[vlc-devel] [RFC PATCH] modules: add SoX Resampler audio_filter

Thomas Guillem thomas at gllm.fr
Tue Oct 27 13:23:57 CET 2015


Moar context:

Discussion about the SoX Resampler patch in pulse audio here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/22158

Quality, performance and delay test here:
http://lastique.github.io/src_test/

The delay may be too high for what we want to do (Variable resampling).

On Tue, Oct 27, 2015, at 13:00, Thomas Guillem wrote:
> Tested only with FL32/S16/S32 and various sample rates conversions.
> 
> TODO: compare perfs and quality with speex/src/ugly.
> 
> ---
>  NEWS                                  |   3 +
>  configure.ac                          |  18 +++
>  modules/MODULES_LIST                  |   1 +
>  modules/audio_filter/Makefile.am      |   7 +
>  modules/audio_filter/resampler/soxr.c | 272
>  ++++++++++++++++++++++++++++++++++
>  5 files changed, 301 insertions(+)
>  create mode 100644 modules/audio_filter/resampler/soxr.c
> 
> diff --git a/NEWS b/NEWS
> index df0354f..bd9d0ef 100644
> --- a/NEWS
> +++ b/NEWS
> @@ -83,6 +83,9 @@ Audio output:
>     It now supports HDMI/SPDIF passthrough for AC3, 5.1/7.1 and float
>     output.
>   * Added Tizen audio module.
>  
> +Audio filters and output:
> + * Add SoX Resampler library audio filter module (converter and
> resampler)
> +
>  Video ouput:
>   * Linux/BSD default video output is now OpenGL, instead of Xvideo
>   * Wayland shell surface window provider
> diff --git a/configure.ac b/configure.ac
> index 2776a8c..6481d57 100644
> --- a/configure.ac
> +++ b/configure.ac
> @@ -3657,6 +3657,24 @@ dnl
>  PKG_ENABLE_MODULES_VLC([SAMPLERATE], [], [samplerate], [Resampler with
>  libsamplerate], [auto])
>  
>  dnl
> +dnl  soxr module
> +dnl
> +AC_ARG_ENABLE(soxr,
> +  [AS_HELP_STRING([--enable-soxr],
> +    [use the SoX Resampler library (default auto)])])
> +have_soxr="no"
> +AS_IF([test "${enable_soxr}" != "no"], [
> +  PKG_CHECK_MODULES([SOXR], [soxr >= 0.1], [
> +    have_soxr="yes"
> +  ], [
> +    AS_IF([test "x${enable_soxr}" != "x"], [
> +      AC_MSG_ERROR([$SOXR_PKG_ERRORS. soxr 0.1 or later required.])
> +    ])
> +  ])
> +])
> +AM_CONDITIONAL([HAVE_SOXR], [test "${have_soxr}" = "yes"])
> +
> +dnl
>  dnl  OS/2 KAI plugin
>  dnl
>  AC_ARG_ENABLE(kai,
> diff --git a/modules/MODULES_LIST b/modules/MODULES_LIST
> index 0c4892a..c2faebe 100644
> --- a/modules/MODULES_LIST
> +++ b/modules/MODULES_LIST
> @@ -344,6 +344,7 @@ $Id$
>   * smf: Standard MIDI file demuxer
>   * smooth: Microsoft Smooth Streaming input
>   * sndio: OpenBSD sndio audio output
> + * soxr: SoX Resampler library audio filter
>   * spatializer: A spatializer audio filter
>   * speex: a speex audio decoder/packetizer using the libspeex library
>   * speex_resampler: audio resampler using the libspeexdsp library
> diff --git a/modules/audio_filter/Makefile.am
> b/modules/audio_filter/Makefile.am
> index 372d08e..2090133 100644
> --- a/modules/audio_filter/Makefile.am
> +++ b/modules/audio_filter/Makefile.am
> @@ -124,3 +124,10 @@ libspeex_resampler_plugin_la_LIBADD =
> $(SPEEXDSP_LIBS)
>  if HAVE_SPEEXDSP
>  audio_filter_LTLIBRARIES += libspeex_resampler_plugin.la
