]> git.sesse.net Git - vlc/blobdiff - modules/codec/dts.c
* modules/codec/ffmpeg/encoder.c, modules/stream_out/transcode.c: cosmetic fixes.
[vlc] / modules / codec / dts.c
index 59a9fbd410c5d23b49738c398e02b546578eec8e..c6adbc070048bba3edbbb092b6802d66a0c85a9c 100644 (file)
@@ -2,7 +2,7 @@
  * dts.c: parse DTS audio sync info and packetize the stream
  *****************************************************************************
  * Copyright (C) 2003 VideoLAN
- * $Id: dts.c,v 1.10 2004/01/21 17:56:05 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
@@ -56,12 +56,6 @@ struct decoder_sys_t
 
     int i_frame_size, i_bit_rate;
     unsigned int i_frame_length, i_rate, i_channels, i_channels_conf;
-
-    /* This is very hacky. For DTS over S/PDIF we apparently need to send
-     * 3 frames at a time. This should likely be moved to the output stage. */
-    int i_frames_in_buf;
-    aout_buffer_t *p_aout_buffer;        /* current aout buffer being filled */
-
 };
 
 enum {
@@ -96,6 +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();
@@ -130,13 +126,13 @@ static int OpenDecoder( vlc_object_t *p_this )
     p_sys->b_packetizer = VLC_FALSE;
     p_sys->i_state = STATE_NOSYNC;
     aout_DateSet( &p_sys->end_date, 0 );
-    p_sys->i_frames_in_buf = 0;
 
     p_sys->bytestream = block_BytestreamInit( p_dec );
 
     /* 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 **))
@@ -179,7 +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|BLOCK_FLAG_CORRUPTED) )
     {
         p_sys->i_state = STATE_NOSYNC;
     }
@@ -261,7 +257,7 @@ static void *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
             if( SyncCode( p_header ) != VLC_SUCCESS )
             {
                 msg_Dbg( p_dec, "emulated sync word "
-                         "(no sync on following frame) %2.2x%2.2x%2.2x%2.2x",
+                         "(no sync on following frame): %2.2x%2.2x%2.2x%2.2x",
                          (int)p_header[0], (int)p_header[1],
                          (int)p_header[2], (int)p_header[3] );
                 p_sys->i_state = STATE_NOSYNC;
@@ -297,20 +293,10 @@ static void *DecodeBlock( decoder_t *p_dec, block_t **pp_block )
             if( p_sys->i_pts == p_sys->bytestream.p_block->i_pts )
                 p_sys->i_pts = p_sys->bytestream.p_block->i_pts = 0;
 
-            /* So p_block doesn't get re-added several times */
-            *pp_block = block_BytestreamPop( &p_sys->bytestream );
-
             p_sys->i_state = STATE_NOSYNC;
 
-            if( !p_sys->b_packetizer )
-            {
-                if( p_sys->i_frames_in_buf != 3 ) return NULL;
-                else
-                {
-                    p_sys->i_frames_in_buf = 0;
-                    p_sys->p_aout_buffer = 0;
-                }
-            }
+            /* So p_block doesn't get re-added several times */
+            *pp_block = block_BytestreamPop( &p_sys->bytestream );
 
             return p_out_buffer;
         }
@@ -351,7 +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;
-    p_dec->fmt_out.audio.i_bytes_per_frame = p_sys->i_frame_size;
+    /* 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;
@@ -368,17 +356,11 @@ static uint8_t *GetOutBuffer( decoder_t *p_dec, void **pp_out_buffer )
     }
     else
     {
-        if( !p_sys->i_frames_in_buf )
-        {
-            p_sys->p_aout_buffer = GetAoutBuffer( p_dec );
-        }
-        p_buf = p_sys->p_aout_buffer ? p_sys->p_aout_buffer->p_buffer +
-            p_sys->i_frames_in_buf * p_sys->i_frame_size : NULL;
-        *pp_out_buffer = p_sys->p_aout_buffer;
+        aout_buffer_t *p_aout_buffer = GetAoutBuffer( p_dec );
+        p_buf = p_aout_buffer ? p_aout_buffer->p_buffer : NULL;
+        *pp_out_buffer = p_aout_buffer;
     }
 
