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4a0d435053
Going over integer arithmetic will lead to minimal rounding errors, leading to +/-1 changes for volume==1.0. Implement the controlled processing with floating point arithmetic, which was already done for the C versions anyway.
1048 lines
31 KiB
C
1048 lines
31 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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* Copyright (C) 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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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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* The volume element changes the volume of the audio data.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
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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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* </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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#ifdef HAVE_ORC
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#include <orc/orcfunctions.h>
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#else
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#define orc_memset memset
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#endif
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#include "gstvolumeorc.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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/* 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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#define DEFAULT_PROP_MUTE FALSE
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#define DEFAULT_PROP_VOLUME 1.0
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enum
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{
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PROP_0,
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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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static const GInterfaceInfo svol_info = { \
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NULL, \
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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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g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_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_before_transform (GstBaseTransform * base,
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GstBuffer * buffer);
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static GstFlowReturn volume_transform_ip (GstBaseTransform * base,
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GstBuffer * outbuf);
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static gboolean volume_stop (GstBaseTransform * base);
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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 * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_double (GstVolume * self, gpointer bytes,
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gdouble * volume, guint channels, guint n_bytes);
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static void volume_process_float (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_float (GstVolume * self, gpointer bytes,
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gdouble * volume, guint channels, guint n_bytes);
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static void volume_process_int32 (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_int32_clamp (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_int32_clamp (GstVolume * self,
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gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
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static void volume_process_int24 (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_int24_clamp (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_int24_clamp (GstVolume * self,
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gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
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static void volume_process_int16 (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_int16_clamp (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_int16_clamp (GstVolume * self,
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gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
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static void volume_process_int8 (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_int8_clamp (GstVolume * self, gpointer bytes,
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guint n_bytes);
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static void volume_process_controlled_int8_clamp (GstVolume * self,
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gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
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/* helper functions */
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static gboolean
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volume_choose_func (GstVolume * self)
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{
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self->process = NULL;
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self->process_controlled = NULL;
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if (GST_AUDIO_FILTER (self)->format.caps == NULL)
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return FALSE;
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switch (GST_AUDIO_FILTER (self)->format.type) {
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case GST_BUFTYPE_LINEAR:
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switch (GST_AUDIO_FILTER (self)->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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*/
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if (self->current_vol_i32 > VOLUME_UNITY_INT32) {
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self->process = volume_process_int32_clamp;
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} else {
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self->process = volume_process_int32;
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}
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self->process_controlled = volume_process_controlled_int32_clamp;
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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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*/
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if (self->current_vol_i24 > VOLUME_UNITY_INT24) {
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self->process = volume_process_int24_clamp;
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} else {
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self->process = volume_process_int24;
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}
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self->process_controlled = volume_process_controlled_int24_clamp;
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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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*/
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if (self->current_vol_i16 > VOLUME_UNITY_INT16) {
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self->process = volume_process_int16_clamp;
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} else {
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self->process = volume_process_int16;
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}
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self->process_controlled = volume_process_controlled_int16_clamp;
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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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*/
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if (self->current_vol_i16 > VOLUME_UNITY_INT8) {
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self->process = volume_process_int8_clamp;
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} else {
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self->process = volume_process_int8;
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}
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self->process_controlled = volume_process_controlled_int8_clamp;
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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 (self)->format.width) {
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case 32:
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self->process = volume_process_float;
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self->process_controlled = volume_process_controlled_float;
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break;
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case 64:
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self->process = volume_process_double;
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self->process_controlled = volume_process_controlled_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 (self->process != NULL);
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}
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static gboolean
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volume_update_volume (GstVolume * self, gfloat volume, gboolean mute)
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{
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gboolean passthrough;
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gboolean res;
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GstController *controller;
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GST_DEBUG_OBJECT (self, "configure mute %d, volume %f", mute, volume);
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if (mute) {
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self->current_mute = TRUE;
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self->current_volume = 0.0;
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self->current_vol_i8 = 0;
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self->current_vol_i16 = 0;
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self->current_vol_i24 = 0;
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self->current_vol_i32 = 0;
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passthrough = FALSE;
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} else {
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self->current_mute = FALSE;
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self->current_volume = volume;
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self->current_vol_i8 = volume * VOLUME_UNITY_INT8;
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self->current_vol_i16 = volume * VOLUME_UNITY_INT16;
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self->current_vol_i24 = volume * VOLUME_UNITY_INT24;
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self->current_vol_i32 = volume * VOLUME_UNITY_INT32;
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passthrough = (self->current_vol_i16 == VOLUME_UNITY_INT16);
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}
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/* If a controller is used, never use passthrough mode
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* because the property can change from 1.0 to something
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* else in the middle of a buffer.