>  endif
> +
> +libsoxr_resampler_plugin_la_SOURCES = audio_filter/resampler/soxr.c
> +libsoxr_resampler_plugin_la_CFLAGS = $(AM_CFLAGS) $(SOXR_CFLAGS)
> +libsoxr_resampler_plugin_la_LIBADD = $(SOXR_LIBS)
> +if HAVE_SOXR
> +audio_filter_LTLIBRARIES += libsoxr_resampler_plugin.la
> +endif
> diff --git a/modules/audio_filter/resampler/soxr.c
> b/modules/audio_filter/resampler/soxr.c
> new file mode 100644
> index 0000000..bf7d3a5
> --- /dev/null
> +++ b/modules/audio_filter/resampler/soxr.c
> @@ -0,0 +1,272 @@
> +/*****************************************************************************
> + * soxr.c: resampler/converter using The SoX Resampler library
> +
> *****************************************************************************
> + * Copyright (C) 2015 VLC authors, VideoLAN and VideoLabs
> + *
> + * Authors: Thomas Guillem <thomas at gllm.fr>
> + *
> + * This program is free software; you can redistribute it and/or modify
> it
> + * under the terms of the GNU Lesser General Public License as published
> by
> + * the Free Software Foundation; either version 2.1 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 Lesser General Public License for more details.
> + *
> + * You should have received a copy of the GNU Lesser 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_aout.h>
> +#include <vlc_filter.h>
> +#include <vlc_plugin.h>
> +
> +#include <math.h>
> +#include <soxr.h>
> +
> +#define SOXR_QUALITY_TEXT "Sox Resampling quality"
> +
> +const int soxr_resampler_quality_vlclist[] = { 0, 1, 2, 3, 4 };
> +static const char *const soxr_resampler_quality_vlctext[] =
> +{
> +    "Quick cubic interpolation",
> +    "Low 16-bit with larger rolloff",
> +    "Medium 16-bit with medium rolloff",
> +    "High quality",
> +    "Very high quality"
> +};
> +const soxr_datatype_t soxr_resampler_quality_list[] =
> +{
> +    SOXR_QQ,
> +    SOXR_LQ,
> +    SOXR_MQ,
> +    SOXR_HQ,
> +    SOXR_VHQ
> +};
> +#define MAX_SOXR_QUALITY 4
> +
> +static int Open( vlc_object_t * );
> +static int OpenResampler( vlc_object_t * );
> +static void Close( vlc_object_t * );
> +
> +vlc_module_begin ()
> +    set_shortname( "SoX Resampler" )
> +    set_category( CAT_AUDIO )
> +    set_subcategory( SUBCAT_AUDIO_MISC )
> +    add_integer( "soxr-resampler-quality", 2,
> +                SOXR_QUALITY_TEXT, NULL, true )
> +        change_integer_list( soxr_resampler_quality_vlclist,
> +                             soxr_resampler_quality_vlctext )
> +    set_capability ( "audio converter", 3 )
> +    set_callbacks( Open, Close )
> +
> +    add_submodule()
> +    set_capability( "audio resampler", 3 )
> +    set_callbacks( OpenResampler, Close )
> +    add_shortcut( "soxr" )
> +vlc_module_end ()
> +
> +struct filter_sys_t
> +{
> +    soxr_t soxr;
> +    block_t *p_last_in;
> +};
> +
> +static block_t *Resample( filter_t *, block_t * );
> +
> +static bool
> +SoXR_GetFormat( vlc_fourcc_t i_format, soxr_datatype_t *p_type )
> +{
> +    switch( i_format )
> +    {
> +        case VLC_CODEC_FL64:
> +            *p_type = SOXR_FLOAT64_I;
> +            return true;
> +        case VLC_CODEC_FL32:
> +            *p_type = SOXR_FLOAT32_I;
> +            return true;
