modules: add SoX Resampler audio_filter

This resampler supports 5 quality presets: from 0 to 4, 2 is the default and
correspond to "mq". It supports integer and float samples. Performances are way
better when this module is used as an "audio converter" (fixed sample rate).

See http://lastique.github.io/src_test/ for comparison with speexdsp.

It is deactivated for now.

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

configure.ac

@@ -3656,6 +3656,11 @@ dnl libsamplerate plugin
 dnl
 PKG_ENABLE_MODULES_VLC([SAMPLERATE], [], [samplerate], [Resampler with libsamplerate], [auto])
 
+dnl
+dnl  soxr module
+dnl
+PKG_ENABLE_MODULES_VLC([SOXR], [], [soxr >= 0.1], [SoX Resampler library], [auto])
+
 dnl
 dnl  OS/2 KAI plugin
 dnl

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

modules/audio_filter/Makefile.am

@@ -111,12 +111,19 @@ libsamplerate_plugin_la_CPPFLAGS = $(AM_CPPFLAGS) $(SAMPLERATE_CFLAGS)
 libsamplerate_plugin_la_LDFLAGS = $(AM_LDFLAGS) -rpath '$(audio_filterdir)'
 libsamplerate_plugin_la_LIBADD = $(LIBM) $(SAMPLERATE_LIBS)
 
+libsoxr_plugin_la_SOURCES = audio_filter/resampler/soxr.c
+libsoxr_plugin_la_CPPFLAGS = $(AM_CPPFLAGS) $(SOXR_CFLAGS)
+libsoxr_plugin_la_LDFLAGS = $(AM_LDFLAGS) -rpath '$(audio_filterdir)'
+libsoxr_plugin_la_LIBADD = $(LIBM) $(SOXR_LIBS)
+
 audio_filter_LTLIBRARIES += \
 	$(LTLIBsamplerate) \
+	$(LTLIBsoxr) \
 	libugly_resampler_plugin.la
 EXTRA_LTLIBRARIES += \
 	libbandlimited_resampler_plugin.la \
-	libsamplerate_plugin.la
+	libsamplerate_plugin.la \
+	libsoxr_plugin.la
 
 libspeex_resampler_plugin_la_SOURCES = audio_filter/resampler/speex.c
 libspeex_resampler_plugin_la_CFLAGS = $(AM_CFLAGS) $(SPEEXDSP_CFLAGS)

modules/audio_filter/resampler/soxr.c

+/*****************************************************************************
+ * soxr.c: resampler/converter using The SoX Resampler library
+ *****************************************************************************
+ * Copyright (C) 2015 VLC authors, VideoLAN and VideoLabs
+ *
+ * Authors: Thomas Guillem <thomas@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 N_( "Sox Resampling quality" )
+
+static const int soxr_resampler_quality_vlclist[] = { 0, 1, 2, 3, 4 };
+static const char *const soxr_resampler_quality_vlctext[] =
+{
+    N_( "Quick cubic interpolation" ),
+    N_( "Low 16-bit with larger roll-off" ),
+    N_( "Medium 16-bit with medium roll-off" ),
+    N_( "High quality" ),
+    N_( "Very high quality" )
+};
+static 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 OpenConverter( 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", 0 )
+    set_callbacks( OpenConverter, Close )
+
+    add_submodule()
+    set_capability( "audio resampler", 0 )
+    set_callbacks( OpenResampler, Close )
+    add_shortcut( "soxr" )
+vlc_module_end ()
+
+struct filter_sys_t
+{
+    soxr_t  soxr;
+    double  f_fixed_ratio;
+    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
+Open( vlc_object_t *p_obj, bool b_change_ratio )
+{
+    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;
+    /* XXX: Performances are worse with Variable-Rate */
+    const unsigned long i_flags = b_change_ratio ? SOXR_VR : 0;
+
+    p_sys->f_fixed_ratio = b_change_ratio ? 0.0f : f_ratio;
+    soxr_error_t error;
+    /* IO spec */
+    soxr_io_spec_t io_spec = soxr_io_spec( i_itype, i_otype );
+    /* Quality spec */
+    soxr_quality_spec_t q_spec = soxr_quality_spec( i_recipe, i_flags );
+    /* Create SoXR */
+    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;
+    }
+    if( b_change_ratio )
+        soxr_set_io_ratio( p_sys->soxr, 1 / f_ratio, 0 );
+
+    msg_Dbg( p_filter, "Using SoX Resampler with '%s' engine and '%s' quality "
+             "to convert %4.4s/%dHz to %4.4s/%dHz.",
+             soxr_engine( p_sys->soxr ), soxr_resampler_quality_vlctext[i_vlc_q],
+             (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 );
+
+    p_filter->p_sys = p_sys;
+    p_filter->pf_audio_filter = Resample;
+    return VLC_SUCCESS;
+}
+
+static int
+OpenResampler( vlc_object_t *p_obj )
+{
+    filter_t *p_filter = (filter_t *)p_obj;
+
+    /* A resampler doesn't convert the format */
+    if( p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format )
+        return VLC_EGENERIC;
+    return Open( p_obj, true );
+}
+
+static int
+OpenConverter( vlc_object_t *p_obj )
+{
+    filter_t *p_filter = (filter_t *)p_obj;
+
+    /* Don't use SoXR to convert format. Prefer to use converter/format.c that
+     * has better performances */
+    if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate )
+        return VLC_EGENERIC;
+    return Open( p_obj, false );
+}
+
+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 size_t i_oframesize = p_filter->fmt_out.audio.i_bytes_per_frame;
+    const size_t i_ilen = p_in->i_nb_samples;
+    size_t i_olen, i_idone, i_odone;
+
+    if( p_sys->f_fixed_ratio == 0.0f )
+    {
+        /* "audio resampler" with variable ratio */
+
+        const double f_ratio = p_filter->fmt_out.audio.i_rate
+                             / (double) p_filter->fmt_in.audio.i_rate;
+        if( f_ratio == 1.0f )
+            return p_in;
+
+        /* processed output len might be a little bigger than expected */
+        i_olen = lrint( ( i_ilen + 2 ) * f_ratio * 11. / 10. );
+
+        soxr_set_io_ratio( p_sys->soxr, 1 / f_ratio, i_olen );
+    }
+    else
+        i_olen = lrint( i_ilen * p_sys->f_fixed_ratio );
+
+    /* 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_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;
+}