From 270e903b7165045d4c0e29c77591b7e905f9a978 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?R=C3=A9mi=20Denis-Courmont?= <remi@remlab.net>
Date: Tue, 9 Aug 2011 19:22:38 +0300
Subject: [PATCH] libsamplerate resampling filter
---
NEWS | 1 +
configure.ac | 7 +-
modules/LIST | 1 +
modules/audio_filter/Modules.am | 1 +
modules/audio_filter/resampler/src.c | 153 +++++++++++++++++++++++++++
po/POTFILES.in | 1 +
6 files changed, 163 insertions(+), 1 deletion(-)
create mode 100644 modules/audio_filter/resampler/src.c
diff --git a/NEWS b/NEWS
index 58dfe7f4622f..7e846a008667 100644
--- a/NEWS
+++ b/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
diff --git a/configure.ac b/configure.ac
index 0dd1565eae9d..1befae90b9ba 100644
--- a/configure.ac
+++ b/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)
diff --git a/modules/LIST b/modules/LIST
index 7a6620f37509..ae234e080e95 100644
--- a/modules/LIST
+++ b/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
diff --git a/modules/audio_filter/Modules.am b/modules/audio_filter/Modules.am
index 4dc2602601ef..db4d9a795e3d 100644
--- a/modules/audio_filter/Modules.am
+++ b/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
diff --git a/modules/audio_filter/resampler/src.c b/modules/audio_filter/resampler/src.c
new file mode 100644
index 000000000000..75e531c2eecc
--- /dev/null
+++ b/modules/audio_filter/resampler/src.c
@@ -0,0 +1,153 @@
+/*****************************************************************************
+ * 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;
+}
diff --git a/po/POTFILES.in b/po/POTFILES.in
index 84a674f8135c..69ffe9e4c89b 100644
--- a/po/POTFILES.in
+++ b/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
--
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