Enable DTS passthru for matroska container

since we have no way to create silent dts frames for insertion to
maintain sync, frames from other streams may be dropped when there is
a sync problem on the DTS stream

git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@2339 b64f7644-9d1e-0410-96f1-a4d463321fa5

diff --git a/gtk/src/hb-backend.c b/gtk/src/hb-backend.c
index ce4225a995e7d8489261912f4e2e660192c781f4..94e5c216bbff375d16a9bf34409ca0c8ffe6513a 100644 (file)
--- a/gtk/src/hb-backend.c
+++ b/gtk/src/hb-backend.c
@@ -196,7 +196,7 @@ static options_map_t d_acodec_opts[] =
        {"MP3 (lame)",      "lame",   HB_ACODEC_LAME, "lame"},
        {"Vorbis",          "vorbis", HB_ACODEC_VORBIS, "vorbis"},
        {"AC3 (pass-thru)", "ac3",    HB_ACODEC_AC3, "ac3"},
-//     {"DTS (pass-thru)", "dts",    HB_ACODEC_DCA, "dts"},
+       {"DTS (pass-thru)", "dts",    HB_ACODEC_DCA, "dts"},
 };
 combo_opts_t acodec_opts =
 {

diff --git a/libhb/common.c b/libhb/common.c
index 0ddcccef576384690b08f411d42ac71ef8961591..88a8ea8ec07e4e0d98e26a1b33db9db208952a9a 100644 (file)
--- a/libhb/common.c
+++ b/libhb/common.c
@@ -234,6 +234,7 @@ int hb_calc_bitrate( hb_job_t * job, int size )
                 samples_per_frame = 1152;
                 break;
             case HB_ACODEC_AC3:
+            case HB_ACODEC_DCA:
                 samples_per_frame = 1536;
                 break;
             default:

diff --git a/libhb/decdca.c b/libhb/decdca.c
index 9d1bde71c3041f9b1cafcc025ccf9e524399ab7b..ec625c5bd6726fe049a0d62a7d0902046fd9922e 100644 (file)
--- a/libhb/decdca.c
+++ b/libhb/decdca.c
@@ -153,6 +153,7 @@ 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;
     int64_t       pts, pos;
     int           num_blocks;
@@ -210,16 +211,6 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
         pts = -1;
     }
 
-    /* Feed libdca */
-    dca_frame( pv->state, pv->frame, &pv->flags_out, &pv->level, 0 );
-
-    /* find out how many blocks are in this frame */
-    num_blocks = dca_blocks_num( pv->state );
-
-    /* num_blocks blocks per frame, 256 samples per block, channelsused channels */
-    int nsamp = num_blocks * 256;
-    buf = hb_buffer_init( nsamp * pv->out_discrete_channels * sizeof( float ) );
-
     // mkv files typically use a 1ms timebase which results in a lot of
     // truncation error in their timestamps. Also, TSMuxer or something
     // in the m2ts-to-mkv toolchain seems to take a very casual attitude
@@ -230,6 +221,31 @@ static hb_buffer_t * Decode( hb_work_object_t * w )
     {
         pts = pv->next_pts;
     }
+
+    double frame_dur = (double)(pv->frame_length & ~0xFF) / (double)pv->rate * 90000.;
+
+    /* DCA passthrough: don't decode the DCA frame */
+    if( audio->config.out.codec == HB_ACODEC_DCA )
+    {
+        buf = hb_buffer_init( pv->size );
+        memcpy( buf->data, pv->frame, pv->size );
+        buf->start = pts;
+        pv->next_pts = pts + frame_dur;
+        buf->stop  = pv->next_pts;
+        pv->sync = 0;
+        return buf;
+    }
+
+    /* Feed libdca */
+    dca_frame( pv->state, pv->frame, &pv->flags_out, &pv->level, 0 );
+
+    /* find out how many blocks are in this frame */
+    num_blocks = dca_blocks_num( pv->state );
+
+    /* num_blocks blocks per frame, 256 samples per block, channelsused channels */
+    int nsamp = num_blocks * 256;
+    buf = hb_buffer_init( nsamp * pv->out_discrete_channels * sizeof( float ) );
+
     buf->start = pts;
     pv->next_pts = pts + (double)nsamp / (double)pv->rate * 90000.;
     buf->stop  = pv->next_pts;

