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>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-volume
*
* <refsect2>
* <title>Example launch line</title>
* <para>
* The volume element changes the volume of the audio data.
* </para>
* <para>
* <programlisting>
* gst-launch -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
* </programlisting>
* This pipeline shows that the level of audiotestsrc has been halved
* (peak values are around -6 dB and RMS around -9 dB) compared to
* the same pipeline without the volume element.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <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 <liboil/liboil.h>
#include "gstvolume.h"
/* some defines for audio processing */
/* the volume factor is a range from 0.0 to (arbitrary) 4.0
* we map 1.0 to VOLUME_UNITY_INT
*/
#define VOLUME_UNITY_INT 8192 /* internal int for unity */
#define VOLUME_UNITY_BIT_SHIFT 13 /* number of bits to shift for unity */
#define VOLUME_MAX_DOUBLE 10.0
#define VOLUME_MAX_INT16 32767
#define VOLUME_MIN_INT16 -32768
/* 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);
make GstElementDetails const Original commit message from CVS: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsasrc.c: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/ogg/gstogmparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst/audioconvert/gstaudioconvert.c: * gst/audiorate/gstaudiorate.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: * gst/typefind/gsttypefindfunctions.c: (plugin_init): * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/v4l/gstv4ljpegsrc.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * sys/v4l/gstv4lsrc.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: * tests/check/libs/cddabasesrc.c: make GstElementDetails const
2006-04-28 19:46:37 +00:00
static const GstElementDetails volume_details = GST_ELEMENT_DETAILS ("Volume",
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
"Filter/Effect/Audio",
"Set volume on audio/raw streams",
"Andy Wingo <wingo@pobox.com>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_SILENT,
PROP_MUTE,
PROP_VOLUME
};
static GstStaticPadTemplate volume_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32; "
"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")
);
static GstStaticPadTemplate volume_src_template = GST_STATIC_PAD_TEMPLATE
("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32; "
"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")
);
static void gst_volume_interface_init (GstImplementsInterfaceClass * klass);
static void gst_volume_mixer_init (GstMixerClass * iface);
#define _init_interfaces(type) \
{ \
static const GInterfaceInfo voliface_info = { \
(GInterfaceInitFunc) gst_volume_interface_init, \
NULL, \
NULL \
}; \
static const GInterfaceInfo volmixer_info = { \
(GInterfaceInitFunc) gst_volume_mixer_init, \
NULL, \
NULL \
}; \
\
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, \
&voliface_info); \
g_type_add_interface_static (type, GST_TYPE_MIXER, &volmixer_info); \
}
GST_BOILERPLATE_FULL (GstVolume, gst_volume, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, _init_interfaces);
static void volume_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void volume_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void volume_update_volume (const GValue * value, gpointer data);
static void volume_update_mute (const GValue * value, gpointer data);
static GstFlowReturn volume_transform_ip (GstBaseTransform * base,
GstBuffer * outbuf);
static gboolean volume_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps);
static void volume_process_float (GstVolume * this, gpointer bytes,
gint n_bytes);
static void volume_process_int16 (GstVolume * this, gpointer bytes,
gint n_bytes);
static void volume_process_int16_clamp (GstVolume * this, gpointer bytes,
gint n_bytes);
/* helper functions */
static void
volume_choose_func (GstVolume * this)
{
switch (this->format) {
case GST_VOLUME_FORMAT_INT:
/* only clamp if the gain is greater than 1.0
* FIXME: real_vol_i can change while processing the buffer!
*/
if (this->real_vol_i > VOLUME_UNITY_INT)
this->process = volume_process_int16_clamp;
else
this->process = volume_process_int16;
break;
case GST_VOLUME_FORMAT_FLOAT:
this->process = volume_process_float;
break;
default:
this->process = NULL;
}
}
static void
volume_update_real_volume (GstVolume * this)
{
gboolean passthrough = FALSE;
if (this->mute) {
this->real_vol_f = 0.0;
this->real_vol_i = 0;
} else {
this->real_vol_f = this->volume_f;
this->real_vol_i = this->volume_i;
passthrough = (this->volume_i == VOLUME_UNITY_INT);
}
volume_choose_func (this);
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (this), passthrough);
}
/* Mixer interface */
static gboolean
gst_volume_interface_supported (GstImplementsInterface * iface, GType type)
{
g_assert (type == GST_TYPE_MIXER);
return TRUE;
}
static void
gst_volume_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_volume_interface_supported;
}
static const GList *
gst_volume_list_tracks (GstMixer * mixer)
{
GstVolume *this = GST_VOLUME (mixer);
g_return_val_if_fail (this != NULL, NULL);
g_return_val_if_fail (GST_IS_VOLUME (this), NULL);
return this->tracklist;
}
static void
gst_volume_set_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
{
GstVolume *this = GST_VOLUME (mixer);
g_return_if_fail (this != NULL);
g_return_if_fail (GST_IS_VOLUME (this));
