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bs2b: add new plugin (Effect/Audio, crossfeed)

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https://bugzilla.gnome.org/show_bug.cgi?id=611689
This commit is contained in:
Christoph Reiter 2015-01-10 21:41:12 +01:00 committed by Stefan Sauer
parent efb74ca0df
commit 75f8cca325
5 changed files with 513 additions and 0 deletions
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12
configure.ac
View file

@ -1820,6 +1820,16 @@ AG_GST_CHECK_FEATURE(APEXSINK, [AirPort Express Wireless sink], apexsink, [
])
])
dnl *** bs2b ***
translit(dnm, m, l) AM_CONDITIONAL(USE_BS2B, true)
AG_GST_CHECK_FEATURE(BS2B, [bs2b], bs2b, [
PKG_CHECK_MODULES(BS2B, libbs2b >= 3.1.0, HAVE_BS2B="yes", [
HAVE_BS2B="no"
])
AC_SUBST(BS2B_CFLAGS)
AC_SUBST(BS2B_LIBS)
])
dnl *** BZ2 ***
translit(dnm, m, l) AM_CONDITIONAL(USE_BZ2, true)
AG_GST_CHECK_FEATURE(BZ2, [bz2 library], bz2, [
@ -2998,6 +3008,7 @@ AM_CONDITIONAL(USE_ASSRENDER, false)
AM_CONDITIONAL(USE_VOAMRWBENC, false)
AM_CONDITIONAL(USE_VOAACENC, false)
AM_CONDITIONAL(USE_APEXSINK, false)
AM_CONDITIONAL(USE_BS2B, false)
AM_CONDITIONAL(USE_BZ2, false)
AM_CONDITIONAL(USE_CHROMAPRINT, false)
AM_CONDITIONAL(USE_CURL, false)
@ -3308,6 +3319,7 @@ ext/voamrwbenc/Makefile
ext/voaacenc/Makefile
ext/assrender/Makefile
ext/apexsink/Makefile
ext/bs2b/Makefile
ext/bz2/Makefile
ext/chromaprint/Makefile
ext/curl/Makefile

8
ext/Makefile.am
View file

@ -22,6 +22,12 @@ endif
AUDIOFILE_DIR=
# endif
if USE_BS2B
BS2B_DIR=bs2b
else
BS2B_DIR=
endif
if USE_BZ2
BZ2_DIR=bz2
else
@ -418,6 +424,7 @@ SUBDIRS=\
$(VOAMRWBENC_DIR) \
$(APEXSINK_DIR) \
$(AUDIOFILE_DIR) \
$(BS2B_DIR) \
$(BZ2_DIR) \
$(CHROMAPRINT_DIR) \
$(CURL_DIR) \
@ -485,6 +492,7 @@ SUBDIRS=\
DIST_SUBDIRS = \
assrender \
apexsink \
bs2b \
bz2 \
chromaprint \
curl \

15
ext/bs2b/Makefile.am Normal file
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@ -0,0 +1,15 @@
plugin_LTLIBRARIES = libgstbs2b.la
libgstbs2b_la_SOURCES = gstbs2b.c gstbs2b.h
libgstbs2b_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) \
$(BS2B_CFLAGS)
libgstbs2b_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
$(GST_BASE_LIBS) $(GST_LIBS) \
$(BS2B_LIBS)
libgstbs2b_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstbs2b_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstbs2b.h

