--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst/volume/gstvolume.c Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,817 @@
+/* -*- c-basic-offset: 2 -*-
+ * vi:si:et:sw=2:sts=8:ts=8:expandtab
+ *
+ * GStreamer
+ * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) 2005 Andy Wingo <wingo@pobox.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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-volume
+ *
+ * <refsect2>
+ * <para>
+ * The volume element changes the volume of the audio data.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
+ * </programlisting>
+ * This pipeline shows that the level of audiotestsrc has been halved
+ * (peak values are around -6 dB and RMS around -9 dB) compared to
+ * the same pipeline without the volume element.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <gst/gst_global.h>
+#include <string.h>
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/interfaces/mixer.h>
+#include <gst/controller/gstcontroller.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+#include <gst/liboil.h>
+
+#include "gstvolume.h"
+
+/* some defines for audio processing */
+/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
+ * we map 1.0 to VOLUME_UNITY_INT*
+ */
+#define VOLUME_UNITY_INT8 32 /* internal int for unity 2^(8-3) */
+#define VOLUME_UNITY_INT8_BIT_SHIFT 5 /* number of bits to shift for unity */
+#define VOLUME_UNITY_INT16 8192 /* internal int for unity 2^(16-3) */
+#define VOLUME_UNITY_INT16_BIT_SHIFT 13 /* number of bits to shift for unity */
+#define VOLUME_UNITY_INT24 2097152 /* internal int for unity 2^(24-3) */
+#define VOLUME_UNITY_INT24_BIT_SHIFT 21 /* number of bits to shift for unity */
+#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
+#define VOLUME_UNITY_INT32_BIT_SHIFT 27
+#define VOLUME_MAX_DOUBLE 10.0
+#define VOLUME_MAX_INT8 G_MAXINT8
+#define VOLUME_MIN_INT8 G_MININT8
+#define VOLUME_MAX_INT16 G_MAXINT16
+#define VOLUME_MIN_INT16 G_MININT16
+#define VOLUME_MAX_INT24 8388607
+#define VOLUME_MIN_INT24 -8388608
+#define VOLUME_MAX_INT32 G_MAXINT32
+#define VOLUME_MIN_INT32 G_MININT32
+
+/* number of steps we use for the mixer interface to go from 0.0 to 1.0 */
+# define VOLUME_STEPS 100
+
+#define GST_CAT_DEFAULT gst_volume_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails volume_details = GST_ELEMENT_DETAILS ("Volume",
+ "Filter/Effect/Audio",
+ "Set volume on audio/raw streams",
+ "Andy Wingo <wingo@pobox.com>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_SILENT,
+ PROP_MUTE,
+ PROP_VOLUME
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) {32, 64}; " \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 8, " \
+ "depth = (int) 8, " \
+ "signed = (bool) TRUE; " \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE; " \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 24, " \
+ "depth = (int) 24, " \
+ "signed = (bool) TRUE; " \
+ "audio/x-raw-int, " \
+ "channels = (int) [ 1, MAX ], " \
+ "rate = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 32, " \
+ "depth = (int) 32, " \
+ "signed = (bool) TRUE"
+
+static void gst_volume_interface_init (GstImplementsInterfaceClass * klass);
+static void gst_volume_mixer_init (GstMixerClass * iface);
+
+#define _init_interfaces(type) \
+ { \
+ static const GInterfaceInfo voliface_info = { \
+ (GInterfaceInitFunc) gst_volume_interface_init, \
+ NULL, \
+ NULL \
+ }; \
+ static const GInterfaceInfo volmixer_info = { \
+ (GInterfaceInitFunc) gst_volume_mixer_init, \
+ NULL, \
+ NULL \
+ }; \
+ \
+ g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, \
+ &voliface_info); \
+ g_type_add_interface_static (type, GST_TYPE_MIXER, &volmixer_info); \
+ }
+
+GST_BOILERPLATE_FULL (GstVolume, gst_volume, GstAudioFilter,
+ GST_TYPE_AUDIO_FILTER, _init_interfaces);
+
+static void volume_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void volume_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void volume_update_volume (const GValue * value, gpointer data);
+static void volume_update_mute (const GValue * value, gpointer data);
+
+static GstFlowReturn volume_transform_ip (GstBaseTransform * base,
+ GstBuffer * outbuf);
+static gboolean volume_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+
+static void volume_process_double (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_float (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int32 (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int32_clamp (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int24 (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int24_clamp (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int16 (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int16_clamp (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int8 (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+static void volume_process_int8_clamp (GstVolume * this, gpointer bytes,
+ guint n_bytes);
+
+
+/* helper functions */
+
+static gboolean
+volume_choose_func (GstVolume * this)
+{
+ this->process = NULL;
+
+ if (GST_AUDIO_FILTER (this)->format.caps == NULL)
+ return FALSE;
+
+ switch (GST_AUDIO_FILTER (this)->format.type) {
+ case GST_BUFTYPE_LINEAR:
+ switch (GST_AUDIO_FILTER (this)->format.width) {
+ case 32:
+ /* only clamp if the gain is greater than 1.0
+ * FIXME: real_vol_i can change while processing the buffer!
