* fix author field of OSX info dialog

[vlc] / modules / codec / dts.c

diff --git a/modules/codec/dts.c b/modules/codec/dts.c
index ef2b26e928b546f1b0bd29f0a5caaf9ce2cf33a2..6c651c9000b7e06f7a4aaa38e688780d1f31c74d 100644 (file)
--- a/modules/codec/dts.c
+++ b/modules/codec/dts.c
@@ -2,7 +2,7 @@
  * dts.c: parse DTS audio sync info and packetize the stream
  *****************************************************************************
  * Copyright (C) 2003 VideoLAN
- * $Id: dts.c,v 1.13 2004/02/02 23:49:46 gbazin Exp $
+ * $Id$
  *
  * Authors: Jon Lech Johansen <jon-vl@nanocrew.net>
  *          Gildas Bazin <gbazin@netcourrier.com>
@@ -30,7 +30,7 @@
 
 #include "vlc_block_helper.h"
 
-#define DTS_HEADER_SIZE 12
+#define DTS_HEADER_SIZE 14
 
 /*****************************************************************************
  * decoder_sys_t : decoder descriptor
@@ -90,6 +90,8 @@ static block_t       *GetSoutBuffer( decoder_t * );
 vlc_module_begin();
     set_description( _("DTS parser") );
     set_capability( "decoder", 100 );
+    set_category( CAT_INPUT );
+    set_subcategory( SUBCAT_INPUT_ACODEC );
     set_callbacks( OpenDecoder, CloseDecoder );
 
     add_submodule();
@@ -130,6 +132,7 @@ static int OpenDecoder( vlc_object_t *p_this )
     /* Set output properties */
     p_dec->fmt_out.i_cat = AUDIO_ES;
     p_dec->fmt_out.i_codec = VLC_FOURCC('d','t','s',' ');
+    p_dec->fmt_out.audio.i_rate = 0; /* So end_date gets initialized */
 
     /* Set callback */
     p_dec->pf_decode_audio = (aout_buffer_t *(*)(decoder_t *, block_t **))
@@ -172,7 +175,7 @@ static void *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
         return NULL;
     }
 
-    if( (*pp_block)->b_discontinuity )
+    if( (*pp_block)->i_flags&BLOCK_FLAG_DISCONTINUITY )
     {
         p_sys->i_state = STATE_NOSYNC;
     }
@@ -334,8 +337,9 @@ static uint8_t *GetOutBuffer( decoder_t *p_dec, void **pp_out_buffer )
 
     p_dec->fmt_out.audio.i_rate     = p_sys->i_rate;
     p_dec->fmt_out.audio.i_channels = p_sys->i_channels;
-    /* Hack for DTS S/PDIF filter which needs to send 3 frames at a time */
-    p_dec->fmt_out.audio.i_bytes_per_frame = p_sys->i_frame_size * 3;
+    /* Hack for DTS S/PDIF filter which needs to pad the DTS frames */
+    p_dec->fmt_out.audio.i_bytes_per_frame =
+        __MAX( p_sys->i_frame_size, p_sys->i_frame_length * 4 );
     p_dec->fmt_out.audio.i_frame_length = p_sys->i_frame_length;
 
     p_dec->fmt_out.audio.i_original_channels = p_sys->i_channels_conf;
@@ -368,8 +372,12 @@ static aout_buffer_t *GetAoutBuffer( decoder_t *p_dec )
     decoder_sys_t *p_sys = p_dec->p_sys;
     aout_buffer_t *p_buf;
 
-    p_buf = p_dec->pf_aout_buffer_new( p_dec, p_sys->i_frame_length );
+    /* Hack for DTS S/PDIF filter which needs to send 3 frames at a time
+     * (plus a few header bytes) */
+    p_buf = p_dec->pf_aout_buffer_new( p_dec, p_sys->i_frame_length * 4 );
     if( p_buf == NULL ) return NULL;
+    p_buf->i_nb_samples = p_sys->i_frame_length;
+    p_buf->i_nb_bytes = p_sys->i_frame_size;
 
     p_buf->start_date = aout_DateGet( &p_sys->end_date );
     p_buf->end_date =
@@ -402,8 +410,7 @@ static block_t *GetSoutBuffer( decoder_t *p_dec )
  *****************************************************************************/
 static const unsigned int ppi_dts_samplerate[] =
 {
-    0, 8000, 16000, 32000, 64000, 128000,
-    11025, 22050, 44010, 88020, 176400,
+    0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0,
     12000, 24000, 48000, 96000, 192000
 };
 
@@ -414,7 +421,7 @@ static const unsigned int ppi_dts_bitrate[] =
     448000, 512000, 576000, 640000, 768000,
     896000, 1024000, 1152000, 1280000, 1344000,
     1408000, 1411200, 1472000, 1536000, 1920000,
-    2048000, 3072000, 3840000, 4096000, 0, 0
+    2048000, 3072000, 3840000, 1/*open*/, 2/*variable*/, 3/*lossless*/
 };
 
 static int SyncInfo16be( const uint8_t *p_buf,
@@ -424,6 +431,7 @@ static int SyncInfo16be( const uint8_t *p_buf,
                          unsigned int *pi_frame_length )
 {
     unsigned int i_frame_size;
+    unsigned int i_lfe;
 
