[vlc-devel] commit: trivial resampler: switch to audio filter2 ( Rémi Denis-Courmont )
git version control
git at videolan.org
Sun Sep 27 21:37:42 CEST 2009
vlc | branch: master | Rémi Denis-Courmont <remi at remlab.net> | Sun Sep 27 20:19:59 2009 +0300| [8ad84a264e307170f33b08e2856d8f2224876af5] | committer: Rémi Denis-Courmont
trivial resampler: switch to audio filter2
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=8ad84a264e307170f33b08e2856d8f2224876af5
---
modules/audio_filter/resampler/trivial.c | 56 ++++++++++++-----------------
1 files changed, 23 insertions(+), 33 deletions(-)
diff --git a/modules/audio_filter/resampler/trivial.c b/modules/audio_filter/resampler/trivial.c
index 4cc25d0..c7bb418 100644
--- a/modules/audio_filter/resampler/trivial.c
+++ b/modules/audio_filter/resampler/trivial.c
@@ -1,7 +1,7 @@
/*****************************************************************************
* trivial.c : trivial resampler (skips samples or pads with zeroes)
*****************************************************************************
- * Copyright (C) 2002, 2006 the VideoLAN team
+ * Copyright (C) 2002-2009 the VideoLAN team
* $Id$
*
* Authors: Christophe Massiot <massiot at via.ecp.fr>
@@ -32,21 +32,21 @@
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
+#include <vlc_filter.h>
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Create ( vlc_object_t * );
-static void DoWork ( aout_instance_t *, aout_filter_t *, aout_buffer_t *,
- aout_buffer_t * );
+static block_t *DoWork( filter_t *, block_t * );
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin ()
set_description( N_("Audio filter for trivial resampling") )
- set_capability( "audio filter", 1 )
+ set_capability( "audio filter2", 1 )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_MISC )
set_callbacks( Create, NULL )
@@ -57,7 +57,7 @@ vlc_module_end ()
*****************************************************************************/
static int Create( vlc_object_t *p_this )
{
- aout_filter_t * p_filter = (aout_filter_t *)p_this;
+ filter_t * p_filter = (filter_t *)p_this;
if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
|| p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
@@ -68,51 +68,41 @@ static int Create( vlc_object_t *p_this )
|| (p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32
&& p_filter->fmt_in.audio.i_format != VLC_CODEC_FI32) )
{
- return -1;
+ return VLC_EGENERIC;
}
- p_filter->pf_do_work = DoWork;
- p_filter->b_in_place = true;
-
- return 0;
+ p_filter->pf_audio_filter = DoWork;
+ return VLC_SUCCESS;
}
/*****************************************************************************
* DoWork: convert a buffer
*****************************************************************************/
-static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
- aout_buffer_t * p_in_buf, aout_buffer_t * p_out_buf )
+static block_t *DoWork( filter_t * p_filter, block_t * p_block )
{
- int i_in_nb = p_in_buf->i_nb_samples;
+ /* Check if we really need to run the resampler */
+ if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
+ return p_block;
+
+ int i_in_nb = p_block->i_nb_samples;
int i_out_nb = i_in_nb * p_filter->fmt_out.audio.i_rate
/ p_filter->fmt_in.audio.i_rate;
int i_sample_bytes = aout_FormatNbChannels( &p_filter->fmt_in.audio )
* sizeof(int32_t);
- /* Check if we really need to run the resampler */
- if( p_aout->mixer_format.i_rate == p_filter->fmt_in.audio.i_rate )
- {
- return;
- }
-
- if ( p_out_buf != p_in_buf )
- {
- /* For whatever reason the buffer allocator decided to allocate
- * a new buffer. Currently, this never happens. */
- vlc_memcpy( p_out_buf->p_buffer, p_in_buf->p_buffer,
- __MIN(i_out_nb, i_in_nb) * i_sample_bytes );
- }
+ p_block = block_Realloc( p_block, 0, i_out_nb * i_sample_bytes );
+ if( !p_block )
+ return NULL;
- if ( i_out_nb > i_in_nb )
+ if( i_out_nb > i_in_nb )
{
/* Pad with zeroes. */
- memset( p_out_buf->p_buffer + i_in_nb * i_sample_bytes,
+ memset( p_block->p_buffer + i_in_nb * i_sample_bytes,
0, (i_out_nb - i_in_nb) * i_sample_bytes );
}
- p_out_buf->i_nb_samples = i_out_nb;
- p_out_buf->i_buffer = i_out_nb * i_sample_bytes;
- p_out_buf->i_pts = p_in_buf->i_pts;
- p_out_buf->i_length = p_out_buf->i_nb_samples *
- 1000000 / p_filter->fmt_out.audio.i_rate;
+ p_block->i_nb_samples = i_out_nb;
+ p_block->i_length = p_block->i_nb_samples *
+ CLOCK_FREQ / p_filter->fmt_out.audio.i_rate;
+ return p_block;
}
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