diff --git a/libhb/muxavi.c b/libhb/muxavi.c
index 913e84508fa6cce52c8056f5373e336dd0c608f4..4838ee1330e80b502546be0fa7e574ae48878b90 100644 (file)
--- a/libhb/muxavi.c
+++ b/libhb/muxavi.c
@@ -332,6 +332,7 @@ static int AVIInit( hb_mux_object_t * m )
     hb_mux_data_t * mux_data;
 
     int audio_count = hb_list_count( title->list_audio );
+    int is_passthru = 0;
     int is_ac3      = 0;
     int hdrl_bytes;
     int i;
@@ -427,6 +428,8 @@ static int AVIInit( hb_mux_object_t * m )
         audio = hb_list_item( title->list_audio, i );
 
         is_ac3 = (audio->config.out.codec == HB_ACODEC_AC3);
+        is_passthru = (audio->config.out.codec == HB_ACODEC_AC3) ||
+                      (audio->config.out.codec == HB_ACODEC_DCA);
 
         mux_data = calloc( sizeof( hb_mux_data_t ), 1 );
         audio->priv.mux_data = mux_data;
@@ -440,17 +443,17 @@ static int AVIInit( hb_mux_object_t * m )
         h.Type          = FOURCC( "auds" );
         h.InitialFrames = 1;
         h.Scale         = 1;
-        h.Rate          = is_ac3 ? ( audio->config.in.bitrate / 8 ) :
+        h.Rate          = is_passthru ? ( audio->config.in.bitrate / 8 ) :
                                    ( audio->config.out.bitrate * 1000 / 8 );
         h.Quality       = 0xFFFFFFFF;
         h.SampleSize    = 1;
 
         /* Audio stream format */
         f.FourCC         = FOURCC( "strf" );
-        if( is_ac3 )
+        if( is_passthru )
         {
             f.BytesCount     = sizeof( hb_wave_formatex_t ) - 8;
-            f.FormatTag      = 0x2000;
+            f.FormatTag      = is_ac3 ? 0x2000 : 0x2001;
             f.Channels       = HB_INPUT_CH_LAYOUT_GET_DISCRETE_COUNT(audio->config.in.channel_layout);
             f.SamplesPerSec  = audio->config.in.samplerate;
         }
@@ -464,7 +467,7 @@ static int AVIInit( hb_mux_object_t * m )
         }
         f.AvgBytesPerSec = h.Rate;
         f.BlockAlign     = 1;
-        if( is_ac3 )
+        if( is_passthru )
         {
             f.Size       = 0;
         }
@@ -493,7 +496,7 @@ static int AVIInit( hb_mux_object_t * m )
         ( job->anamorphic.mode ? sizeof( hb_avi_vprp_info_t ) : 0 ) +
         /* audios strf */
         audio_count * ( sizeof( hb_wave_formatex_t ) +
-                        ( is_ac3 ? 0 : sizeof( hb_wave_mp3_t ) ) );
+                        ( is_passthru ? 0 : sizeof( hb_wave_mp3_t ) ) );
 
     /* Here we really start to write into the file */
 
@@ -537,11 +540,11 @@ static int AVIInit( hb_mux_object_t * m )
         WriteInt32( m->file, FOURCC( "LIST" ) );
         WriteInt32( m->file, 4 + sizeof( hb_avi_stream_header_t ) +
                              sizeof( hb_wave_formatex_t ) +
-                             ( is_ac3 ? 0 : sizeof( hb_wave_mp3_t ) ) );
+                             ( is_passthru ? 0 : sizeof( hb_wave_mp3_t ) ) );
         WriteInt32( m->file, FOURCC( "strl" ) );
         WriteStreamHeader( m->file, &mux_data->header );
         WriteWaveFormatEx( m->file, &mux_data->format.a.f );
-        if( !is_ac3 )
+        if( !is_passthru )
         {
             WriteWaveMp3( m->file, &mux_data->format.a.m );
         }
@@ -600,9 +603,12 @@ static int AVIMux( hb_mux_object_t * m, hb_mux_data_t * mux_data,
     WriteInt32( m->file, job->mux_data->header.Length );
     for( i = 0; i < hb_list_count( title->list_audio ); i++ )
     {
+        int is_passthru;
         audio = hb_list_item( title->list_audio, i );
+        is_passthru = (audio->config.out.codec == HB_ACODEC_AC3) ||
+                      (audio->config.out.codec == HB_ACODEC_DCA);
         fseek( m->file, 264 + i *
-               ( 102 + ( ( audio->config.out.codec == HB_ACODEC_AC3 ) ? 0 :
+               ( 102 + ( is_passthru ? 0 :
                  sizeof( hb_wave_mp3_t ) ) ), SEEK_SET );
         WriteInt32( m->file, audio->priv.mux_data->header.Length );
     }