+ */
+ if (this->real_vol_i32 > VOLUME_UNITY_INT32)
+ this->process = volume_process_int32_clamp;
+ else
+ this->process = volume_process_int32;
+ break;
+ case 24:
+ /* only clamp if the gain is greater than 1.0
+ * FIXME: real_vol_i can change while processing the buffer!
+ */
+ if (this->real_vol_i24 > VOLUME_UNITY_INT24)
+ this->process = volume_process_int24_clamp;
+ else
+ this->process = volume_process_int24;
+ break;
+ case 16:
+ /* only clamp if the gain is greater than 1.0
+ * FIXME: real_vol_i can change while processing the buffer!
+ */
+ if (this->real_vol_i16 > VOLUME_UNITY_INT16)
+ this->process = volume_process_int16_clamp;
+ else
+ this->process = volume_process_int16;
+ break;
+ case 8:
+ /* only clamp if the gain is greater than 1.0
+ * FIXME: real_vol_i can change while processing the buffer!
+ */
+ if (this->real_vol_i16 > VOLUME_UNITY_INT8)
+ this->process = volume_process_int8_clamp;
+ else
+ this->process = volume_process_int8;
+ break;
+ }
+ break;
+ case GST_BUFTYPE_FLOAT:
+ switch (GST_AUDIO_FILTER (this)->format.width) {
+ case 32:
+ this->process = volume_process_float;
+ break;
+ case 64:
+ this->process = volume_process_double;
+ break;
+ }
+ break;
+ default:
+ break;
+ }
+
+ return (this->process != NULL);
+}
+
+static void
+volume_update_real_volume (GstVolume * this)
+{
+ gboolean passthrough = FALSE;
+
+ if (this->mute) {
+ this->real_vol_f = 0.0;
+ this->real_vol_i8 = this->real_vol_i16 = this->real_vol_i24 =
+ this->real_vol_i32 = 0;
+ } else {
+ this->real_vol_f = this->volume_f;
+ this->real_vol_i8 = this->volume_i8;
+ this->real_vol_i16 = this->volume_i16;
+ this->real_vol_i24 = this->volume_i24;
+ this->real_vol_i32 = this->volume_i32;
+ passthrough = (this->volume_i16 == VOLUME_UNITY_INT16);
+ }
+ if (this->real_vol_f != 0.0)
+ this->silent_buffer = FALSE;
+ volume_choose_func (this);
+ gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (this), passthrough);
+}
+
+/* Mixer interface */
+
+static gboolean
+gst_volume_interface_supported (GstImplementsInterface * iface, GType type)
+{
+ g_assert (type == GST_TYPE_MIXER);
+ return TRUE;
+}
+
+static void
+gst_volume_interface_init (GstImplementsInterfaceClass * klass)
+{
+ klass->supported = gst_volume_interface_supported;
+}
+
+static const GList *
+gst_volume_list_tracks (GstMixer * mixer)
+{
+ GstVolume *this = GST_VOLUME (mixer);
+
+ g_return_val_if_fail (this != NULL, NULL);
+ g_return_val_if_fail (GST_IS_VOLUME (this), NULL);
+
+ return this->tracklist;
+}
+
+static void
+gst_volume_set_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
+{
+ GstVolume *this = GST_VOLUME (mixer);
+
+ g_return_if_fail (this != NULL);
+ g_return_if_fail (GST_IS_VOLUME (this));
+
+ this->volume_f = (gfloat) volumes[0] / VOLUME_STEPS;
+ this->volume_i32 = this->volume_f * VOLUME_UNITY_INT32;
+ this->volume_i24 = this->volume_f * VOLUME_UNITY_INT24;
+ this->volume_i16 = this->volume_f * VOLUME_UNITY_INT16;
+ this->volume_i8 = this->volume_f * VOLUME_UNITY_INT8;
+
+ volume_update_real_volume (this);
+}
+
+static void
+gst_volume_get_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
+{
+ GstVolume *this = GST_VOLUME (mixer);
+
+ g_return_if_fail (this != NULL);
+ g_return_if_fail (GST_IS_VOLUME (this));
+
+ volumes[0] = (gint) this->volume_f * VOLUME_STEPS;
