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cd5f49f47f
Original commit message from CVS: * gst/volume/gstvolume.c: (volume_process_double): Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
810 lines
24 KiB
C
810 lines
24 KiB
C
/* -*- c-basic-offset: 2 -*-
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* vi:si:et:sw=2:sts=8:ts=8:expandtab
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*
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* GStreamer
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* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) 2005 Andy Wingo <wingo@pobox.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-volume
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*
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* <refsect2>
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* <para>
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* The volume element changes the volume of the audio data.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
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* </programlisting>
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* This pipeline shows that the level of audiotestsrc has been halved
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* (peak values are around -6 dB and RMS around -9 dB) compared to
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* the same pipeline without the volume element.
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* </para>
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/interfaces/mixer.h>
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#include <gst/controller/gstcontroller.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <liboil/liboil.h>
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#include "gstvolume.h"
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/* some defines for audio processing */
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/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
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* we map 1.0 to VOLUME_UNITY_INT*
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*/
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#define VOLUME_UNITY_INT8 32 /* internal int for unity 2^(8-3) */
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#define VOLUME_UNITY_INT8_BIT_SHIFT 5 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT16 8192 /* internal int for unity 2^(16-3) */
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#define VOLUME_UNITY_INT16_BIT_SHIFT 13 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT24 2097152 /* internal int for unity 2^(24-3) */
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#define VOLUME_UNITY_INT24_BIT_SHIFT 21 /* number of bits to shift for unity */
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#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
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#define VOLUME_UNITY_INT32_BIT_SHIFT 27
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#define VOLUME_MAX_DOUBLE 10.0
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#define VOLUME_MAX_INT8 G_MAXINT8
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#define VOLUME_MIN_INT8 G_MININT8
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#define VOLUME_MAX_INT16 G_MAXINT16
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#define VOLUME_MIN_INT16 G_MININT16
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#define VOLUME_MAX_INT24 8388607
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#define VOLUME_MIN_INT24 -8388608
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#define VOLUME_MAX_INT32 G_MAXINT32
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#define VOLUME_MIN_INT32 G_MININT32
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/* number of steps we use for the mixer interface to go from 0.0 to 1.0 */
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# define VOLUME_STEPS 100
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#define GST_CAT_DEFAULT gst_volume_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static const GstElementDetails volume_details = GST_ELEMENT_DETAILS ("Volume",
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"Filter/Effect/Audio",
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"Set volume on audio/raw streams",
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"Andy Wingo <wingo@pobox.com>");
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_SILENT,
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PROP_MUTE,
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PROP_VOLUME
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};
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#define ALLOWED_CAPS \
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"audio/x-raw-float, " \
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"rate = (int) [ 1, MAX ], " \
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"channels = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) {32, 64}; " \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 8, " \
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"depth = (int) 8, " \
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"signed = (bool) TRUE; " \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (bool) TRUE; " \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 24, " \
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"depth = (int) 24, " \
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"signed = (bool) TRUE; " \
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"audio/x-raw-int, " \
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"channels = (int) [ 1, MAX ], " \
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"rate = (int) [ 1, MAX ], " \
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"endianness = (int) BYTE_ORDER, " \
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"width = (int) 32, " \
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"depth = (int) 32, " \
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"signed = (bool) TRUE"
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static void gst_volume_interface_init (GstImplementsInterfaceClass * klass);
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static void gst_volume_mixer_init (GstMixerClass * iface);
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#define _init_interfaces(type) \
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{ \
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static const GInterfaceInfo voliface_info = { \
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(GInterfaceInitFunc) gst_volume_interface_init, \
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NULL, \
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NULL \
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}; \
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static const GInterfaceInfo volmixer_info = { \
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(GInterfaceInitFunc) gst_volume_mixer_init, \
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NULL, \
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NULL \
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}; \
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\
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g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, \
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&voliface_info); \
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g_type_add_interface_static (type, GST_TYPE_MIXER, &volmixer_info); \
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}
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GST_BOILERPLATE_FULL (GstVolume, gst_volume, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, _init_interfaces);
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static void volume_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void volume_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void volume_update_volume (const GValue * value, gpointer data);
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static void volume_update_mute (const GValue * value, gpointer data);
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static GstFlowReturn volume_transform_ip (GstBaseTransform * base,
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GstBuffer * outbuf);
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static gboolean volume_setup (GstAudioFilter * filter,
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GstRingBufferSpec * format);
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static void volume_process_double (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_float (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int32 (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int32_clamp (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int24 (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int24_clamp (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int16 (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int16_clamp (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int8 (GstVolume * this, gpointer bytes,
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guint n_bytes);
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static void volume_process_int8_clamp (GstVolume * this, gpointer bytes,
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guint n_bytes);
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/* helper functions */
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static gboolean
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volume_choose_func (GstVolume * this)
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{
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this->process = NULL;
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if (GST_AUDIO_FILTER (this)->format.caps == NULL)
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return FALSE;
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switch (GST_AUDIO_FILTER (this)->format.type) {
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case GST_BUFTYPE_LINEAR:
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switch (GST_AUDIO_FILTER (this)->format.width) {
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case 32:
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/* only clamp if the gain is greater than 1.0
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* FIXME: real_vol_i can change while processing the buffer!
