pv->out_discrete_channels * sizeof( float ) );
 
     buf->s.start = pts;
-    pv->next_pts = pts + (double)nsamp / (double)pv->rate * 90000.;
+    buf->s.duration = frame_dur;
+    pv->next_pts = pts + frame_dur;
     buf->s.stop  = pv->next_pts;
 
     for( i = 0; i < num_blocks; i++ )
diff --git a/libhb/declpcm.c b/libhb/declpcm.c
index 6a7cbe56c..98ecc7fc7 100644
--- a/libhb/declpcm.c
+++ b/libhb/declpcm.c
@@ -219,6 +219,7 @@ static hb_buffer_t *Decode( hb_work_object_t *w )
     out = hb_buffer_init( pv->samples * sizeof( float ) );
 
     out->s.start  = pv->next_pts;
+    out->s.duration = pv->duration;
     pv->next_pts += pv->duration;
     out->s.stop = pv->next_pts;
 
diff --git a/libhb/internal.h b/libhb/internal.h
index 0d6ee2776..4c2c6e2a9 100644
--- a/libhb/internal.h
+++ b/libhb/internal.h
@@ -73,6 +73,7 @@ struct hb_buffer_s
 
         int           id;           // ID of the track that the packet comes from
         int64_t       start;        // start time of frame
+        double        duration;     // Actual duration, may be fractional ticks
         int64_t       stop;         // stop time of frame
         int64_t       renderOffset; // DTS used by b-frame offsets in muxmp4
         int64_t       pcr;
diff --git a/libhb/sync.c b/libhb/sync.c
index c76cc45d2..1a10e5e5e 100644
--- a/libhb/sync.c
+++ b/libhb/sync.c
@@ -1050,7 +1050,15 @@ static hb_buffer_t * OutputAudioFrame( hb_audio_t *audio, hb_buffer_t *buf,
                                        hb_sync_audio_t *sync )
 {
     int64_t start = (int64_t)sync->next_start;
-    double duration = buf->s.stop - buf->s.start;
+
+    // Can't count of buf->s.stop - buf->s.start for accurate duration
+    // due to integer rounding, so use buf->s.duration when it is set
+    // (which should be always if I didn't miss anything)
+    double duration;
+    if ( buf->s.duration > 0 )
+        duration = buf->s.duration;
+    else
+        duration = buf->s.stop - buf->s.start;
 
     if ( !( audio->config.out.codec & HB_ACODEC_PASS_FLAG ) )
     {