+}
+
+static void
+gst_volume_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
+{
+ GstVolume *this = GST_VOLUME (mixer);
+
+ g_return_if_fail (this != NULL);
+ g_return_if_fail (GST_IS_VOLUME (this));
+
+ this->mute = mute;
+
+ volume_update_real_volume (this);
+}
+
+static void
+gst_volume_mixer_init (GstMixerClass * klass)
+{
+ GST_MIXER_TYPE (klass) = GST_MIXER_SOFTWARE;
+
+ /* default virtual functions */
+ klass->list_tracks = gst_volume_list_tracks;
+ klass->set_volume = gst_volume_set_volume;
+ klass->get_volume = gst_volume_get_volume;
+ klass->set_mute = gst_volume_set_mute;
+}
+
+/* Element class */
+
+static void
+gst_volume_dispose (GObject * object)
+{
+ GstVolume *volume = GST_VOLUME (object);
+
+ if (volume->tracklist) {
+ if (volume->tracklist->data)
+ g_object_unref (volume->tracklist->data);
+ g_list_free (volume->tracklist);
+ volume->tracklist = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_volume_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+ GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (g_class);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &volume_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (filter_class, caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_volume_class_init (GstVolumeClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseTransformClass *trans_class;
+ GstAudioFilterClass *filter_class;
+
+ gobject_class = (GObjectClass *) klass;
+ trans_class = (GstBaseTransformClass *) klass;
+ filter_class = (GstAudioFilterClass *) (klass);
+
+ gobject_class->set_property = volume_set_property;
+ gobject_class->get_property = volume_get_property;
+ gobject_class->dispose = gst_volume_dispose;
+
+ g_object_class_install_property (gobject_class, PROP_MUTE,
+ g_param_spec_boolean ("mute", "Mute", "mute channel",
+ FALSE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ g_object_class_install_property (gobject_class, PROP_VOLUME,
+ g_param_spec_double ("volume", "Volume", "volume factor",
+ 0.0, VOLUME_MAX_DOUBLE, 1.0,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
+
+ trans_class->transform_ip = GST_DEBUG_FUNCPTR (volume_transform_ip);
+ filter_class->setup = GST_DEBUG_FUNCPTR (volume_setup);
+}
+
+static void
+gst_volume_init (GstVolume * this, GstVolumeClass * g_class)
+{
+ GstMixerTrack *track = NULL;
+
+ this->mute = FALSE;
+ this->volume_i8 = this->real_vol_i8 = VOLUME_UNITY_INT8;
+ this->volume_i16 = this->real_vol_i16 = VOLUME_UNITY_INT16;
+ this->volume_i24 = this->real_vol_i24 = VOLUME_UNITY_INT24;
+ this->volume_i32 = this->real_vol_i32 = VOLUME_UNITY_INT32;
+ this->volume_f = this->real_vol_f = 1.0;
+ this->tracklist = NULL;
+
+ track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
+
+ if (GST_IS_MIXER_TRACK (track)) {
+ track->label = g_strdup ("volume");
+ track->num_channels = 1;
+ track->min_volume = 0;
+ track->max_volume = VOLUME_STEPS;
+ track->flags = GST_MIXER_TRACK_SOFTWARE;
+ this->tracklist = g_list_append (this->tracklist, track);
+ }
+
+ gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
+}
+
+/* NOTE: although it might be tempting to have volume_process_mute() which uses
+ * memset(bytes, 0, nbytes) for the vol=0 case, this has the downside that
+ * unmuting would only take place after processing a buffer.