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*/
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if (this->real_vol_i32 > VOLUME_UNITY_INT32)
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this->process = volume_process_int32_clamp;
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else
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this->process = volume_process_int32;
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break;
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case 24:
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/* only clamp if the gain is greater than 1.0
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* FIXME: real_vol_i can change while processing the buffer!
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*/
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if (this->real_vol_i24 > VOLUME_UNITY_INT24)
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this->process = volume_process_int24_clamp;
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else
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this->process = volume_process_int24;
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break;
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case 16:
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/* only clamp if the gain is greater than 1.0
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* FIXME: real_vol_i can change while processing the buffer!
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*/
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if (this->real_vol_i16 > VOLUME_UNITY_INT16)
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this->process = volume_process_int16_clamp;
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else
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this->process = volume_process_int16;
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break;
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case 8:
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/* only clamp if the gain is greater than 1.0
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* FIXME: real_vol_i can change while processing the buffer!
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*/
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if (this->real_vol_i16 > VOLUME_UNITY_INT8)
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this->process = volume_process_int8_clamp;
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else
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this->process = volume_process_int8;
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break;
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}
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break;
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case GST_BUFTYPE_FLOAT:
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switch (GST_AUDIO_FILTER (this)->format.width) {
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case 32:
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this->process = volume_process_float;
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break;
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case 64:
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this->process = volume_process_double;
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break;
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}
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break;
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default:
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break;
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}
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return (this->process != NULL);
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}
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static void
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volume_update_real_volume (GstVolume * this)
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{
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gboolean passthrough = FALSE;
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if (this->mute) {
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this->real_vol_f = 0.0;
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this->real_vol_i8 = this->real_vol_i16 = this->real_vol_i24 =
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this->real_vol_i32 = 0;
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} else {
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this->real_vol_f = this->volume_f;
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this->real_vol_i8 = this->volume_i8;
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this->real_vol_i16 = this->volume_i16;
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this->real_vol_i24 = this->volume_i24;
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this->real_vol_i32 = this->volume_i32;
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passthrough = (this->volume_i16 == VOLUME_UNITY_INT16);
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}
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if (this->real_vol_f != 0.0)
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this->silent_buffer = FALSE;
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volume_choose_func (this);
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gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (this), passthrough);
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}
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/* Mixer interface */
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static gboolean
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gst_volume_interface_supported (GstImplementsInterface * iface, GType type)
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{
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g_assert (type == GST_TYPE_MIXER);
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return TRUE;
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}
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static void
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gst_volume_interface_init (GstImplementsInterfaceClass * klass)
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{
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klass->supported = gst_volume_interface_supported;
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}
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static const GList *
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gst_volume_list_tracks (GstMixer * mixer)
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{
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GstVolume *this = GST_VOLUME (mixer);
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g_return_val_if_fail (this != NULL, NULL);
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g_return_val_if_fail (GST_IS_VOLUME (this), NULL);
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return this->tracklist;
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}
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static void
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gst_volume_set_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
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{
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GstVolume *this = GST_VOLUME (mixer);
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g_return_if_fail (this != NULL);
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g_return_if_fail (GST_IS_VOLUME (this));
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this->volume_f = (gfloat) volumes[0] / VOLUME_STEPS;
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this->volume_i32 = this->volume_f * VOLUME_UNITY_INT32;
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this->volume_i24 = this->volume_f * VOLUME_UNITY_INT24;
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this->volume_i16 = this->volume_f * VOLUME_UNITY_INT16;
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this->volume_i8 = this->volume_f * VOLUME_UNITY_INT8;
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volume_update_real_volume (this);
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}
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static void
