libsamplerate resampling filter

NEWS

@@ -105,6 +105,7 @@ Audio Output:
  * New audio output in memory (amem)
  * Important simplification and improvements in the core audio output
  * New audio output based on OpenSL ES API for Android
+ * New audio resampler using the Secret Rabbit Code (a.k.a. libsamplerate)
 
 Video Filter:
  * New gradfun filter for debanding videos using dithering

configure.ac

@@ -703,7 +703,7 @@ AC_CHECK_FUNC(getopt_long,, [
 AC_SUBST(GNUGETOPT_LIBS)
 
 AC_CHECK_LIB(m,cos,[
-  VLC_ADD_LIBS([adjust wave ripple psychedelic gradient a52tofloat32 dtstofloat32 x264 goom visual panoramix rotate noise grain scene kate flac lua chorus_flanger freetype avcodec avformat access_avio swscale postproc i420_rgb faad twolame equalizer spatializer param_eq libvlccore freetype mod mpc dmo quicktime realvideo qt4 compressor headphone_channel_mixer normvol audiobargraph_a speex mono colorthres extract ball access_imem hotkeys mosaic gaussianblur dbus x264],[-lm])
+  VLC_ADD_LIBS([adjust wave ripple psychedelic gradient a52tofloat32 dtstofloat32 x264 goom visual panoramix rotate noise grain scene kate flac lua chorus_flanger freetype avcodec avformat access_avio swscale postproc i420_rgb faad twolame equalizer spatializer param_eq samplerate libvlccore freetype mod mpc dmo quicktime realvideo qt4 compressor headphone_channel_mixer normvol audiobargraph_a speex mono colorthres extract ball access_imem hotkeys mosaic gaussianblur dbus x264],[-lm])
   LIBM="-lm"
 ], [
   LIBM=""
@@ -3015,6 +3015,11 @@ dnl libfluidsynth (MIDI synthetizer) plugin
 dnl
 PKG_ENABLE_MODULES_VLC([FLUIDSYNTH], [], [fluidsynth], [MIDI synthetiser with libfluidsynth], [auto])
 
+dnl
+dnl libsamplerate plugin
+dnl
+PKG_ENABLE_MODULES_VLC([SAMPLERATE], [], [samplerate], [Resampler with libsamplerate], [auto])
+
 dnl
 dnl Teletext Modules
 dnl vbi decoder plugin (using libzbvi)

modules/LIST

@@ -275,6 +275,7 @@ $Id$
  * rss: Display a RSS feed on the video output
  * rtp: rtp demux module
  * rv32: RV32 image format conversion module
+ * samplerate: Secret Rabbit Code (libsamplerate) audio resampler
  * sap: Interface module to read SAP/SDP announcements
  * scale: Images rescaler
  * scaletempo: Scale audio tempo in sync with playback rate

modules/audio_filter/Modules.am

@@ -56,6 +56,7 @@ libvlc_LTLIBRARIES += \
 SOURCES_bandlimited_resampler = \
 	resampler/bandlimited.c resampler/bandlimited.h
 SOURCES_ugly_resampler = resampler/ugly.c
+SOURCES_samplerate = resampler/src.c
 