+ */
+
+static void
+volume_process_double (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gdouble *data = (gdouble *) bytes;
+ guint i, num_samples = n_bytes / sizeof (gdouble);
+
+ for (i = 0; i < num_samples; i++) {
+ *data++ *= this->real_vol_f;
+ }
+}
+
+static void
+volume_process_float (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gfloat *data = (gfloat *) bytes;
+ guint num_samples = n_bytes / sizeof (gfloat);
+
+ /*
+ guint i;
+ for (i = 0; i < num_samples; i++) {
+ *data++ *= this->real_vol_f;
+ }
+ */
+ /* time gst-launch 2>/dev/null audiotestsrc wave=7 num-buffers=10000 ! audio/x-raw-float ! volume volume=1.5 ! fakesink
+ * goes from 0m0.850s -> 0m0.717s with liboil
+ */
+ oil_scalarmultiply_f32_ns (data, data, &this->real_vol_f, num_samples);
+}
+
+static void
+volume_process_int32 (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint *data = (gint *) bytes;
+ guint i, num_samples;
+ gint64 val;
+
+ num_samples = n_bytes / sizeof (gint);
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint64) * data;
+ val = (((gint64) this->real_vol_i32 * val) >> VOLUME_UNITY_INT32_BIT_SHIFT);
+ *data++ = (gint32) val;
+ }
+}
+
+static void
+volume_process_int32_clamp (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint *data = (gint *) bytes;
+ guint i, num_samples;
+ gint64 val;
+
+ num_samples = n_bytes / sizeof (gint);
+
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint64) * data;
+ val = (((gint64) this->real_vol_i32 * val) >> VOLUME_UNITY_INT32_BIT_SHIFT);
+ *data++ = (gint32) CLAMP (val, VOLUME_MIN_INT32, VOLUME_MAX_INT32);
+ }
+}
+
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+#define get_unaligned_i24(_x) ( (((guint8*)_x)[0]) | ((((guint8*)_x)[1]) << 8) | ((((gint8*)_x)[2]) << 16) )
+#define write_unaligned_u24(_x,samp) do { *(_x)++ = samp & 0xFF; *(_x)++ = (samp >> 8) & 0xFF; *(_x)++ = (samp >> 16) & 0xFF; } while (0)
+#else /* BIG ENDIAN */
+#define get_unaligned_i24(_x) ( (((guint8*)_x)[2]) | ((((guint8*)_x)[1]) << 8) | ((((gint8*)_x)[0]) << 16) )
+#define write_unaligned_u24(_x,samp) do { *(_x)++ = (samp >> 16) & 0xFF; *(_x)++ = (samp >> 8) & 0xFF; *(_x)++ = samp & 0xFF; } while (0)
+#endif
+
+static void
+volume_process_int24 (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint8 *data = (gint8 *) bytes; /* treat the data as a byte stream */
+ guint i, num_samples;
+ guint32 samp;
+ gint64 val;
+
+ num_samples = n_bytes / (sizeof (gint8) * 3);
+ for (i = 0; i < num_samples; i++) {
+ samp = get_unaligned_i24 (data);
+
+ val = (gint32) samp;
+ val = (((gint64) this->real_vol_i24 * val) >> VOLUME_UNITY_INT24_BIT_SHIFT);
+ samp = (guint32) val;
+
+ /* write the value back into the stream */
+ write_unaligned_u24 (data, samp);
+ }
+}
+
+static void
+volume_process_int24_clamp (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint8 *data = (gint8 *) bytes; /* treat the data as a byte stream */
+ guint i, num_samples;
+ guint32 samp;
+ gint64 val;
+
+ num_samples = n_bytes / (sizeof (gint8) * 3);
+ for (i = 0; i < num_samples; i++) {
+ samp = get_unaligned_i24 (data);
+
+ val = (gint32) samp;
+ val = (((gint64) this->real_vol_i24 * val) >> VOLUME_UNITY_INT24_BIT_SHIFT);
