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gstreamer/gst/volume/gstvolume.c

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/* -*- c-basic-offset: 2 -*-
* vi:si:et:sw=2:sts=8:ts=8:expandtab
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2005 Andy Wingo <wingo@pobox.com>
* Copyright (C) 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-volume
*
* The volume element changes the volume of the audio data.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
* ]| This pipeline shows that the level of audiotestsrc has been halved
* (peak values are around -6 dB and RMS around -9 dB) compared to
* the same pipeline without the volume element.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/interfaces/mixer.h>
#include <gst/controller/gstcontroller.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#ifdef HAVE_ORC
#include <orc/orcfunctions.h>
#else
#define orc_memset memset
#endif
#include "gstvolumeorc.h"
#include "gstvolume.h"
/* some defines for audio processing */
/* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0
* we map 1.0 to VOLUME_UNITY_INT*
*/
#define VOLUME_UNITY_INT8 32 /* internal int for unity 2^(8-3) */
#define VOLUME_UNITY_INT8_BIT_SHIFT 5 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT16 8192 /* internal int for unity 2^(16-3) */
#define VOLUME_UNITY_INT16_BIT_SHIFT 13 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT24 2097152 /* internal int for unity 2^(24-3) */
#define VOLUME_UNITY_INT24_BIT_SHIFT 21 /* number of bits to shift for unity */
#define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */
#define VOLUME_UNITY_INT32_BIT_SHIFT 27
#define VOLUME_MAX_DOUBLE 10.0
#define VOLUME_MAX_INT8 G_MAXINT8
#define VOLUME_MIN_INT8 G_MININT8
#define VOLUME_MAX_INT16 G_MAXINT16
#define VOLUME_MIN_INT16 G_MININT16
#define VOLUME_MAX_INT24 8388607
#define VOLUME_MIN_INT24 -8388608
#define VOLUME_MAX_INT32 G_MAXINT32
#define VOLUME_MIN_INT32 G_MININT32
/* number of steps we use for the mixer interface to go from 0.0 to 1.0 */
# define VOLUME_STEPS 100
#define GST_CAT_DEFAULT gst_volume_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_PROP_MUTE FALSE
#define DEFAULT_PROP_VOLUME 1.0
enum
{
PROP_0,
PROP_MUTE,
PROP_VOLUME
};
#define ALLOWED_CAPS \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) {32, 64}; " \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 8, " \
"depth = (int) 8, " \
"signed = (bool) TRUE; " \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (bool) TRUE; " \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 24, " \
"depth = (int) 24, " \
"signed = (bool) TRUE; " \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"depth = (int) 32, " \
"signed = (bool) TRUE"
static void gst_volume_interface_init (GstImplementsInterfaceClass * klass);
static void gst_volume_mixer_init (GstMixerClass * iface);
#define _init_interfaces(type) \
{ \
static const GInterfaceInfo voliface_info = { \
(GInterfaceInitFunc) gst_volume_interface_init, \
NULL, \
NULL \
}; \
static const GInterfaceInfo volmixer_info = { \
(GInterfaceInitFunc) gst_volume_mixer_init, \
NULL, \
NULL \
}; \
static const GInterfaceInfo svol_info = { \
NULL, \
NULL, \
NULL \
}; \
\
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, \
&voliface_info); \
g_type_add_interface_static (type, GST_TYPE_MIXER, &volmixer_info); \
g_type_add_interface_static (type, GST_TYPE_STREAM_VOLUME, &svol_info); \
}
GST_BOILERPLATE_FULL (GstVolume, gst_volume, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, _init_interfaces);
static void volume_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void volume_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void volume_before_transform (GstBaseTransform * base,
GstBuffer * buffer);
static GstFlowReturn volume_transform_ip (GstBaseTransform * base,
GstBuffer * outbuf);
static gboolean volume_stop (GstBaseTransform * base);
static gboolean volume_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static void volume_process_double (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_double (GstVolume * self, gpointer bytes,
gdouble * volume, guint channels, guint n_bytes);
static void volume_process_float (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_float (GstVolume * self, gpointer bytes,
gdouble * volume, guint channels, guint n_bytes);
static void volume_process_int32 (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_int32_clamp (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_int32_clamp (GstVolume * self,
gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
static void volume_process_int24 (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_int24_clamp (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_int24_clamp (GstVolume * self,
gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
static void volume_process_int16 (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_int16_clamp (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_int16_clamp (GstVolume * self,
gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
static void volume_process_int8 (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_int8_clamp (GstVolume * self, gpointer bytes,
guint n_bytes);
static void volume_process_controlled_int8_clamp (GstVolume * self,
gpointer bytes, gdouble * volume, guint channels, guint n_bytes);
/* helper functions */
static gboolean
volume_choose_func (GstVolume * self)
{
self->process = NULL;
self->process_controlled = NULL;
if (GST_AUDIO_FILTER (self)->format.caps == NULL)
return FALSE;
switch (GST_AUDIO_FILTER (self)->format.type) {
case GST_BUFTYPE_LINEAR:
switch (GST_AUDIO_FILTER (self)->format.width) {
case 32:
/* only clamp if the gain is greater than 1.0
*/
if (self->current_vol_i32 > VOLUME_UNITY_INT32) {
self->process = volume_process_int32_clamp;
} else {
self->process = volume_process_int32;
}
self->process_controlled = volume_process_controlled_int32_clamp;
break;
case 24:
/* only clamp if the gain is greater than 1.0
*/
if (self->current_vol_i24 > VOLUME_UNITY_INT24) {
self->process = volume_process_int24_clamp;
} else {
self->process = volume_process_int24;
}
self->process_controlled = volume_process_controlled_int24_clamp;
break;
case 16:
/* only clamp if the gain is greater than 1.0
*/
if (self->current_vol_i16 > VOLUME_UNITY_INT16) {
self->process = volume_process_int16_clamp;
} else {
self->process = volume_process_int16;
}
self->process_controlled = volume_process_controlled_int16_clamp;
break;
case 8:
/* only clamp if the gain is greater than 1.0
*/
if (self->current_vol_i16 > VOLUME_UNITY_INT8) {
self->process = volume_process_int8_clamp;
} else {
self->process = volume_process_int8;
}
self->process_controlled = volume_process_controlled_int8_clamp;
break;
}
break;
case GST_BUFTYPE_FLOAT:
switch (GST_AUDIO_FILTER (self)->format.width) {
case 32:
self->process = volume_process_float;
self->process_controlled = volume_process_controlled_float;
break;
case 64:
self->process = volume_process_double;
self->process_controlled = volume_process_controlled_double;
break;
}
break;
default:
break;
}
return (self->process != NULL);
}
static gboolean
volume_update_volume (GstVolume * self, gfloat volume, gboolean mute)
{
gboolean passthrough;
gboolean res;
GstController *controller;
GST_DEBUG_OBJECT (self, "configure mute %d, volume %f", mute, volume);
if (mute) {
self->current_mute = TRUE;
self->current_volume = 0.0;
self->current_vol_i8 = 0;
self->current_vol_i16 = 0;
self->current_vol_i24 = 0;
self->current_vol_i32 = 0;
passthrough = FALSE;
} else {
self->current_mute = FALSE;
self->current_volume = volume;
self->current_vol_i8 = volume * VOLUME_UNITY_INT8;
self->current_vol_i16 = volume * VOLUME_UNITY_INT16;
self->current_vol_i24 = volume * VOLUME_UNITY_INT24;
self->current_vol_i32 = volume * VOLUME_UNITY_INT32;
passthrough = (self->current_vol_i16 == VOLUME_UNITY_INT16);
}
/* If a controller is used, never use passthrough mode
* because the property can change from 1.0 to something
* else in the middle of a buffer.
*/
controller = gst_object_get_controller (G_OBJECT (self));
passthrough = passthrough && (controller == NULL);
GST_DEBUG_OBJECT (self, "set passthrough %d", passthrough);
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (self), passthrough);
res = self->negotiated = volume_choose_func (self);
return res;
}
/* Mixer interface */
static gboolean
gst_volume_interface_supported (GstImplementsInterface * iface, GType type)
{
return (type == GST_TYPE_MIXER || type == GST_TYPE_STREAM_VOLUME);
}
static void
gst_volume_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_volume_interface_supported;
}
static const GList *
gst_volume_list_tracks (GstMixer * mixer)
{
GstVolume *self = GST_VOLUME (mixer);
g_return_val_if_fail (self != NULL, NULL);
g_return_val_if_fail (GST_IS_VOLUME (self), NULL);
return self->tracklist;
}
static void
gst_volume_set_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
{
GstVolume *self = GST_VOLUME (mixer);
g_return_if_fail (self != NULL);
g_return_if_fail (GST_IS_VOLUME (self));
GST_OBJECT_LOCK (self);
self->volume = (gfloat) volumes[0] / VOLUME_STEPS;
GST_OBJECT_UNLOCK (self);
}
static void
gst_volume_get_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
{
GstVolume *self = GST_VOLUME (mixer);
g_return_if_fail (self != NULL);
g_return_if_fail (GST_IS_VOLUME (self));
GST_OBJECT_LOCK (self);
volumes[0] = (gint) self->volume * VOLUME_STEPS;
GST_OBJECT_UNLOCK (self);
}
static void
gst_volume_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
{
GstVolume *self = GST_VOLUME (mixer);
g_return_if_fail (self != NULL);
g_return_if_fail (GST_IS_VOLUME (self));
GST_OBJECT_LOCK (self);
self->mute = mute;
GST_OBJECT_UNLOCK (self);
}
static void
gst_volume_mixer_init (GstMixerClass * klass)
{
GST_MIXER_TYPE (klass) = GST_MIXER_SOFTWARE;
/* default virtual functions */
klass->list_tracks = gst_volume_list_tracks;
klass->set_volume = gst_volume_set_volume;
klass->get_volume = gst_volume_get_volume;
klass->set_mute = gst_volume_set_mute;
}
/* Element class */
static void
gst_volume_dispose (GObject * object)
{
GstVolume *volume = GST_VOLUME (object);
if (volume->tracklist) {
if (volume->tracklist->data)
g_object_unref (volume->tracklist->data);
g_list_free (volume->tracklist);
volume->tracklist = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_volume_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details_simple (element_class, "Volume",
"Filter/Effect/Audio",
"Set volume on audio/raw streams", "Andy Wingo <wingo@pobox.com>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (filter_class, caps);
gst_caps_unref (caps);
}
static void
gst_volume_class_init (GstVolumeClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) (klass);
gobject_class->set_property = volume_set_property;
gobject_class->get_property = volume_get_property;
gobject_class->dispose = gst_volume_dispose;
g_object_class_install_property (gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute", "mute channel",
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 if (channels == 4) {
orc_process_controlled_int8_4ch (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);
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