[vlc-devel] [RFC PATCH] modules: add SoX Resampler audio_filter
Thomas Guillem
thomas at gllm.fr
Wed Oct 28 15:02:04 CET 2015
This module can also be used to convert PCM formats, but performances are worse
than with resampler/format.c, so it's deactivated.
---
Change since last patch:
- fixed static / N_;
- simplify configure.ac using PKG_ENABLE_MODULES_VLC.
- use variable-rate engine (SOXR_VR) only from "audio resampler".
- don't call soxr_set_io_ratio when ratio is fixed (from "audio converter").
- fix typo in calloc size.
NEWS | 3 +
configure.ac | 5 +
modules/MODULES_LIST | 1 +
modules/audio_filter/Makefile.am | 9 +-
modules/audio_filter/resampler/soxr.c | 302 ++++++++++++++++++++++++++++++++++
5 files changed, 319 insertions(+), 1 deletion(-)
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..02da362 100644
--- a/configure.ac
+++ b/configure.ac
@@ -3657,6 +3657,11 @@ 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
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..c04b25c 100644
--- a/modules/audio_filter/Makefile.am
+++ b/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)
diff --git a/modules/audio_filter/resampler/soxr.c b/modules/audio_filter/resampler/soxr.c
new file mode 100644
index 0000000..9e2f54b
--- /dev/null
+++ b/modules/audio_filter/resampler/soxr.c
@@ -0,0 +1,302 @@
+/*****************************************************************************
+ * 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 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", 5 )
+ set_callbacks( OpenConverter, Close )
+
+ add_submodule()
+ set_capability( "audio resampler", 5 )
+ 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;
+ 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 = p_sys->f_fixed_ratio == 1.0f ? i_ilen
+ : (size_t) 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;
+}
--
2.1.4
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