> +        case VLC_CODEC_S32N:
> +            *p_type = SOXR_INT32_I;
> +            return true;
> +        case VLC_CODEC_S16N:
> +            *p_type = SOXR_INT16_I;
> +            return true;
> +        default:
> +            return false;
> +    }
> +}
> +
> +static int
> +OpenResampler( vlc_object_t *p_obj )
> +{
> +    filter_t *p_filter = (filter_t *)p_obj;
> +
> +    /* Cannot remix */
> +    if( p_filter->fmt_in.audio.i_physical_channels
> +            != p_filter->fmt_out.audio.i_physical_channels
> +     || p_filter->fmt_in.audio.i_original_channels
> +            != p_filter->fmt_out.audio.i_original_channels )
> +        return VLC_EGENERIC;
> +
> +    /* Get SoXR input/output format */
> +    soxr_datatype_t i_itype, i_otype;
> +    if( !SoXR_GetFormat( p_filter->fmt_in.audio.i_format, &i_itype )
> +     || !SoXR_GetFormat( p_filter->fmt_out.audio.i_format, &i_otype ) )
> +        return VLC_EGENERIC;
> +
> +    filter_sys_t *p_sys = calloc( 1, sizeof( struct filter_sys_t * ) );
> +    if( unlikely( p_sys == NULL ) )
> +        return VLC_ENOMEM;
> +
> +    /* Setup SoXR */
> +    int64_t i_vlc_q = var_InheritInteger( p_obj,
> "soxr-resampler-quality" );
> +    if( i_vlc_q < 0 )
> +        i_vlc_q = 0;
> +    else if( i_vlc_q > MAX_SOXR_QUALITY )
> +        i_vlc_q = MAX_SOXR_QUALITY;
> +    const unsigned long i_recipe = soxr_resampler_quality_list[i_vlc_q];
> +    const unsigned i_channels = aout_FormatNbChannels(
> &p_filter->fmt_in.audio );
> +    const double f_ratio = p_filter->fmt_out.audio.i_rate
> +                           / (double) p_filter->fmt_in.audio.i_rate;
> +    const unsigned long i_flags = f_ratio == 1.f ? 0
> +                                : SOXR_VR; /* variable rate */
> +
> +    /* Create SoXR */
> +    soxr_error_t error;
> +    soxr_io_spec_t io_spec = soxr_io_spec( i_itype, i_otype );
> +    soxr_quality_spec_t q_spec = soxr_quality_spec( i_recipe, i_flags );
> +    p_sys->soxr = soxr_create( 1, f_ratio, i_channels,
> +                               &error, &io_spec, &q_spec, NULL );
> +    if( error )
> +    {
> +        msg_Err( p_filter, "soxr_create failed: %s", soxr_strerror(
> error ) );
> +        free( p_sys );
> +        return VLC_EGENERIC;
> +    }
> +    soxr_set_io_ratio( p_sys->soxr, 1 / f_ratio, 0 );
> +
> +    msg_Dbg( p_filter, "Using SoX Resampler: %4.4s, %d Hz to %4.4s %d
> Hz. "
> +             "quality: '%s'",
> +             (const char *)&p_filter->fmt_in.audio.i_format,
> +             p_filter->fmt_in.audio.i_rate,
> +             (const char *)&p_filter->fmt_out.audio.i_format,
> +             p_filter->fmt_out.audio.i_rate,
> +             soxr_resampler_quality_vlctext[i_vlc_q] );
> +
> +    p_filter->p_sys = p_sys;
> +    p_filter->pf_audio_filter = Resample;
> +    return VLC_SUCCESS;
> +}
> +
> +static int
> +Open( vlc_object_t *p_obj )
> +{
> +    filter_t *p_filter = (filter_t *)p_obj;
> +
> +    /* Will change rate */
> +    if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
> )
> +        return VLC_EGENERIC;
> +    return OpenResampler( p_obj );
> +}
> +
> +static void
> +Close( vlc_object_t *p_obj )
> +{
> +    filter_t *p_filter = (filter_t *)p_obj;