+ samp = (guint32) CLAMP (val, VOLUME_MIN_INT24, VOLUME_MAX_INT24);
+
+ /* write the value back into the stream */
+ write_unaligned_u24 (data, samp);
+ }
+}
+
+static void
+volume_process_int16 (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint16 *data = (gint16 *) bytes;
+ guint num_samples = n_bytes / sizeof (gint16);
+
+#if 1
+ guint i;
+ gint val;
+
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint) * data;
+ *data++ =
+ (gint16) ((this->real_vol_i16 * val) >> VOLUME_UNITY_INT16_BIT_SHIFT);
+ }
+#else
+ /* FIXME: need oil_scalarmultiply_s16_ns ?
+ * https://bugs.freedesktop.org/show_bug.cgi?id=7060
+ * code below
+ * - crashes :/
+ * - real_vol_i is scaled by VOLUME_UNITY_INT16 and needs the bitshift
+ * time gst-launch 2>/dev/null audiotestsrc wave=7 num-buffers=100 ! volume volume=1.5 ! fakesink
+ */
+ oil_scalarmult_s16 (data, 0, data, 0,
+ ((gint16 *) (void *) (&this->real_vol_i)), num_samples);
+#endif
+}
+
+static void
+volume_process_int16_clamp (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint16 *data = (gint16 *) bytes;
+ guint i, num_samples;
+ gint val;
+
+ num_samples = n_bytes / sizeof (gint16);
+
+ /* FIXME: oil_scalarmultiply_s16_ns ?
+ * https://bugs.freedesktop.org/show_bug.cgi?id=7060
+ */
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint) * data;
+ *data++ =
+ (gint16) CLAMP ((this->real_vol_i16 *
+ val) >> VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MIN_INT16,
+ VOLUME_MAX_INT16);
+ }
+}
+
+static void
+volume_process_int8 (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint8 *data = (gint8 *) bytes;
+ guint num_samples = n_bytes / sizeof (gint8);
+ guint i;
+ gint val;
+
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint) * data;
+ *data++ =
+ (gint8) ((this->real_vol_i8 * val) >> VOLUME_UNITY_INT8_BIT_SHIFT);
+ }
+}
+
+static void
+volume_process_int8_clamp (GstVolume * this, gpointer bytes, guint n_bytes)
+{
+ gint8 *data = (gint8 *) bytes;
+ guint i, num_samples;
+ gint val;
+
+ num_samples = n_bytes / sizeof (gint8);
+
+ for (i = 0; i < num_samples; i++) {
+ /* we use bitshifting instead of dividing by UNITY_INT for speed */
+ val = (gint) * data;
+ *data++ =
+ (gint8) CLAMP ((this->real_vol_i8 *
+ val) >> VOLUME_UNITY_INT8_BIT_SHIFT, VOLUME_MIN_INT8,
+ VOLUME_MAX_INT8);
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+
+/* get notified of caps and plug in the correct process function */
+static gboolean
+volume_setup (GstAudioFilter * filter, GstRingBufferSpec * format)
+{
+ GstVolume *this = GST_VOLUME (filter);
+
+ if (volume_choose_func (this)) {
+ return TRUE;
+ } else {
+ GST_ELEMENT_ERROR (this, CORE, NEGOTIATION,
+ ("Invalid incoming format"), (NULL));
+ return FALSE;
+ }
+}
+
+/* call the plugged-in process function for this instance
+ * needs to be done with this indirection since volume_transform is
+ * a class-global method
+ */
+static GstFlowReturn
+volume_transform_ip (GstBaseTransform * base, GstBuffer * outbuf)
+{
+ GstVolume *this = GST_VOLUME (base);
+ GstClockTime timestamp;
+
+ /* FIXME: if controllers are bound, subdivide GST_BUFFER_SIZE into small
+ * chunks for smooth fades, what is small? 1/10th sec.
