static void CloseFilter( vlc_object_t * );
static block_t *Convert( filter_t *, block_t * );
+/* libdts channel order */
+static const uint32_t pi_channels_in[] =
+{ AOUT_CHAN_CENTER, AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT,
+ AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT, AOUT_CHAN_LFE, 0 };
+/* our internal channel order (WG-4 order) */
+static const uint32_t pi_channels_out[] =
+{ AOUT_CHAN_LEFT, AOUT_CHAN_RIGHT, AOUT_CHAN_REARLEFT, AOUT_CHAN_REARRIGHT,
+ AOUT_CHAN_CENTER, AOUT_CHAN_LFE, 0 };
+
/*****************************************************************************
* Local structures
*****************************************************************************/
int i_flags; /* libdts flags, see dtsdec/doc/libdts.txt */
vlc_bool_t b_dontwarn;
int i_nb_channels; /* number of float32 per sample */
+
+ int pi_chan_table[AOUT_CHAN_MAX]; /* channel reordering */
};
/*****************************************************************************
"listening room.")
vlc_module_begin();
+ set_category( CAT_INPUT );
+ set_subcategory( SUBCAT_INPUT_ACODEC );
+ set_shortname( _("DTS" ) );
set_description( _("DTS Coherent Acoustics audio decoder") );
add_bool( "dts-dynrng", 1, NULL, DYNRNG_TEXT, DYNRNG_LONGTEXT, VLC_FALSE );
set_capability( "audio filter", 100 );
vlc_module_end();
/*****************************************************************************
- * Create:
+ * Create:
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
return VLC_EGENERIC;
}
+ aout_CheckChannelReorder( pi_channels_in, pi_channels_out,
+ output.i_physical_channels & AOUT_CHAN_PHYSMASK,
+ p_sys->i_nb_channels,
+ p_sys->pi_chan_table );
+
return VLC_SUCCESS;
}
/*****************************************************************************
* Interleave: helper function to interleave channels
*****************************************************************************/
-static void Interleave( float * p_out, const float * p_in, int i_nb_channels )
+static void Interleave( float * p_out, const float * p_in, int i_nb_channels,
+ int *pi_chan_table )
{
- /* We do not only have to interleave, but also reorder the channels
- * Channel reordering according to number of output channels of libdts
- * The reordering needs to be different for different channel configurations
- * (3F2R, 1F2R etc), so this is only temporary.
- * The WG-4 order is appropriate for stereo, quadrophonia, and 5.1 surround.
- *
- * 6 channel mode
- * channel libdts order WG-4 order
- * 0 C // L
- * 1 L // R
- * 2 R // LS
- * 3 LS // RS
- * 4 RS // C
- * 5 LFE // LFE
- *
- * The libdts moves channels to the front if there are unused spaces, so
- * there is no gap between channels. The translation table says which
- * channel of the new stream is taken from which original channel [use
- * the new channel as the array index, use the number you get from the
- * array to address the original channel].
- */
-
- static const int translation[7][6] =
- {{ 0, 0, 0, 0, 0, 0 }, /* 0 channels (rarely used) */
- { 0, 0, 0, 0, 0, 0 }, /* 1 ch */
- { 0, 1, 0, 0, 0, 0 }, /* 2 */
- { 1, 2, 0, 0, 0, 0 }, /* 3 */
- { 0, 1, 2, 3, 0, 0 }, /* 4 */
- { 1, 2, 3, 4, 0, 0 }, /* 5 */
- { 1, 2, 3, 4, 0, 5 }}; /* 6 */
+ /* We do not only have to interleave, but also reorder the channels. */
int i, j;
for ( j = 0; j < i_nb_channels; j++ )
{
for ( i = 0; i < 256; i++ )
{
- p_out[i * i_nb_channels + j] = p_in[translation[i_nb_channels][j]
- * 256 + i];
+ p_out[i * i_nb_channels + pi_chan_table[j]] = p_in[j * 256 + i];
}
}
}
{
/* Interleave the *$£%ù samples. */
Interleave( (float *)(p_out_buf->p_buffer + i * i_bytes_per_block),
- p_samples, p_sys->i_nb_channels );
+ p_samples, p_sys->i_nb_channels, p_sys->pi_chan_table);
}
}
aout_filter_t aout_filter;
aout_buffer_t in_buf, out_buf;
block_t *p_out;
+ int i_out_size;
- int i_out_size = p_block->i_samples *
+ if( !p_block || !p_block->i_samples )
+ {
+ if( p_block ) p_block->pf_release( p_block );
+ return NULL;
+ }
+
+ i_out_size = p_block->i_samples *
p_filter->fmt_out.audio.i_bitspersample *
- p_filter->fmt_out.audio.i_channels;
+ p_filter->fmt_out.audio.i_channels / 8;
p_out = p_filter->pf_audio_buffer_new( p_filter, i_out_size );
if( !p_out )
{
msg_Warn( p_filter, "can't get output buffer" );
+ p_block->pf_release( p_block );
return NULL;
}
p_out->i_buffer = out_buf.i_nb_bytes;
p_out->i_samples = out_buf.i_nb_samples;
+ p_block->pf_release( p_block );
+
return p_out;
}