415
ext/bs2b/gstbs2b.c Normal file
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@ -0,0 +1,415 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
* Copyright (C) <2011,2014> Christoph Reiter <reiter.christoph@gmail.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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* SECTION:element-bs2b
*
* Improve headphone listening of stereo audio records using the bs2b library.
* It does so by mixing the left and right channel in a way that simulates
* a stereo speaker setup while using headphones.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 audiotestsrc ! "audio/x-raw,channel-mask=(bitmask)0x1" ! interleave name=i ! bs2b ! autoaudiosink audiotestsrc freq=330 ! "audio/x-raw,channel-mask=(bitmask)0x2" ! i.
* ]| Play two independent sine test sources and crossfeed them.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include "gstbs2b.h"
#define GST_BS2B_DP_LOCK(obj) g_mutex_lock (&obj->bs2b_lock)
#define GST_BS2B_DP_UNLOCK(obj) g_mutex_unlock (&obj->bs2b_lock)
#define SUPPORTED_FORMAT \
"(string) { S8, U8, S16LE, S16BE, U16LE, U16BE, S32LE, S32BE, U32LE, " \
"U32BE, S24LE, S24BE, U24LE, U24BE, F32LE, F32BE, F64LE, F64BE }"
#define SUPPORTED_RATE \
"(int) [ " G_STRINGIFY (BS2B_MINSRATE) ", " G_STRINGIFY (BS2B_MAXSRATE) " ]"
#define FRONT_L_FRONT_R "(bitmask) 0x3"
#define PAD_CAPS \
"audio/x-raw, " \
"format = " SUPPORTED_FORMAT ", " \
"rate = " SUPPORTED_RATE ", " \
"channels = (int) 2, " \
"channel-mask = " FRONT_L_FRONT_R ", " \
"layout = (string) interleaved" \
"; " \
"audio/x-raw, " \
"channels = (int) 1" \
enum
{
PROP_FCUT = 1,
PROP_FEED,
PROP_PRESET,
PROP_LAST,
};
static GParamSpec *properties[PROP_LAST];
enum
{
PRESET_DEFAULT,
PRESET_CMOY,
PRESET_JMEIER,
PRESET_NONE
};
G_DEFINE_TYPE (GstBs2b, gst_bs2b, GST_TYPE_AUDIO_FILTER);
static GType gst_bs2b_preset_get_type (void);
static void gst_bs2b_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_bs2b_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_bs2b_finalize (GObject * object);
static GstFlowReturn gst_bs2b_transform_inplace (GstBaseTransform *
base_transform, GstBuffer * buffer);
static gboolean gst_bs2b_setup (GstAudioFilter * self,
const GstAudioInfo * audio_info);
static void
gst_bs2b_class_init (GstBs2bClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
GstCaps *caps;
gobject_class->set_property = gst_bs2b_set_property;
gobject_class->get_property = gst_bs2b_get_property;
gobject_class->finalize = gst_bs2b_finalize;
trans_class->transform_ip = gst_bs2b_transform_inplace;
trans_class->transform_ip_on_passthrough = FALSE;
filter_class->setup = gst_bs2b_setup;
properties[PROP_FCUT] = g_param_spec_int ("fcut", "Frequency cut",
"Low-pass filter cut frequency (Hz)",
BS2B_MINFCUT, BS2B_MAXFCUT, BS2B_DEFAULT_CLEVEL & 0xFFFF,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS);
properties[PROP_FEED] =
g_param_spec_int ("feed", "Feed level", "Feed Level (dB/10)",
BS2B_MINFEED, BS2B_MAXFEED, BS2B_DEFAULT_CLEVEL >> 16,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS);
properties[PROP_PRESET] =
g_param_spec_enum ("preset", "Preset", "Bs2b filter preset",
gst_bs2b_preset_get_type (), PRESET_DEFAULT,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS);
g_object_class_install_properties (gobject_class, PROP_LAST, properties);
gst_element_class_set_metadata (element_class,
"Crossfeed effect",
"Filter/Effect/Audio",
"Improve headphone listening of stereo audio records using the bs2b "
"library.", "Christoph Reiter <reiter.christoph@gmail.com>");
caps = gst_caps_from_string (PAD_CAPS);
gst_audio_filter_class_add_pad_templates (filter_class, caps);