diff --git a/libhb/muxmkv.c b/libhb/muxmkv.c
index 83a7daa45ca0c696e5a0fd58010abe654c4d8bc6..fa9241fe6ffbe3e7cc66fa19bad42e2b501d5bab 100644 (file)
--- a/libhb/muxmkv.c
+++ b/libhb/muxmkv.c
@@ -165,6 +165,11 @@ static int MKVInit( hb_mux_object_t * m )
 
         switch (audio->config.out.codec)
         {
+            case HB_ACODEC_DCA:
+                track->codecPrivate = NULL;
+                track->codecPrivateSize = 0;
+                track->codecID = MK_ACODEC_DTS;
+                break;
             case HB_ACODEC_AC3:
                 track->codecPrivate = NULL;
                 track->codecPrivateSize = 0;
@@ -221,7 +226,15 @@ static int MKVInit( hb_mux_object_t * m )
         track->trackType = MK_TRACK_AUDIO;
         track->language = audio->config.lang.iso639_2;
         track->extra.audio.samplingFreq = (float)audio->config.out.samplerate;
-        track->extra.audio.channels = (audio->config.out.codec == HB_ACODEC_AC3 ) ? HB_INPUT_CH_LAYOUT_GET_DISCRETE_COUNT(audio->config.in.channel_layout) : HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
+        if (audio->config.out.codec == HB_ACODEC_AC3 ||
+            audio->config.out.codec == HB_ACODEC_DCA)
+        {
+            track->extra.audio.channels = HB_INPUT_CH_LAYOUT_GET_DISCRETE_COUNT(audio->config.in.channel_layout);
+        }
+               else
+        {
+            track->extra.audio.channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
+        }
 //        track->defaultDuration = job->arate * 1000;
         mux_data->track = mk_createTrack(m->file, track);
         if (audio->config.out.codec == HB_ACODEC_VORBIS && track->codecPrivate != NULL)

diff --git a/libhb/sync.c b/libhb/sync.c
index 920a0e3889a7948acbeebe421d28755c8b8473df..f7b342b61676d0a79a74431c4fd70cdd4c986c72 100644 (file)
--- a/libhb/sync.c
+++ b/libhb/sync.c
@@ -60,6 +60,7 @@ struct hb_work_private_s
 
     /* Audio */
     hb_sync_audio_t sync_audio[8];
+    int64_t audio_passthru_slip;
 
     /* Statistics */
     uint64_t st_counts[4];
@@ -351,7 +352,8 @@ static void SyncVideo( hb_work_object_t * w )
          * can deal with overlaps of up to a frame time but anything larger
          * we handle by dropping frames here.
          */
-        if ( (int64_t)( next->start - cur->start ) <= 0 )
+        if ( (int64_t)( next->start - cur->start ) <= 0 ||
+             (int64_t)( (cur->start - pv->audio_passthru_slip ) - pv->next_pts ) < 0 )
         {
             if ( pv->first_drop == 0 )
             {
@@ -730,8 +732,10 @@ static void SyncAudio( hb_work_object_t * w, int i )
     hb_audio_t      * audio = sync->audio;
     hb_buffer_t     * buf;
     hb_fifo_t       * fifo;
+    int64_t start;
 
-    if( audio->config.out.codec == HB_ACODEC_AC3 )
+    if( audio->config.out.codec == HB_ACODEC_AC3 ||
+        audio->config.out.codec == HB_ACODEC_DCA )
     {
         fifo = audio->priv.fifo_out;
     }
@@ -742,6 +746,7 @@ static void SyncAudio( hb_work_object_t * w, int i )
 