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*/
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controller = gst_object_get_controller (G_OBJECT (self));
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passthrough = passthrough && (controller == NULL);
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GST_DEBUG_OBJECT (self, "set passthrough %d", passthrough);
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gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (self), passthrough);
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res = self->negotiated = volume_choose_func (self);
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return res;
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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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return (type == GST_TYPE_MIXER || type == GST_TYPE_STREAM_VOLUME);
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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 *self = GST_VOLUME (mixer);
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g_return_val_if_fail (self != NULL, NULL);
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g_return_val_if_fail (GST_IS_VOLUME (self), NULL);
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return self->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 *self = GST_VOLUME (mixer);
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g_return_if_fail (self != NULL);
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g_return_if_fail (GST_IS_VOLUME (self));
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GST_OBJECT_LOCK (self);
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self->volume = (gfloat) volumes[0] / VOLUME_STEPS;
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GST_OBJECT_UNLOCK (self);
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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 *self = GST_VOLUME (mixer);
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g_return_if_fail (self != NULL);
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g_return_if_fail (GST_IS_VOLUME (self));
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GST_OBJECT_LOCK (self);
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volumes[0] = (gint) self->volume * VOLUME_STEPS;
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GST_OBJECT_UNLOCK (self);
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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 *self = GST_VOLUME (mixer);
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g_return_if_fail (self != NULL);
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g_return_if_fail (GST_IS_VOLUME (self));
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GST_OBJECT_LOCK (self);
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self->mute = mute;
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GST_OBJECT_UNLOCK (self);
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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_simple (element_class, "Volume",
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"Filter/Effect/Audio",
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"Set volume on audio/raw streams", "Andy Wingo <wingo@pobox.com>");
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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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DEFAULT_PROP_MUTE,
|
|
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_VOLUME,
|
|
g_param_spec_double ("volume", "Volume", "volume factor, 1.0=100%",
|
|
0.0, VOLUME_MAX_DOUBLE, DEFAULT_PROP_VOLUME,
|
|
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
trans_class->before_transform = GST_DEBUG_FUNCPTR (volume_before_transform);
|
|
trans_class->transform_ip = GST_DEBUG_FUNCPTR (volume_transform_ip);
|
|
trans_class->stop = GST_DEBUG_FUNCPTR (volume_stop);
|
|
filter_class->setup = GST_DEBUG_FUNCPTR (volume_setup);
|
|
}
|
|
|
|
static void
|
|
gst_volume_init (GstVolume * self, GstVolumeClass * g_class)
|
|
{
|
|
GstMixerTrack *track = NULL;
|
|
|
|
self->mute = DEFAULT_PROP_MUTE;;
|
|
self->volume = DEFAULT_PROP_VOLUME;
|
|
|
|
self->tracklist = NULL;
