Commit 270e903b authored by Rémi Denis-Courmont's avatar Rémi Denis-Courmont

libsamplerate resampling filter

parent 0fdaa3d7
......@@ -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
......
......@@ -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)
......
......@@ -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
......
......@@ -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
......
/*****************************************************************************
* 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;
}
......@@ -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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