gst_caps_unref (caps);
}
static void
gst_bs2b_init (GstBs2b * element)
{
g_mutex_init (&element->bs2b_lock);
element->bs2bdp = bs2b_open ();
}
static gboolean
gst_bs2b_setup (GstAudioFilter * filter, const GstAudioInfo * audio_info)
{
GstBaseTransform *base_transform = GST_BASE_TRANSFORM (filter);
GstBs2b *element = GST_BS2B (filter);
gint channels = GST_AUDIO_INFO_CHANNELS (audio_info);
element->func = NULL;
if (channels == 1) {
gst_base_transform_set_passthrough (base_transform, TRUE);
return TRUE;
}
g_assert (channels == 2);
gst_base_transform_set_passthrough (base_transform, FALSE);
switch (GST_AUDIO_INFO_FORMAT (audio_info)) {
case GST_AUDIO_FORMAT_S8:
element->func = &bs2b_cross_feed_s8;
break;
case GST_AUDIO_FORMAT_U8:
element->func = &bs2b_cross_feed_u8;
break;
case GST_AUDIO_FORMAT_S16BE:
element->func = &bs2b_cross_feed_s16be;
break;
case GST_AUDIO_FORMAT_S16LE:
element->func = &bs2b_cross_feed_s16le;
break;
case GST_AUDIO_FORMAT_U16BE:
element->func = &bs2b_cross_feed_u16be;
break;
case GST_AUDIO_FORMAT_U16LE:
element->func = &bs2b_cross_feed_u16le;
break;
case GST_AUDIO_FORMAT_S24BE:
element->func = &bs2b_cross_feed_s24be;
break;
case GST_AUDIO_FORMAT_S24LE:
element->func = &bs2b_cross_feed_s24le;
break;
case GST_AUDIO_FORMAT_U24BE:
element->func = &bs2b_cross_feed_u24be;
break;
case GST_AUDIO_FORMAT_U24LE:
element->func = &bs2b_cross_feed_u24le;
break;
case GST_AUDIO_FORMAT_S32BE:
element->func = &bs2b_cross_feed_s32be;
break;
case GST_AUDIO_FORMAT_S32LE:
element->func = &bs2b_cross_feed_s32le;
break;
case GST_AUDIO_FORMAT_U32BE:
element->func = &bs2b_cross_feed_u32be;
break;
case GST_AUDIO_FORMAT_U32LE:
element->func = &bs2b_cross_feed_u32le;
break;
case GST_AUDIO_FORMAT_F32BE:
element->func = &bs2b_cross_feed_fbe;
break;
case GST_AUDIO_FORMAT_F32LE:
element->func = &bs2b_cross_feed_fle;
break;
case GST_AUDIO_FORMAT_F64BE:
element->func = &bs2b_cross_feed_dbe;
break;
case GST_AUDIO_FORMAT_F64LE:
element->func = &bs2b_cross_feed_dle;
break;
default:
return FALSE;
}
g_assert (element->func);
element->bytes_per_sample =
(GST_AUDIO_INFO_WIDTH (audio_info) * channels) / 8;
GST_BS2B_DP_LOCK (element);
bs2b_set_srate (element->bs2bdp, GST_AUDIO_INFO_RATE (audio_info));
GST_BS2B_DP_UNLOCK (element);
return TRUE;
}
static void
gst_bs2b_finalize (GObject * object)
{
GstBs2b *element = GST_BS2B (object);
bs2b_close (element->bs2bdp);
element->bs2bdp = NULL;
G_OBJECT_CLASS (gst_bs2b_parent_class)->finalize (object);
}
static GstFlowReturn
gst_bs2b_transform_inplace (GstBaseTransform * base_transform,
GstBuffer * buffer)
{
GstBs2b *element = GST_BS2B (base_transform);
GstMapInfo map_info;
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ | GST_MAP_WRITE))
return GST_FLOW_ERROR;
GST_BS2B_DP_LOCK (element);
if (GST_BUFFER_IS_DISCONT (buffer))
bs2b_clear (element->bs2bdp);
element->func (element->bs2bdp, map_info.data,
map_info.size / element->bytes_per_sample);
GST_BS2B_DP_UNLOCK (element);
gst_buffer_unmap (buffer, &map_info);
return GST_FLOW_OK;
}
static GType
gst_bs2b_preset_get_type (void)
{
static GType bs2b_preset_type = 0;
if (!bs2b_preset_type) {
static GEnumValue types[] = {
{
PRESET_DEFAULT,
"Closest to virtual speaker placement (30°, 3 meter) [700Hz, 4.5dB]",
"default"},
{
PRESET_CMOY,
"Close to Chu Moy's crossfeeder (popular) [700Hz, 6.0dB]",
"cmoy"},
{
PRESET_JMEIER,
"Close to Jan Meier's CORDA amplifiers (little change) [650Hz, 9.0dB]",
"jmeier"},
{
PRESET_NONE,
"No preset",
"none"},
{0, NULL, NULL},
};
bs2b_preset_type = g_enum_register_static ("GstBs2bPreset", types);
}
return bs2b_preset_type;
}