this->volume_f = (gfloat) volumes[0] / VOLUME_STEPS;
this->volume_i = this->volume_f * VOLUME_UNITY_INT;
volume_update_real_volume (this);
}
static void
gst_volume_get_volume (GstMixer * mixer, GstMixerTrack * track, gint * volumes)
{
GstVolume *this = GST_VOLUME (mixer);
g_return_if_fail (this != NULL);
g_return_if_fail (GST_IS_VOLUME (this));
volumes[0] = (gint) this->volume_f * VOLUME_STEPS;
}
static void
gst_volume_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
{
GstVolume *this = GST_VOLUME (mixer);
g_return_if_fail (this != NULL);
g_return_if_fail (GST_IS_VOLUME (this));
this->mute = mute;
volume_update_real_volume (this);
}
static void
gst_volume_mixer_init (GstMixerClass * klass)
{
GST_MIXER_TYPE (klass) = GST_MIXER_SOFTWARE;
/* default virtual functions */
klass->list_tracks = gst_volume_list_tracks;
klass->set_volume = gst_volume_set_volume;
klass->get_volume = gst_volume_get_volume;
klass->set_mute = gst_volume_set_mute;
}
/* Element class */
static void
gst_volume_dispose (GObject * object)
{
GstVolume *volume = GST_VOLUME (object);
if (volume->tracklist) {
if (volume->tracklist->data)
g_object_unref (volume->tracklist->data);
g_list_free (volume->tracklist);
volume->tracklist = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_volume_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&volume_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&volume_sink_template));
gst_element_class_set_details (element_class, &volume_details);
}
static void
gst_volume_class_init (GstVolumeClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
gobject_class->set_property = volume_set_property;
gobject_class->get_property = volume_get_property;
gobject_class->dispose = gst_volume_dispose;
g_object_class_install_property (gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute", "mute channel",
FALSE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume", "volume factor",
0.0, VOLUME_MAX_DOUBLE, 1.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "volume", 0, "Volume gain");
trans_class->transform_ip = GST_DEBUG_FUNCPTR (volume_transform_ip);
trans_class->set_caps = GST_DEBUG_FUNCPTR (volume_set_caps);
}
static void
gst_volume_init (GstVolume * this, GstVolumeClass * g_class)
{
GstMixerTrack *track = NULL;
this->mute = FALSE;
this->volume_i = VOLUME_UNITY_INT;
this->volume_f = 1.0;
this->real_vol_i = VOLUME_UNITY_INT;
this->real_vol_f = 1.0;
this->tracklist = NULL;
this->format = GST_VOLUME_FORMAT_NONE;
track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
if (GST_IS_MIXER_TRACK (track)) {
track->label = g_strdup ("volume");
track->num_channels = 1;
track->min_volume = 0;
track->max_volume = VOLUME_STEPS;
track->flags = GST_MIXER_TRACK_SOFTWARE;
this->tracklist = g_list_append (this->tracklist, track);
}
}
/* NOTE: although it might be tempting to have volume_process_mute() which uses
* memset(bytes, 0, nbytes) for the vol=0 case, this has the downside that
* unmuting would unly take place after processing a buffer.
*/
static void
volume_process_float (GstVolume * this, gpointer bytes, gint n_bytes)
{
gfloat *data;
gint i, num_samples;
data = (gfloat *) bytes;
num_samples = n_bytes / sizeof (gfloat);
for (i = 0; i < num_samples; i++) {
*data++ *= this->real_vol_f;
}
/* FIXME: seems to be slower than above!
oil_scalarmultiply_f32_ns (data, data, &this->real_vol_f, num_samples);
*/
}
static void
volume_process_int16 (GstVolume * this, gpointer bytes, gint n_bytes)
{
gint16 *data;
gint i, val, num_samples;
data = (gint16 *) bytes;
num_samples = n_bytes / sizeof (gint16);
/* FIXME: need... liboil...
* 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) ((this->real_vol_i * val) >> VOLUME_UNITY_BIT_SHIFT);
}
}
static void
volume_process_int16_clamp (GstVolume * this, gpointer bytes, gint n_bytes)
{
gint16 *data;
gint i, val, num_samples;
data = (gint16 *) bytes;
num_samples = n_bytes / sizeof (gint16);
/* FIXME: need... liboil...
* 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_i * val) >> VOLUME_UNITY_BIT_SHIFT,
VOLUME_MIN_INT16, VOLUME_MAX_INT16);
}
}
/* GstBaseTransform vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
volume_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps)
{
GstVolume *this = GST_VOLUME (base);
const gchar *mimetype;
GST_DEBUG_OBJECT (this,
"set_caps: in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT, incaps, outcaps);
mimetype = gst_structure_get_name (gst_caps_get_structure (incaps, 0));
/* based on mimetype, choose the correct volume_process format */
if (strcmp (mimetype, "audio/x-raw-int") == 0) {
this->format = GST_VOLUME_FORMAT_INT;
GST_INFO_OBJECT (this, "use int16");
} else if (strcmp (mimetype, "audio/x-raw-float") == 0) {
this->format = GST_VOLUME_FORMAT_FLOAT;
GST_INFO_OBJECT (this, "use float");
} else {
this->process = NULL;
goto invalid_format;
}
volume_choose_func (this);
return TRUE;
/* ERRORS */
invalid_format:
{
GST_ELEMENT_ERROR (this, CORE, NEGOTIATION,
("Invalid incoming caps: %" GST_PTR_FORMAT, incaps), (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;
/* don't process data in passthrough-mode */
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (this), timestamp);
this->process (this, GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
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_i = this->volume_f * VOLUME_UNITY_INT;
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);