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gst_volume_get_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
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{
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GstVolume *this = GST_VOLUME (mixer);
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g_return_if_fail (this != NULL);
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g_return_if_fail (GST_IS_VOLUME (this));
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volumes[0] = (gint) this->volume_f * VOLUME_STEPS;
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}
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static void
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gst_volume_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
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{
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GstVolume *this = GST_VOLUME (mixer);
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g_return_if_fail (this != NULL);
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g_return_if_fail (GST_IS_VOLUME (this));
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this->mute = mute;
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volume_update_real_volume (this);
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}
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static void
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gst_volume_mixer_init (GstMixerClass * klass)
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{
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GST_MIXER_TYPE (klass) = GST_MIXER_SOFTWARE;
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/* default virtual functions */
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klass->list_tracks = gst_volume_list_tracks;
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klass->set_volume = gst_volume_set_volume;
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klass->get_volume = gst_volume_get_volume;
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klass->set_mute = gst_volume_set_mute;
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}
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/* Element class */
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static void
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gst_volume_dispose (GObject * object)
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{
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GstVolume *volume = GST_VOLUME (object);
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if (volume->tracklist) {
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if (volume->tracklist->data)
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g_object_unref (volume->tracklist->data);
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g_list_free (volume->tracklist);
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volume->tracklist = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_volume_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (g_class);
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GstCaps *caps;
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gst_element_class_set_details (element_class, &volume_details);
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (filter_class, caps);
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gst_caps_unref (caps);
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}
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static void
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gst_volume_class_init (GstVolumeClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseTransformClass *trans_class;
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GstAudioFilterClass *filter_class;
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gobject_class = (GObjectClass *) klass;
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trans_class = (GstBaseTransformClass *) klass;
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filter_class = (GstAudioFilterClass *) (klass);
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gobject_class->set_property = volume_set_property;
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gobject_class->get_property = volume_get_property;
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gobject_class->dispose = gst_volume_dispose;
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g_object_class_install_property (gobject_class, PROP_MUTE,
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g_param_spec_boolean ("mute", "Mute", "mute channel",
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FALSE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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g_object_class_install_property (gobject_class, PROP_VOLUME,
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g_param_spec_double ("volume", "Volume", "volume factor",
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0.0, VOLUME_MAX_DOUBLE, 1.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
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trans_class->transform_ip = GST_DEBUG_FUNCPTR (volume_transform_ip);
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filter_class->setup = GST_DEBUG_FUNCPTR (volume_setup);
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}
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static void
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gst_volume_init (GstVolume * this, GstVolumeClass * g_class)
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{
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GstMixerTrack *track = NULL;
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this->mute = FALSE;
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this->volume_i8 = this->real_vol_i8 = VOLUME_UNITY_INT8;
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this->volume_i16 = this->real_vol_i16 = VOLUME_UNITY_INT16;
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this->volume_i24 = this->real_vol_i24 = VOLUME_UNITY_INT24;
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this->volume_i32 = this->real_vol_i32 = VOLUME_UNITY_INT32;
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this->volume_f = this->real_vol_f = 1.0;
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this->tracklist = NULL;
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track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
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if (GST_IS_MIXER_TRACK (track)) {
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track->label = g_strdup ("volume");
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track->num_channels = 1;
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track->min_volume = 0;
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track->max_volume = VOLUME_STEPS;
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track->flags = GST_MIXER_TRACK_SOFTWARE;
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this->tracklist = g_list_append (this->tracklist, track);
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}
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (this), TRUE);
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}
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/* NOTE: although it might be tempting to have volume_process_mute() which uses
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* memset(bytes, 0, nbytes) for the vol=0 case, this has the downside that
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* unmuting would only take place after processing a buffer.
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*/
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static void
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volume_process_double (GstVolume * this, gpointer bytes, guint n_bytes)
|
|
{
|
|
gdouble *data = (gdouble *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gdouble);
|
|
|
|
#ifdef oil_scalarmultiply_f64_ns
|
|
gdouble vol = this->real_vol_f;
|
|
|
|
oil_scalarmultiply_f64_ns (data, data, &vol, num_samples);
|
|
#else
|
|
for (i = 0; i < num_samples; i++) {
|
|
*data++ *= this->real_vol_f;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
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)
|
|
{
|
|
oil_init ();
|
|
|
|
/* 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);
|