[vlc-devel] [RFC PATCH] modules: add SoX Resampler audio_filter
Thomas Guillem
thomas at gllm.fr
Tue Oct 27 13:00:53 CET 2015
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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