> +    filter_sys_t *p_sys = p_filter->p_sys;
> +
> +    soxr_delete( p_sys->soxr );
> +
> +    if( unlikely( p_sys->p_last_in ) )
> +        block_Release( p_sys->p_last_in );
> +
> +    free( p_sys );
> +}
> +
> +static block_t *
> +Resample( filter_t *p_filter, block_t *p_in )
> +{
> +    filter_sys_t *p_sys = p_filter->p_sys;
> +
> +    /* Prepend last remaining input buffer to the current one */
> +    if( unlikely( p_sys->p_last_in ) )
> +    {
> +        p_in = block_Realloc( p_in, p_sys->p_last_in->i_buffer,
> p_in->i_buffer );
> +        if( unlikely( p_in == NULL ) )
> +            return NULL;
> +
> +        memcpy( p_in->p_buffer, p_sys->p_last_in->p_buffer,
> +                p_sys->p_last_in->i_buffer );
> +        p_in->i_nb_samples += p_sys->p_last_in->i_nb_samples;
> +        block_Release( p_sys->p_last_in );
> +        p_sys->p_last_in = NULL;
> +    }
> +
> +    const double f_ratio = p_filter->fmt_out.audio.i_rate
> +                         / (double) p_filter->fmt_in.audio.i_rate;
> +    const size_t i_ilen = p_in->i_nb_samples;
> +    const size_t i_olen = ceil( i_ilen * f_ratio );
> +    const size_t i_oframesize =
> p_filter->fmt_out.audio.i_bytes_per_frame;
> +    size_t i_idone, i_odone;
> +
> +    /* Use input buffer as output if there is enough room */
> +    block_t *p_out = i_ilen > i_olen ? p_in
> +                   : block_Alloc( i_olen * i_oframesize );
> +    if( unlikely(p_out == NULL) )
> +        goto error;
> +
> +    /* Process SoXR */
> +    soxr_set_io_ratio( p_sys->soxr, 1 / f_ratio, i_olen );
> +    soxr_error_t error = soxr_process( p_sys->soxr,
> +                                       p_in->p_buffer, i_ilen, &i_idone,
> +                                       p_out->p_buffer, i_olen, &i_odone
> );
> +    if( error )
> +    {
> +        msg_Err( p_filter, "soxr_process failed: %s", soxr_strerror(
> error ) );
> +        goto error;
> +    }
> +
> +    if( unlikely( i_idone < i_ilen ) )
> +    {
> +        msg_Warn( p_filter, "processed input len < input len, "
> +                 "keeping buffer for next Resample call" );
> +        const size_t i_done_size = i_idone
> +                                 *
> p_filter->fmt_out.audio.i_bytes_per_frame;
> +
> +        /* Realloc since p_in can be used as p_out */
> +        p_sys->p_last_in = block_Alloc( p_in->i_buffer - i_done_size );
> +        if( unlikely( p_sys->p_last_in == NULL ) )
> +            goto error;
> +        memcpy( p_sys->p_last_in->p_buffer,
> +                p_in->p_buffer + i_done_size, p_in->i_buffer -
> i_done_size );
> +        p_sys->p_last_in->i_nb_samples = p_in->i_nb_samples - i_idone;
> +    }
> +
> +    p_out->i_buffer = i_odone * i_oframesize;
> +    p_out->i_nb_samples = i_odone;
> +    p_out->i_pts = p_in->i_pts;
> +    p_out->i_length = i_odone * CLOCK_FREQ /
> p_filter->fmt_out.audio.i_rate;
> +
> +    if( p_out != p_in )
> +        block_Release( p_in );
> +    return p_out;
> +
> +error:
> +
> +    if( p_out && p_out != p_in )
> +        block_Release( p_out );
> +    block_Release( p_in );
> +    return NULL;
> +}
> -- 
> 2.1.4
> 


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