|
|
self->negotiated = FALSE;
|
|
|
|
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;
|
|
self->tracklist = g_list_append (self->tracklist, track);
|
|
}
|
|
|
|
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (self), TRUE);
|
|
}
|
|
|
|
static void
|
|
volume_process_double (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gdouble *data = (gdouble *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gdouble);
|
|
|
|
orc_scalarmultiply_f64_ns (data, self->current_volume, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_controlled_double (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gdouble *data = (gdouble *) bytes;
|
|
guint num_samples = n_bytes / (sizeof (gdouble) * channels);
|
|
guint i, j;
|
|
gdouble vol;
|
|
|
|
if (channels == 1) {
|
|
orc_process_controlled_f64_1ch (data, volume, num_samples);
|
|
} else {
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
*data++ *= vol;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
volume_process_float (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gfloat *data = (gfloat *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gfloat);
|
|
|
|
orc_scalarmultiply_f32_ns (data, self->current_volume, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_controlled_float (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gfloat *data = (gfloat *) bytes;
|
|
guint num_samples = n_bytes / (sizeof (gfloat) * channels);
|
|
guint i, j;
|
|
gdouble vol;
|
|
|
|
if (channels == 1) {
|
|
orc_process_controlled_f32_1ch (data, volume, num_samples);
|
|
} else if (channels == 2) {
|
|
orc_process_controlled_f32_2ch (data, volume, num_samples);
|
|
} else {
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
*data++ *= vol;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
volume_process_int32 (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint32 *data = (gint32 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT32_BIT_SHIFT == 27);
|
|
orc_process_int32 (data, self->current_vol_i32, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_int32_clamp (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint32 *data = (gint32 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT32_BIT_SHIFT == 27);
|
|
|
|
orc_process_int32_clamp (data, self->current_vol_i32, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_controlled_int32_clamp (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gint32 *data = (gint32 *) bytes;
|
|
guint i, j;
|
|
guint num_samples = n_bytes / (sizeof (gint32) * channels);
|
|
gdouble vol, val;
|
|
|
|
if (channels == 1) {
|
|
orc_process_controlled_int32_1ch (data, volume, num_samples);
|
|
} else {
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
val = *data * vol;
|
|
*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) \
|
|
G_STMT_START { \
|
|
*(_x)++ = samp & 0xFF; \
|
|
*(_x)++ = (samp >> 8) & 0xFF; \
|
|
*(_x)++ = (samp >> 16) & 0xFF; \
|
|
} G_STMT_END
|
|
|
|
#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) \
|
|
G_STMT_START { \
|
|
*(_x)++ = (samp >> 16) & 0xFF; \
|
|
*(_x)++ = (samp >> 8) & 0xFF; \
|
|
*(_x)++ = samp & 0xFF; \
|
|
} G_STMT_END
|
|
#endif
|
|
|
|
static void
|
|
volume_process_int24 (GstVolume * self, 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) self->current_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 * self, 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) self->current_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_controlled_int24_clamp (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gint8 *data = (gint8 *) bytes; /* treat the data as a byte stream */
|
|
guint i, j;
|
|
guint num_samples = n_bytes / (sizeof (gint8) * 3 * channels);
|
|
gdouble vol, val;
|
|
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
val = get_unaligned_i24 (data) * vol;
|
|
val = CLAMP (val, VOLUME_MIN_INT24, VOLUME_MAX_INT24);
|
|
write_unaligned_u24 (data, (gint32) val);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
volume_process_int16 (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint16 *data = (gint16 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint16);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT16_BIT_SHIFT == 13);