 libvlc_LTLIBRARIES += \
 	libugly_resampler_plugin.la

modules/audio_filter/resampler/src.c

+/*****************************************************************************
+ * src.c : Secret Rabbit Code (a.k.a. libsamplerate) resampler
+ *****************************************************************************
+ * Copyright (C) 2011 Rémi Denis-Courmont
+ *
+ * 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.
+ *****************************************************************************/
+
+#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 <samplerate.h>
+#include <math.h>
+
+#define SRC_CONV_TYPE_TEXT N_("Sample rate converter type")
+#define SRC_CONV_TYPE_LONGTEXT N_( \
+    "Different resampling algorithm are supported. " \
+    "The best one is slower, while the fast one exhibits low quality.")
+static const int conv_type_values[] = {
+    SRC_SINC_BEST_QUALITY, SRC_SINC_MEDIUM_QUALITY, SRC_SINC_FASTEST,
+    SRC_ZERO_ORDER_HOLD, SRC_LINEAR,
+};
+static const char *const conv_type_texts[] = {
+    "Sinc function (best quality)", "Sinc function (medium quality)",
+    "Sinc function (fast)", "Zero Order Hold (fastest)", "Linear (fastest)",
+};
+
+static int Open (vlc_object_t *);
+static void Close (vlc_object_t *);
+
+vlc_module_begin ()
+    set_shortname (N_("SRC resampler"))
+    set_description (N_("Secret Rabbit Code (libsamplerate) resampler") )
+    set_category (CAT_AUDIO)
+    set_subcategory (SUBCAT_AUDIO_MISC)
+    add_integer ("src-converter-type", SRC_SINC_MEDIUM_QUALITY,
+                 SRC_CONV_TYPE_TEXT, SRC_CONV_TYPE_LONGTEXT, true)
+        change_integer_list (conv_type_values, conv_type_texts)
+    set_capability ("audio filter", 50)
+    set_callbacks (Open, Close)
+vlc_module_end ()
+
+static block_t *Resample (filter_t *, block_t *);
+
+static int Open (vlc_object_t *obj)
+{
+    filter_t *filter = (filter_t *)obj;
+
+    /* Only float->float */
+    if (filter->fmt_in.audio.i_format != VLC_CODEC_FL32
+     || filter->fmt_out.audio.i_format != VLC_CODEC_FL32
+    /* No channels remapping */
+     || filter->fmt_in.audio.i_physical_channels
+                                  != filter->fmt_out.audio.i_physical_channels
+     || filter->fmt_in.audio.i_original_channels
+                                  != filter->fmt_out.audio.i_original_channels
+    /* Different sample rate */
+     || filter->fmt_in.audio.i_rate == filter->fmt_out.audio.i_rate)
+        return VLC_EGENERIC;
+
+    int type = var_InheritInteger (obj, "src-converter-type");
+    int channels = aout_FormatNbChannels (&filter->fmt_in.audio);
+    int err;
+
+    SRC_STATE *s = src_new (type, channels, &err);
+    if (s == NULL)
+    {
+        msg_Err (obj, "cannot initialize resampler: %s", src_strerror (err));
+        return VLC_EGENERIC;
+    }
+
+    filter->p_sys = (filter_sys_t *)s;
+    filter->pf_audio_filter = Resample;
+    return VLC_SUCCESS;
+}
+
+static void Close (vlc_object_t *obj)
+{
+    filter_t *filter = (filter_t *)obj;
+    SRC_STATE *s = (SRC_STATE *)filter->p_sys;
+
+    src_delete (s);
+}
+
+static block_t *Resample (filter_t *filter, block_t *in)
+{
+    block_t *out = NULL;
+    const size_t framesize = filter->fmt_out.audio.i_bytes_per_frame;
+
+    SRC_STATE *s = (SRC_STATE *)filter->p_sys;
+    SRC_DATA src;
+
+    src.src_ratio = (double)filter->fmt_out.audio.i_rate
+                  / (double)filter->fmt_in.audio.i_rate;
+
+    int err = src_set_ratio (s, src.src_ratio);
+    if (err != 0)
+    {
+        msg_Err (filter, "cannot update resampling ratio: %s",
+                 src_strerror (err));
+        goto error;
+    }
+
+    src.input_frames = in->i_nb_samples;
+    src.output_frames = ceil (src.src_ratio * src.input_frames);
+    src.end_of_input = 0;
+
+    out = block_Alloc (src.output_frames * framesize);
+    if (unlikely(out == NULL))
+        goto error;
+
+    src.data_in = (float *)in->p_buffer;
+    src.data_out = (float *)out->p_buffer;
+
+    err = src_process (s, &src);
+    if (err != 0)
+    {
+        msg_Err (filter, "cannot resample: %s", src_strerror (err));
+        block_Release (out);
+        out = NULL;
+        goto error;
+    }
+
+    if (src.input_frames_used < src.input_frames)
+        msg_Warn (filter, "lost %ld of %ld input frames",
+                  src.input_frames - src.input_frames_used, src.input_frames);
+
+    out->i_buffer = src.output_frames_gen * framesize;
+    out->i_nb_samples = src.output_frames_gen;
+    out->i_pts = in->i_pts;
+    out->i_length = src.output_frames_gen * CLOCK_FREQ
+                  / filter->fmt_out.audio.i_rate;
+error:
+    block_Release (in);
+    return out;
+}

po/POTFILES.in

@@ -301,6 +301,7 @@ modules/audio_filter/normvol.c
 modules/audio_filter/param_eq.c
 modules/audio_filter/resampler/bandlimited.c
 modules/audio_filter/resampler/bandlimited.h
+modules/audio_filter/resampler/src.c
 modules/audio_filter/resampler/ugly.c
 modules/audio_filter/scaletempo.c
 modules/audio_filter/spatializer/allpass.cpp