/* GStreamer * Copyright (C) <1999> Erik Walthinsen * Copyright (C) <2003> David Schleef * Copyright (C) <2011,2014> Christoph Reiter * * 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 * @title: 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. * * ## Example pipelines * |[ * 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. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #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_LAST, }; static GParamSpec *properties[PROP_LAST]; typedef struct { const gchar *name; const gchar *desc; gint preset; } GstBs2bPreset; static const GstBs2bPreset presets[3] = { { "default", "Closest to virtual speaker placement (30°, 3 meter) [700Hz, 4.5dB]", BS2B_DEFAULT_CLEVEL}, { "cmoy", "Close to Chu Moy's crossfeeder (popular) [700Hz, 6.0dB]", BS2B_CMOY_CLEVEL}, { "jmeier", "Close to Jan Meier's CORDA amplifiers (little change) [650Hz, 9.0dB]", BS2B_JMEIER_CLEVEL} }; static void gst_preset_interface_init (gpointer g_iface, gpointer iface_data); G_DEFINE_TYPE_WITH_CODE (GstBs2b, gst_bs2b, GST_TYPE_AUDIO_FILTER, G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, gst_preset_interface_init)); GST_ELEMENT_REGISTER_DEFINE (bs2b, "bs2b", GST_RANK_NONE, GST_TYPE_BS2B); 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 gchar ** gst_bs2b_get_preset_names (GstPreset * preset) { gchar **names; gint i; names = g_new (gchar *, 1 + G_N_ELEMENTS (presets)); for (i = 0; i < G_N_ELEMENTS (presets); i++) { names[i] = g_strdup (presets[i].name); } names[i] = NULL; return names; } static gchar ** gst_bs2b_get_property_names (GstPreset * preset) { gchar **names = g_new (gchar *, 3); names[0] = g_strdup ("fcut"); names[1] = g_strdup ("feed"); names[2] = NULL; return names; } static gboolean gst_bs2b_load_preset (GstPreset * preset, const gchar * name) { GstBs2b *element = GST_BS2B (preset); GObject *object = (GObject *) preset; gint i; for (i = 0; i < G_N_ELEMENTS (presets); i++) { if (!g_strcmp0 (presets[i].name, name)) { bs2b_set_level (element->bs2bdp, presets[i].preset); g_object_notify_by_pspec (object, properties[PROP_FCUT]); g_object_notify_by_pspec (object, properties[PROP_FEED]); return TRUE; } } return FALSE; } static gboolean gst_bs2b_get_meta (GstPreset * preset, const gchar * name, const gchar * tag, gchar ** value) { if (!g_strcmp0 (tag, "comment")) { gint i; for (i = 0; i < G_N_ELEMENTS (presets); i++) { if (!g_strcmp0 (presets[i].name, name)) { *value = g_strdup (presets[i].desc); return TRUE; } } } *value = NULL; return FALSE; } static void gst_preset_interface_init (gpointer g_iface, gpointer iface_data) { GstPresetInterface *iface = g_iface; iface->get_preset_names = gst_bs2b_get_preset_names; iface->get_property_names = gst_bs2b_get_property_names; iface->load_preset = gst_bs2b_load_preset; iface->save_preset = NULL; iface->rename_preset = NULL; iface->delete_preset = NULL; iface->get_meta = gst_bs2b_get_meta; iface->set_meta = NULL; } 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); 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 "); 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 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); 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); 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; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean plugin_init (GstPlugin * plugin) { return GST_ELEMENT_REGISTER (bs2b, plugin); } 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)