|
|
|
|
orc_process_int16 (data, self->current_vol_i16, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_int16_clamp (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint16 *data = (gint16 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint16);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT16_BIT_SHIFT == 13);
|
|
|
|
orc_process_int16_clamp (data, self->current_vol_i16, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_controlled_int16_clamp (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gint16 *data = (gint16 *) bytes;
|
|
guint i, j;
|
|
guint num_samples = n_bytes / (sizeof (gint16) * channels);
|
|
gdouble vol, val;
|
|
|
|
if (channels == 1) {
|
|
orc_process_controlled_int16_1ch (data, volume, num_samples);
|
|
} else if (channels == 2) {
|
|
orc_process_controlled_int16_2ch (data, volume, num_samples);
|
|
} else {
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
val = *data * vol;
|
|
*data++ = (gint16) CLAMP (val, VOLUME_MIN_INT16, VOLUME_MAX_INT16);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
volume_process_int8 (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint8 *data = (gint8 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint8);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT8_BIT_SHIFT == 5);
|
|
|
|
orc_process_int8 (data, self->current_vol_i8, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_int8_clamp (GstVolume * self, gpointer bytes, guint n_bytes)
|
|
{
|
|
gint8 *data = (gint8 *) bytes;
|
|
guint num_samples = n_bytes / sizeof (gint8);
|
|
|
|
/* hard coded in volume.orc */
|
|
g_assert (VOLUME_UNITY_INT8_BIT_SHIFT == 5);
|
|
|
|
orc_process_int8_clamp (data, self->current_vol_i8, num_samples);
|
|
}
|
|
|
|
static void
|
|
volume_process_controlled_int8_clamp (GstVolume * self, gpointer bytes,
|
|
gdouble * volume, guint channels, guint n_bytes)
|
|
{
|
|
gint8 *data = (gint8 *) bytes;
|
|
guint i, j;
|
|
guint num_samples = n_bytes / (sizeof (gint8) * channels);
|
|
gdouble val, vol;
|
|
|
|
if (channels == 1) {
|
|
orc_process_controlled_int8_1ch (data, volume, num_samples);
|
|
} else if (channels == 2) {
|
|
orc_process_controlled_int8_2ch (data, volume, num_samples);
|
|
} else {
|
|
for (i = 0; i < num_samples; i++) {
|
|
vol = *volume++;
|
|
for (j = 0; j < channels; j++) {
|
|
val = *data * vol;
|
|
*data++ = (gint8) CLAMP (val, 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)
|
|
{
|
|
gboolean res;
|
|
GstVolume *self = GST_VOLUME (filter);
|
|
gfloat volume;
|
|
gboolean mute;
|
|
|
|
GST_OBJECT_LOCK (self);
|
|
volume = self->volume;
|
|
mute = self->mute;
|
|
GST_OBJECT_UNLOCK (self);
|
|
|
|
res = volume_update_volume (self, volume, mute);
|
|
if (!res) {
|
|
GST_ELEMENT_ERROR (self, CORE, NEGOTIATION,
|
|
("Invalid incoming format"), (NULL));
|
|
}
|
|
self->negotiated = res;
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
volume_stop (GstBaseTransform * base)
|
|
{
|
|
GstVolume *self = GST_VOLUME (base);
|
|
|
|
g_free (self->volumes);
|
|
self->volumes = NULL;
|
|
self->volumes_count = 0;
|
|
|
|
g_free (self->mutes);
|
|
self->mutes = NULL;
|
|
self->mutes_count = 0;
|
|
|
|
return GST_CALL_PARENT_WITH_DEFAULT (GST_BASE_TRANSFORM_CLASS, stop, (base),
|
|
TRUE);
|
|
}
|
|
|
|
static void
|
|
volume_before_transform (GstBaseTransform * base, GstBuffer * buffer)
|
|
{
|
|
GstClockTime timestamp;
|
|
GstVolume *self = GST_VOLUME (base);
|
|
gfloat volume;
|
|
gboolean mute;
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
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 (self), timestamp);
|
|
|
|
/* get latest values */
|
|
GST_OBJECT_LOCK (self);
|
|
volume = self->volume;
|
|
mute = self->mute;
|
|
GST_OBJECT_UNLOCK (self);
|
|
|
|
if ((volume != self->current_volume) || (mute != self->current_mute)) {
|
|
/* the volume or mute was updated, update our internal state before
|
|
* we continue processing. */
|
|
volume_update_volume (self, volume, mute);
|
|
}
|
|
}
|
|
|
|
/* 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 *self = GST_VOLUME (base);
|
|
guint8 *data;
|
|
guint size;
|
|
GstControlSource *mute_csource, *volume_csource;
|
|
|
|
if (G_UNLIKELY (!self->negotiated))
|
|