-    p_sys->i_frames_in_buf++;
-
     return p_buf;
 }
 
@@ -390,12 +372,16 @@ 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 * 3 );
+    /* 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 =
-        aout_DateIncrement( &p_sys->end_date, p_sys->i_frame_length * 3 );
+        aout_DateIncrement( &p_sys->end_date, p_sys->i_frame_length );
 
     return p_buf;
 }
@@ -424,8 +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
 };
 
@@ -436,7 +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,
@@ -446,6 +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) |
@@ -455,10 +441,24 @@ 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;
 }
 
-static int Buf14leTO16be( uint8_t *p_out, const uint8_t *p_in, int i_in )
+static void BufLeToBe( uint8_t *p_out, const uint8_t *p_in, int i_in )
+{
+    int i;
+
+    for( i = 0; i < i_in/2; i++  )
+    {
+        p_out[i*2] = p_in[i*2+1];
+        p_out[i*2+1] = p_in[i*2];
+    }
+}
+
+static int Buf14To16( uint8_t *p_out, const uint8_t *p_in, int i_in, int i_le )
 {
     unsigned char tmp, cur = 0;
     int bits_in, bits_out = 0;
@@ -466,16 +466,16 @@ static int Buf14leTO16be( uint8_t *p_out, const uint8_t *p_in, int i_in )
 
     for( i = 0; i < i_in; i++  )
     {
-       if( i%2 )
-       {
-           tmp = p_in[i-1];
-           bits_in = 8;
-       }
-       else
-       {
-           tmp = p_in[i+1] & 0x3F;
-           bits_in = 8 - 2;
-       }
+        if( i%2 )
+        {
+            tmp = p_in[i-i_le];
+            bits_in = 8;
+        }
+        else
+        {
+            tmp = p_in[i+i_le] & 0x3F;
+            bits_in = 8 - 2;
+        }
 
         if( bits_out < 8 )
         {
@@ -513,12 +513,25 @@ static inline int SyncCode( const uint8_t *p_buf )
     {
         return VLC_SUCCESS;
     }
+    /* 14 bits, big endian version of the bitstream */
+    else if( p_buf[0] == 0x1f && p_buf[1] == 0xff &&
+             p_buf[2] == 0xe8 && p_buf[3] == 0x00 &&
+             p_buf[4] == 0x07 && (p_buf[5] & 0xf0) == 0xf0 )
+    {
+        return VLC_SUCCESS;
+    }
     /* 16 bits, big endian version of the bitstream */
     else if( p_buf[0] == 0x7f && p_buf[1] == 0xfe &&
              p_buf[2] == 0x80 && p_buf[3] == 0x01 )
     {
         return VLC_SUCCESS;
     }
+    /* 16 bits, little endian version of the bitstream */
+    else if( p_buf[0] == 0xfe && p_buf[1] == 0x7f &&
+             p_buf[2] == 0x01 && p_buf[3] == 0x80 )
+    {
+        return VLC_SUCCESS;
+    }
     else return VLC_EGENERIC;
 }
 
@@ -537,11 +550,22 @@ 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];
-       Buf14leTO16be( conv_buf, p_buf, 12 );
+        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;
+    }
+    /* 14 bits, big endian version of the bitstream */
+    else if( p_buf[0] == 0x1f && p_buf[1] == 0xff &&
+             p_buf[2] == 0xe8 && p_buf[3] == 0x00 &&
+             p_buf[4] == 0x07 && (p_buf[5] & 0xf0) == 0xf0 )
+    {
+        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;
+        i_frame_size = i_frame_size * 8 / 14 * 2;
     }
     /* 16 bits, big endian version of the bitstream */
     else if( p_buf[0] == 0x7f && p_buf[1] == 0xfe &&
@@ -550,9 +574,18 @@ static int SyncInfo( const uint8_t *p_buf,
         i_frame_size = SyncInfo16be( p_buf, &i_audio_mode, pi_sample_rate,
                                      pi_bit_rate, pi_frame_length );
     }
+    /* 16 bits, little endian version of the bitstream */
+    else if( p_buf[0] == 0xfe && p_buf[1] == 0x7f &&
+             p_buf[2] == 0x01 && p_buf[3] == 0x80 )
+    {
+        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 */
@@ -577,16 +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 */
@@ -638,21 +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;