static void
gst_bs2b_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBs2b *element = GST_BS2B (object);
switch (prop_id) {
case PROP_FCUT:
GST_BS2B_DP_LOCK (element);
bs2b_set_level_fcut (element->bs2bdp, g_value_get_int (value));
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
g_object_notify_by_pspec (object, properties[PROP_PRESET]);
break;
case PROP_FEED:
GST_BS2B_DP_LOCK (element);
bs2b_set_level_feed (element->bs2bdp, g_value_get_int (value));
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
g_object_notify_by_pspec (object, properties[PROP_PRESET]);
break;
case PROP_PRESET:
switch (g_value_get_enum (value)) {
case PRESET_DEFAULT:
GST_BS2B_DP_LOCK (element);
bs2b_set_level (element->bs2bdp, BS2B_DEFAULT_CLEVEL);
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
break;
case PRESET_CMOY:
GST_BS2B_DP_LOCK (element);
bs2b_set_level (element->bs2bdp, BS2B_CMOY_CLEVEL);
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
break;
case PRESET_JMEIER:
GST_BS2B_DP_LOCK (element);
bs2b_set_level (element->bs2bdp, BS2B_JMEIER_CLEVEL);
bs2b_clear (element->bs2bdp);
GST_BS2B_DP_UNLOCK (element);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
return;
}
g_object_notify_by_pspec (object, properties[PROP_FCUT]);
g_object_notify_by_pspec (object, properties[PROP_FEED]);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_bs2b_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBs2b *element = GST_BS2B (object);
switch (prop_id) {
case PROP_FCUT:
GST_BS2B_DP_LOCK (element);
g_value_set_int (value, bs2b_get_level_fcut (element->bs2bdp));
GST_BS2B_DP_UNLOCK (element);
break;
case PROP_FEED:
GST_BS2B_DP_LOCK (element);
g_value_set_int (value, bs2b_get_level_feed (element->bs2bdp));
GST_BS2B_DP_UNLOCK (element);
break;
case PROP_PRESET:
GST_BS2B_DP_LOCK (element);
switch (bs2b_get_level (element->bs2bdp)) {
case BS2B_DEFAULT_CLEVEL:
g_value_set_enum (value, PRESET_DEFAULT);
break;
case BS2B_CMOY_CLEVEL:
g_value_set_enum (value, PRESET_CMOY);
break;
case BS2B_JMEIER_CLEVEL:
g_value_set_enum (value, PRESET_JMEIER);
break;
default:
g_value_set_enum (value, PRESET_NONE);
break;
}
GST_BS2B_DP_UNLOCK (element);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "bs2b", GST_RANK_NONE, GST_TYPE_BS2B);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
bs2b,
"Improve headphone listening of stereo audio records"
"using the bs2b library.",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

63
ext/bs2b/gstbs2b.h Normal file
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@ -0,0 +1,63 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
* Copyright (C) <2011,2014> Christoph Reiter <reiter.christoph@gmail.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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef __GST_BS2B_H__
#define __GST_BS2B_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <bs2b/bs2b.h>
G_BEGIN_DECLS
#define GST_TYPE_BS2B \
(gst_bs2b_get_type())
#define GST_BS2B(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BS2B,GstBs2b))
#define GST_BS2B_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BS2B,GstBs2bClass))
#define GST_IS_BS2B(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BS2B))
#define GST_IS_BS2B_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BS2B))
typedef struct _GstBs2b GstBs2b;
typedef struct _GstBs2bClass GstBs2bClass;
struct _GstBs2b
{
GstAudioFilter element;
/*< private > */
GMutex bs2b_lock;
t_bs2bdp bs2bdp;
void (*func) ();
guint bytes_per_sample;
};
struct _GstBs2bClass
{
GstAudioFilterClass parent_class;
};
GType gst_bs2b_get_type (void);
G_END_DECLS
#endif /* __GST_BS2B_H__ */