goto not_negotiated;
|
|
|
|
/* 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;
|
|
|
|
data = GST_BUFFER_DATA (outbuf);
|
|
size = GST_BUFFER_SIZE (outbuf);
|
|
|
|
mute_csource = gst_object_get_control_source (G_OBJECT (self), "mute");
|
|
volume_csource = gst_object_get_control_source (G_OBJECT (self), "volume");
|
|
if (mute_csource || (volume_csource && !self->current_mute)) {
|
|
gint rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
|
|
gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
|
|
gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
|
|
guint nsamples = size / (width * channels);
|
|
GstClockTime interval = gst_util_uint64_scale_int (1, GST_SECOND, rate);
|
|
GstClockTime ts = GST_BUFFER_TIMESTAMP (outbuf);
|
|
|
|
ts = gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, ts);
|
|
|
|
if (self->mutes_count < nsamples && mute_csource) {
|
|
self->mutes = g_realloc (self->mutes, sizeof (gboolean) * nsamples);
|
|
self->mutes_count = nsamples;
|
|
}
|
|
|
|
if (self->volumes_count < nsamples) {
|
|
self->volumes = g_realloc (self->volumes, sizeof (gdouble) * nsamples);
|
|
self->volumes_count = nsamples;
|
|
}
|
|
|
|
if (mute_csource) {
|
|
GstValueArray va = { "mute", nsamples, interval, (gpointer) self->mutes };
|
|
|
|
if (!gst_control_source_get_value_array (mute_csource, ts, &va))
|
|
goto controller_failure;
|
|
|
|
gst_object_unref (mute_csource);
|
|
mute_csource = NULL;
|
|
} else {
|
|
g_free (self->mutes);
|
|
self->mutes = NULL;
|
|
self->mutes_count = 0;
|
|
}
|
|
|
|
if (volume_csource) {
|
|
GstValueArray va =
|
|
{ "volume", nsamples, interval, (gpointer) self->volumes };
|
|
|
|
if (!gst_control_source_get_value_array (volume_csource, ts, &va))
|
|
goto controller_failure;
|
|
|
|
gst_object_unref (volume_csource);
|
|
volume_csource = NULL;
|
|
} else {
|
|
orc_memset_f64 (self->volumes, self->current_volume, nsamples);
|
|
}
|
|
|
|
if (mute_csource) {
|
|
orc_prepare_volumes (self->volumes, self->mutes, nsamples);
|
|
}
|
|
|
|
self->process_controlled (self, data, self->volumes, channels, size);
|
|
|
|
return GST_FLOW_OK;
|
|
} else if (volume_csource) {
|
|
gst_object_unref (volume_csource);
|
|
}
|
|
|
|
if (self->current_volume == 0.0 || self->current_mute) {
|
|
orc_memset (data, 0, size);
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
|
|
} else if (self->current_volume != 1.0) {
|
|
self->process (self, data, size);
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
not_negotiated:
|
|
{
|
|
GST_ELEMENT_ERROR (self, CORE, NEGOTIATION,
|
|
("No format was negotiated"), (NULL));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
controller_failure:
|
|
{
|
|
if (mute_csource)
|
|
gst_object_unref (mute_csource);
|
|
if (volume_csource)
|
|
gst_object_unref (volume_csource);
|
|
|
|
GST_ELEMENT_ERROR (self, CORE, FAILED,
|
|
("Failed to get values from controller"), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static void
|
|
volume_set_property (GObject * object, guint prop_id, const GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstVolume *self = GST_VOLUME (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_MUTE:
|
|
GST_OBJECT_LOCK (self);
|
|
self->mute = g_value_get_boolean (value);
|
|
GST_OBJECT_UNLOCK (self);
|
|
break;
|
|
case PROP_VOLUME:
|
|
GST_OBJECT_LOCK (self);
|
|
self->volume = g_value_get_double (value);
|
|
GST_OBJECT_UNLOCK (self);
|
|
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 *self = GST_VOLUME (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_MUTE:
|
|
GST_OBJECT_LOCK (self);
|
|
g_value_set_boolean (value, self->mute);
|
|
GST_OBJECT_UNLOCK (self);
|
|
break;
|
|
case PROP_VOLUME:
|
|
GST_OBJECT_LOCK (self);
|
|
g_value_set_double (value, self->volume);
|
|
GST_OBJECT_UNLOCK (self);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
gst_volume_orc_init ();
|
|
|
|
/* initialize gst controller library */
|
|
gst_controller_init (NULL, NULL);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
|
|
|
|
/* ref class from a thread-safe context to work around missing bit of
|
|
* thread-safety in GObject */
|
|
g_type_class_ref (GST_TYPE_MIXER_TRACK);
|
|
|
|
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);
|