+ */
+ timestamp = GST_BUFFER_TIMESTAMP (outbuf);
+ timestamp =
+ gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
+
+ GST_DEBUG_OBJECT (base, "sync to %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (timestamp));
+
+ if (GST_CLOCK_TIME_IS_VALID (timestamp))
+ gst_object_sync_values (G_OBJECT (this), timestamp);
+
+ /* don't process data in passthrough-mode */
+ if (gst_base_transform_is_passthrough (base) ||
+ GST_BUFFER_FLAG_IS_SET (outbuf, GST_BUFFER_FLAG_GAP))
+ return GST_FLOW_OK;
+
+ if (this->real_vol_f == 0.0)
+ this->silent_buffer = TRUE;
+
+ this->process (this, GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
+
+ if (this->silent_buffer)
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
+ this->silent_buffer = FALSE;
+
+ return GST_FLOW_OK;
+}
+
+static void
+volume_update_mute (const GValue * value, gpointer data)
+{
+ GstVolume *this = (GstVolume *) data;
+
+ g_return_if_fail (GST_IS_VOLUME (this));
+
+ if (G_VALUE_HOLDS_BOOLEAN (value)) {
+ this->mute = g_value_get_boolean (value);
+ } else if (G_VALUE_HOLDS_INT (value)) {
+ this->mute = (g_value_get_int (value) == 1);
+ }
+
+ volume_update_real_volume (this);
+}
+
+static void
+volume_update_volume (const GValue * value, gpointer data)
+{
+ GstVolume *this = (GstVolume *) data;
+
+ g_return_if_fail (GST_IS_VOLUME (this));
+
+ this->volume_f = g_value_get_double (value);
+ this->volume_i8 = this->volume_f * VOLUME_UNITY_INT8;
+ this->volume_i16 = this->volume_f * VOLUME_UNITY_INT16;
+ this->volume_i24 = this->volume_f * VOLUME_UNITY_INT24;
+ this->volume_i32 = this->volume_f * VOLUME_UNITY_INT32;
+
+ volume_update_real_volume (this);
+}
+
+static void
+volume_set_property (GObject * object, guint prop_id, const GValue * value,
+ GParamSpec * pspec)
+{
+ GstVolume *this = GST_VOLUME (object);
+
+ switch (prop_id) {
+ case PROP_MUTE:
+ volume_update_mute (value, this);
+ break;
+ case PROP_VOLUME:
+ volume_update_volume (value, this);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+volume_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstVolume *this = GST_VOLUME (object);
+
+ switch (prop_id) {
+ case PROP_MUTE:
+ g_value_set_boolean (value, this->mute);
+ break;
+ case PROP_VOLUME:
+ g_value_set_double (value, this->volume_f);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifndef __SYMBIAN32__
+ oil_init ();
+#endif
+
+ /* initialize gst controller library */
+ gst_controller_init (NULL, NULL);
+
+ return gst_element_register (plugin, "volume", GST_RANK_NONE,
+ GST_TYPE_VOLUME);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "volume",
+ "plugin for controlling audio volume",
+ plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
+
+
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+GstPluginDesc* _GST_PLUGIN_DESC()
+{
+ return &gst_plugin_desc;
+}
+