[vlc-devel] [PATCH 2/2] Remove linear resampler

Rémi Denis-Courmont remi at remlab.net
Sun Nov 22 10:54:53 CET 2009


It is computationally more expensive than the ugly one (and has higher
priority). Yet, in my opinion, it sounds even worse.
---
 modules/audio_filter/resampler/Modules.am |    1 -
 modules/audio_filter/resampler/linear.c   |  256 -----------------------------
 2 files changed, 0 insertions(+), 257 deletions(-)
 delete mode 100644 modules/audio_filter/resampler/linear.c

diff --git a/modules/audio_filter/resampler/Modules.am b/modules/audio_filter/resampler/Modules.am
index 07b3357..e00d952 100644
--- a/modules/audio_filter/resampler/Modules.am
+++ b/modules/audio_filter/resampler/Modules.am
@@ -1,5 +1,4 @@
 SOURCES_ugly_resampler = ugly.c
-SOURCES_linear_resampler = linear.c
 SOURCES_bandlimited_resampler = bandlimited.c bandlimited.h
 
 libvlc_LTLIBRARIES += \
diff --git a/modules/audio_filter/resampler/linear.c b/modules/audio_filter/resampler/linear.c
deleted file mode 100644
index a3c7cb3..0000000
--- a/modules/audio_filter/resampler/linear.c
+++ /dev/null
@@ -1,256 +0,0 @@
-/*****************************************************************************
- * linear.c : linear interpolation resampler
- *****************************************************************************
- * Copyright (C) 2002, 2006 the VideoLAN team
- * $Id$
- *
- * Authors: Gildas Bazin <gbazin at netcourrier.com>
- *          Sigmund Augdal Helberg <dnumgis at videolan.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
- *****************************************************************************/
-
-/*****************************************************************************
- * Preamble
- *****************************************************************************/
-
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#include <vlc_common.h>
-#include <vlc_plugin.h>
-#include <vlc_aout.h>
-#include <vlc_filter.h>
-#include <vlc_block.h>
-#include <vlc_cpu.h>
-
-/*****************************************************************************
- * Local prototypes
- *****************************************************************************/
-static int  OpenFilter ( vlc_object_t * );
-static void CloseFilter( vlc_object_t * );
-static block_t *Resample( filter_t *, block_t * );
-
-#if HAVE_FPU
-typedef float sample_t;
-# define VLC_CODEC_NATIVE VLC_CODEC_FL32
-#else
-typedef int32_t sample_t;
-# define VLC_CODEC_NATIVE VLC_CODEC_FI32
-#endif
-
-/*****************************************************************************
- * Local structures
- *****************************************************************************/
-struct filter_sys_t
-{
-    sample_t *p_prev_sample;      /* this filter introduces a 1 sample delay */
-
-    unsigned int i_remainder;                /* remainder of previous sample */
-
-    date_t       end_date;
-};
-
-/*****************************************************************************
- * Module descriptor
- *****************************************************************************/
-vlc_module_begin ()
-    set_description( N_("Audio filter for linear interpolation resampling") )
-    set_category( CAT_AUDIO )
-    set_subcategory( SUBCAT_AUDIO_MISC )
-    set_capability( "audio filter", 5 )
-    set_callbacks( OpenFilter, CloseFilter )
-vlc_module_end ()
-
-/*****************************************************************************
- * Resample: convert a buffer
- *****************************************************************************/
-static block_t *Resample( filter_t *p_filter, block_t *p_in_buf )
-{
-    if( !p_in_buf || !p_in_buf->i_nb_samples )
-    {
-        if( p_in_buf )
-            block_Release( p_in_buf );
-        return NULL;
-    }
-
-    filter_sys_t *p_sys = p_filter->p_sys;
-    unsigned i_nb_channels = p_filter->fmt_in.audio.i_channels;
-    sample_t *p_prev_sample = p_sys->p_prev_sample;
-
-    /* Check if we really need to run the resampler */
-    if( p_filter->fmt_out.audio.i_rate == p_filter->fmt_in.audio.i_rate )
-    {
-        if( !(p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY) )
-        {
-            p_in_buf = block_Realloc( p_in_buf,
-                                      sizeof(sample_t) * i_nb_channels,
-                                      p_in_buf->i_buffer );
-            if( !p_in_buf )
-                return NULL;
-
-            memcpy( p_in_buf->p_buffer, p_prev_sample,
-                    i_nb_channels * sizeof(sample_t) );
-        }
-        return p_in_buf;
-    }
-
-    unsigned i_bytes_per_frame = p_filter->fmt_out.audio.i_channels *
-                                 p_filter->fmt_out.audio.i_bitspersample / 8;
-
-    size_t i_out_size = i_bytes_per_frame * (1 + (p_in_buf->i_nb_samples *