     *pi_frame_length = (p_buf[4] & 0x01) << 6 | (p_buf[5] >> 2);
     i_frame_size = (p_buf[5] & 0x03) << 12 | (p_buf[6] << 4) |
@@ -433,6 +441,9 @@ static int SyncInfo16be( const uint8_t *p_buf,
     *pi_sample_rate = (p_buf[8] >> 2) & 0x0f;
     *pi_bit_rate = (p_buf[8] & 0x03) << 3 | ((p_buf[9] >> 5) & 0x07);
 
+    i_lfe = (p_buf[10] >> 1) & 0x03;
+    if( i_lfe ) *pi_audio_mode |= 0x10000;
+
     return i_frame_size + 1;
 }
 
@@ -539,8 +550,8 @@ static int SyncInfo( const uint8_t *p_buf,
         p_buf[2] == 0x00 && p_buf[3] == 0xe8 &&
         (p_buf[4] & 0xf0) == 0xf0 && p_buf[5] == 0x07 )
     {
-        uint8_t conv_buf[12];
-        Buf14To16( conv_buf, p_buf, 12, 1 );
+        uint8_t conv_buf[DTS_HEADER_SIZE];
+        Buf14To16( conv_buf, p_buf, DTS_HEADER_SIZE, 1 );
         i_frame_size = SyncInfo16be( conv_buf, &i_audio_mode, pi_sample_rate,
                                      pi_bit_rate, pi_frame_length );
         i_frame_size = i_frame_size * 8 / 14 * 2;
@@ -550,8 +561,8 @@ static int SyncInfo( const uint8_t *p_buf,
              p_buf[2] == 0xe8 && p_buf[3] == 0x00 &&
              p_buf[4] == 0x07 && (p_buf[5] & 0xf0) == 0xf0 )
     {
-        uint8_t conv_buf[12];
-        Buf14To16( conv_buf, p_buf, 12, 0 );
+        uint8_t conv_buf[DTS_HEADER_SIZE];
+        Buf14To16( conv_buf, p_buf, DTS_HEADER_SIZE, 0 );
         i_frame_size = SyncInfo16be( conv_buf, &i_audio_mode, pi_sample_rate,
                                      pi_bit_rate, pi_frame_length );
         i_frame_size = i_frame_size * 8 / 14 * 2;
@@ -567,14 +578,14 @@ static int SyncInfo( const uint8_t *p_buf,
     else if( p_buf[0] == 0xfe && p_buf[1] == 0x7f &&
              p_buf[2] == 0x01 && p_buf[3] == 0x80 )
     {
-        uint8_t conv_buf[12];
-        BufLeToBe( conv_buf, p_buf, 12 );
+        uint8_t conv_buf[DTS_HEADER_SIZE];
+        BufLeToBe( conv_buf, p_buf, DTS_HEADER_SIZE );
         i_frame_size = SyncInfo16be( p_buf, &i_audio_mode, pi_sample_rate,
                                      pi_bit_rate, pi_frame_length );
     }
     else return 0;
 
-    switch( i_audio_mode )
+    switch( i_audio_mode & 0xFFFF )
     {
         case 0x0:
             /* Mono */
@@ -599,16 +610,16 @@ static int SyncInfo( const uint8_t *p_buf,
                                 AOUT_CHAN_CENTER;
             break;
         case 0x6:
-            /* 2F/LFE */
+            /* 2F/1R */
             *pi_channels = 3;
             *pi_channels_conf = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
-                                AOUT_CHAN_LFE;
+                                AOUT_CHAN_REARCENTER;
             break;
         case 0x7:
-            /* 3F/LFE */
+            /* 3F/1R */
             *pi_channels = 4;
             *pi_channels_conf = AOUT_CHAN_LEFT | AOUT_CHAN_RIGHT |
-                                AOUT_CHAN_CENTER | AOUT_CHAN_LFE;
+                                AOUT_CHAN_CENTER | AOUT_CHAN_REARCENTER;
             break;
         case 0x8:
             /* 2F2R */
@@ -660,21 +671,27 @@ static int SyncInfo( const uint8_t *p_buf,
             break;
     }
 
+    if( i_audio_mode & 0x10000 )
+    {
+        (*pi_channels)++;
+        *pi_channels_conf |= AOUT_CHAN_LFE;
+    }
+
     if( *pi_sample_rate >= sizeof( ppi_dts_samplerate ) /
                            sizeof( ppi_dts_samplerate[0] ) )
     {
         return 0;
     }
-
     *pi_sample_rate = ppi_dts_samplerate[ *pi_sample_rate ];
+    if( !*pi_sample_rate ) return 0;
 
     if( *pi_bit_rate >= sizeof( ppi_dts_bitrate ) /
                         sizeof( ppi_dts_bitrate[0] ) )
     {
         return 0;
     }
-
     *pi_bit_rate = ppi_dts_bitrate[ *pi_bit_rate ];
+    if( !*pi_bit_rate ) return 0;
 
     *pi_frame_length = (*pi_frame_length + 1) * 32;