--- /dev/null
+/*****************************************************************************
+ * 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;
+}