-              p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate));
-    block_t *p_out_buf = filter_NewAudioBuffer( p_filter, i_out_size );
-    if( !p_out_buf )
-        goto out;
-
-    sample_t *p_out = (sample_t *)p_out_buf->p_buffer;
-
-    unsigned i_in_nb = p_in_buf->i_nb_samples;
-    unsigned i_out = 0;
-    const sample_t *p_in = (sample_t *)p_in_buf->p_buffer;
-
-    /* Take care of the previous input sample (if any) */
-    if( p_in_buf->i_flags & BLOCK_FLAG_DISCONTINUITY )
-    {
-        p_out_buf->i_flags |= BLOCK_FLAG_DISCONTINUITY;
-        p_sys->i_remainder = 0;
-        date_Init( &p_sys->end_date, p_filter->fmt_out.audio.i_rate, 1 );
-    }
-    else
-    {
-        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
-        {
-            for( unsigned i = 0; i < i_nb_channels ; i++ )
-            {
-                p_out[i] = p_prev_sample[i];
-#if HAVE_FPU
-                p_out[i] += (p_in[i] - p_prev_sample[i])
-#else
-                p_out[i] += (int64_t)(p_in[i] - p_prev_sample[i])
-#endif
-                    * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
-            }
-            p_out += i_nb_channels;
-            i_out++;
-
-            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-        }
-        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
-    }
-
-    /* Take care of the current input samples (minus last one) */
-    for( unsigned i_in = 0; i_in < i_in_nb - 1; i_in++ )
-    {
-        while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
-        {
-            for( unsigned i = 0; i < i_nb_channels ; i++ )
-            {
-                p_out[i] = p_in[i];
-#if HAVE_FPU
-                p_out[i] += (p_in[i + i_nb_channels] - p_in[i])
-#else
-                p_out[i] += (int64_t)(p_in[i + i_nb_channels] - p_in[i])
-#endif
-                    * p_sys->i_remainder / p_filter->fmt_out.audio.i_rate;
-            }
-            p_out += i_nb_channels;
-            i_out++;
-
-            p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
-        }
-
-        p_in += i_nb_channels;
-        p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
-    }
-
-    /* Backup the last input sample for next time */
-    memcpy( p_prev_sample, p_in, i_nb_channels * sizeof(sample_t) );
-
-    p_out_buf->i_nb_samples = i_out;
-    p_out_buf->i_pts = p_in_buf->i_pts;
-
-    if( p_in_buf->i_pts !=
-        date_Get( &p_sys->end_date ) )
-    {
-        date_Set( &p_sys->end_date, p_in_buf->i_pts );
-    }
-
-    p_out_buf->i_length = date_Increment( &p_sys->end_date,
-                                  p_out_buf->i_nb_samples ) - p_out_buf->i_pts;
-
-    p_out_buf->i_buffer = p_out_buf->i_nb_samples *
-        i_nb_channels * sizeof(sample_t);
-out:
-    block_Release( p_in_buf );
-    return p_out_buf;
-}
-
-/*****************************************************************************
- * OpenFilter:
- *****************************************************************************/
-static int OpenFilter( vlc_object_t *p_this )
-{
-    filter_t *p_filter = (filter_t *)p_this;
-    filter_sys_t *p_sys;
-    int i_out_rate  = p_filter->fmt_out.audio.i_rate;
-
-    if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate ||
-        p_filter->fmt_in.i_codec != VLC_CODEC_NATIVE )
-    {
-        return VLC_EGENERIC;
-    }
- 
-    /* Allocate the memory needed to store the module's structure */
-    p_filter->p_sys = p_sys = malloc( sizeof(struct filter_sys_t) );
-    if( p_sys == NULL )
-        return VLC_ENOMEM;
-
-    p_sys->p_prev_sample = malloc(
-        p_filter->fmt_in.audio.i_channels * sizeof(sample_t) );
-    if( p_sys->p_prev_sample == NULL )
-    {
-        free( p_sys );
-        return VLC_ENOMEM;
-    }
-    date_Init( &p_sys->end_date, p_filter->fmt_in.audio.i_rate, 1 );
-    p_sys->i_remainder = 0;
-
-    p_filter->pf_audio_filter = Resample;
-
-    msg_Dbg( p_this, "%4.4s/%iKHz/%i->%4.4s/%iKHz/%i",
-             (char *)&p_filter->fmt_in.i_codec,
-             p_filter->fmt_in.audio.i_rate,
-             p_filter->fmt_in.audio.i_channels,
-             (char *)&p_filter->fmt_out.i_codec,
-             p_filter->fmt_out.audio.i_rate,
-             p_filter->fmt_out.audio.i_channels);
-
-    p_filter->fmt_out = p_filter->fmt_in;
-    p_filter->fmt_out.audio.i_rate = i_out_rate;
-
-    return 0;
-}
-
-/*****************************************************************************
- * CloseFilter : deallocate data structures
- *****************************************************************************/
-static void CloseFilter( vlc_object_t *p_this )
-{
-    filter_t *p_filter = (filter_t *)p_this;
-    free( p_filter->p_sys->p_prev_sample );
-    free( p_filter->p_sys );
-}
-- 
1.6.5.3




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