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diff --git a/include/gst/audio/audio-resampler.h b/include/gst/audio/audio-resampler.h
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+++ b/include/gst/audio/audio-resampler.h
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+/* GStreamer
+ * Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_AUDIO_RESAMPLER_H__
+#define __GST_AUDIO_RESAMPLER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+/**
+ * GstAudioResampler:
+ *
+ * Opaque #GstAudioResampler struct.
+ *
+ * Since: 1.10
+ */
+typedef struct _GstAudioResampler GstAudioResampler;
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUTOFF:
+ *
+ * G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUTOFF      "GstAudioResampler.cutoff"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION:
+ *
+ * G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
+ * after the stopband for the kaiser window. 85 dB is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH:
+ *
+ * G_TYPE_DOUBLE, transition bandwidth. The width of the
+ * transition band for the kaiser window. 0.087 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUBIC_B:
+ *
+ * G_TYPE_DOUBLE, B parameter of the cubic filter.
+ * Values between 0.0 and 2.0 are accepted. 1.0 is the default.
+ *
+ * Below are some values of popular filters:
+ *                    B       C
+ * Hermite           0.0     0.0
+ * Spline            1.0     0.0
+ * Catmull-Rom       0.0     1/2
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B      "GstAudioResampler.cubic-b"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_CUBIC_C:
+ *
+ * G_TYPE_DOUBLE, C parameter of the cubic filter.
+ * Values between 0.0 and 2.0 are accepted. 0.0 is the default.
+ *
+ * See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
+ */
+#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C      "GstAudioResampler.cubic-c"
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_N_TAPS:
+ *
+ * G_TYPE_INT: the number of taps to use for the filter.
+ * 0 is the default and selects the taps automatically.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_N_TAPS      "GstAudioResampler.n-taps"
+
+/**
+ * GstAudioResamplerFilterMode:
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED: Use interpolated filter tables. This
+ *     uses less memory but more CPU and is slightly less accurate but it allows for more
+ *     efficient variable rate resampling with gst_audio_resampler_update().
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_FULL: Use full filter table. This uses more memory
+ *     but less CPU.
+ * @GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO: Automatically choose between interpolated
+ *     and full filter tables.
+ *
+ * Select for the filter tables should be set up.
+ *
+ * Since: 1.10
+ */
+typedef enum {
+  GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED = (0),
+  GST_AUDIO_RESAMPLER_FILTER_MODE_FULL,
+  GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO,
+} GstAudioResamplerFilterMode;
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_MODE:
+ *
+ * GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
+ * constructed.
+ * GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE      "GstAudioResampler.filter-mode"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD:
+ *
+ * G_TYPE_UINT: the amount of memory to use for full filter tables before
+ * switching to interpolated filter tables.
+ * 1048576 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"
+
+/**
+ * GstAudioResamplerFilterInterpolation:
+ * @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: no interpolation
+ * @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: linear interpolation of the
+ *   filter coefficients.
+ * @GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: cubic interpolation of the
+ *   filter coefficients.
+ *
+ * The different filter interpolation methods.
+ *
+ * Since: 1.10
+ */
+typedef enum {
+  GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE = (0),
+  GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR,
+  GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC,
+} GstAudioResamplerFilterInterpolation;
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION:
+ *
+ * GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be
+ *    interpolated.
+ * GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"
+/**
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE:
+ *
+ * G_TYPE_UINT, oversampling to use when interpolating filters
+ * 8 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE "GstAudioResampler.filter-oversample"
+
+/**
+ * GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR:
+ *
+ * G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
+ * rates.
+ * 0.1 is the default.
+ */
+#define GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR "GstAudioResampler.max-phase-error"
+
+/**
+ * GstAudioResamplerMethod:
+ * @GST_AUDIO_RESAMPLER_METHOD_NEAREST: Duplicates the samples when
+ *    upsampling and drops when downsampling
+ * @GST_AUDIO_RESAMPLER_METHOD_LINEAR: Uses linear interpolation to reconstruct
+ *    missing samples and averaging to downsample
+ * @GST_AUDIO_RESAMPLER_METHOD_CUBIC: Uses cubic interpolation
+ * @GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL: Uses Blackman-Nuttall windowed sinc interpolation
+ * @GST_AUDIO_RESAMPLER_METHOD_KAISER: Uses Kaiser windowed sinc interpolation
+ *
+ * Different subsampling and upsampling methods
+ *
+ * Since: 1.10
+ */
+typedef enum {
+  GST_AUDIO_RESAMPLER_METHOD_NEAREST,
+  GST_AUDIO_RESAMPLER_METHOD_LINEAR,
+  GST_AUDIO_RESAMPLER_METHOD_CUBIC,
+  GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL,
+  GST_AUDIO_RESAMPLER_METHOD_KAISER
+} GstAudioResamplerMethod;
+
+/**
+ * GstAudioResamplerFlags:
+ * @GST_AUDIO_RESAMPLER_FLAG_NONE: no flags
+ * @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN: input samples are non-interleaved.
+ *    an array of blocks of samples, one for each channel, should be passed to the
+ *    resample function.
+ * @GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT: output samples are non-interleaved.
+ *    an array of blocks of samples, one for each channel, should be passed to the
+ *    resample function.
+ * @GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE: optimize for dynamic updates of the sample
+ *    rates with gst_audio_resampler_update(). This will select an interpolating filter
+ *    when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.
+ *
+ * Different resampler flags.
+ *
+ * Since: 1.10
+ */
+typedef enum {
+  GST_AUDIO_RESAMPLER_FLAG_NONE                 = (0),
+  GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN   = (1 << 0),
+  GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT  = (1 << 1),
+  GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE        = (1 << 2),
+} GstAudioResamplerFlags;
+
+#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0
+#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
+#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
+
+GST_AUDIO_API
+void           gst_audio_resampler_options_set_quality   (GstAudioResamplerMethod method,
+                                                          guint quality,
+                                                          gint in_rate, gint out_rate,
+                                                          GstStructure *options);
+
+GST_AUDIO_API
+GstAudioResampler * gst_audio_resampler_new              (GstAudioResamplerMethod method,
+                                                          GstAudioResamplerFlags flags,
+                                                          GstAudioFormat format, gint channels,
+                                                          gint in_rate, gint out_rate,
+                                                          GstStructure *options);
+
+GST_AUDIO_API
+void                gst_audio_resampler_free             (GstAudioResampler *resampler);
+
+GST_AUDIO_API
+void                gst_audio_resampler_reset            (GstAudioResampler *resampler);
+
+GST_AUDIO_API
+gboolean            gst_audio_resampler_update           (GstAudioResampler *resampler,
+                                                          gint in_rate, gint out_rate,
+                                                          GstStructure *options);
+
+GST_AUDIO_API
+gsize               gst_audio_resampler_get_out_frames   (GstAudioResampler *resampler,
+                                                          gsize in_frames);
+
+GST_AUDIO_API
+gsize               gst_audio_resampler_get_in_frames    (GstAudioResampler *resampler,
+                                                          gsize out_frames);
+
+GST_AUDIO_API
+gsize               gst_audio_resampler_get_max_latency  (GstAudioResampler *resampler);
+
+GST_AUDIO_API
+void                gst_audio_resampler_resample         (GstAudioResampler * resampler,
+                                                          gpointer in[], gsize in_frames,
+                                                          gpointer out[], gsize out_frames);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_RESAMPLER_H__ */