     while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
     {
+        start = buf->start - pv->audio_passthru_slip;
         /* if the next buffer is an eof send it downstream */
         if ( buf->size <= 0 )
         {
@@ -756,13 +761,13 @@ static void SyncAudio( hb_work_object_t * w, int i )
             pv->busy &=~ (1 << (i + 1) );
             return;
         }
-        if ( (int64_t)( buf->start - sync->next_pts ) < 0 )
+        if ( (int64_t)( start - sync->next_pts ) < 0 )
         {
             // audio time went backwards.
             // If our output clock is more than a half frame ahead of the
             // input clock drop this frame to move closer to sync.
             // Otherwise drop frames until the input clock matches the output clock.
-            if ( sync->first_drop || sync->next_start - buf->start > 90*15 )
+            if ( sync->first_drop || sync->next_start - start > 90*15 )
             {
                 // Discard data that's in the past.
                 if ( sync->first_drop == 0 )
@@ -774,7 +779,7 @@ static void SyncAudio( hb_work_object_t * w, int i )
                 hb_buffer_close( &buf );
                 continue;
             }
-            sync->next_pts = buf->start;
+            sync->next_pts = start;
         }
         if ( sync->first_drop )
         {
@@ -785,19 +790,19 @@ static void SyncAudio( hb_work_object_t * w, int i )
                     sync->drop_count, sync->first_drop, sync->next_pts );
             sync->first_drop = 0;
             sync->drop_count = 0;
-            sync->next_pts = buf->start;
+            sync->next_pts = start;
         }
-        if ( buf->start - sync->next_pts >= (90 * 70) )
+        if ( start - sync->next_pts >= (90 * 70) )
         {
-            if ( buf->start - sync->next_pts > (90000LL * 60) )
+            if ( start - sync->next_pts > (90000LL * 60) )
             {
                 // there's a gap of more than a minute between the last
                 // frame and this. assume we got a corrupted timestamp
                 // and just drop the next buf.
                 hb_log( "sync: %d minute time gap in audio %d - dropping buf"
                         "  start %lld, next %lld",
-                        (int)((buf->start - sync->next_pts) / (90000*60)),
-                        i, buf->start, sync->next_pts );
+                        (int)((start - sync->next_pts) / (90000*60)),
+                        i, start, sync->next_pts );
                 buf = hb_fifo_get( audio->priv.fifo_raw );
                 hb_buffer_close( &buf );
                 continue;
@@ -805,12 +810,23 @@ static void SyncAudio( hb_work_object_t * w, int i )
             /*
              * there's a gap of at least 70ms between the last
              * frame we processed & the next. Fill it with silence.
+             * Or in the case of DCA, skip some frames from the
+             * other streams.
              */
+            if( sync->audio->config.out.codec == HB_ACODEC_DCA )
+            {
+                hb_log( "sync: audio gap %d ms. Skipping frames. Audio %d"
+                        "  start %lld, next %lld",
+                        (int)((start - sync->next_pts) / 90),
+                        i, start, sync->next_pts );
+                pv->audio_passthru_slip += (start - sync->next_pts);
+                return;
+            }
             hb_log( "sync: adding %d ms of silence to audio %d"
                     "  start %lld, next %lld",
-                    (int)((buf->start - sync->next_pts) / 90),
-                    i, buf->start, sync->next_pts );
-            InsertSilence( w, i, buf->start - sync->next_pts );
+                    (int)((start - sync->next_pts) / 90),
+                    i, start, sync->next_pts );
+            InsertSilence( w, i, start - sync->next_pts );
             return;
         }
 

diff --git a/libhb/work.c b/libhb/work.c
index b6e4aa6c29a2cb8355e15a92bd4f1d6d6cad5381..42613c3fb78c7c666e39ac5261a50b997fb907c4 100644 (file)
--- a/libhb/work.c
+++ b/libhb/work.c
@@ -576,7 +576,8 @@ static void do_job( hb_job_t * job, int cpu_count )
             /* sense-check the requested mixdown */
 
             if( audio->config.out.mixdown == 0 &&
-                audio->config.out.codec != HB_ACODEC_AC3 )
+                audio->config.out.codec != HB_ACODEC_AC3 && 
+                audio->config.out.codec != HB_ACODEC_DCA )
             {
                 /*
                  * Mixdown wasn't specified and this is not pass-through,
@@ -742,7 +743,8 @@ static void do_job( hb_job_t * job, int cpu_count )
         /*
          * Audio Encoder Thread
          */
-        if( audio->config.out.codec != HB_ACODEC_AC3 )
+        if( audio->config.out.codec != HB_ACODEC_AC3 &&
+            audio->config.out.codec != HB_ACODEC_DCA )
         {
             /*
              * Add the encoder thread if not doing AC-3 pass through