diff --git a/gst-libs/audio/Makefile.am b/gst-libs/audio/Makefile.am new file mode 100644 index 0000000000..20a537649d --- /dev/null +++ b/gst-libs/audio/Makefile.am @@ -0,0 +1,14 @@ +filterdir = $(libdir)/gst + +filter_LTLIBRARIES = libgstaudio.la + +libgstaudio_la_SOURCES = gstaudio.c + +libgstaudioincludedir = $(includedir)/gst/libs/gstaudio +libgstaudioinclude_HEADERS = gstaudio.h + +libgstaudio_la_LIBADD = $(GST_LIBS) +libgstaudio_la_CFLAGS = $(GST_CFLAGS) -finline-functions -ffast-math +# FIXME is this needed? +## from merge, kept for reference +## libgstaudio_la_CFLAGS = -O2 $(FOMIT_FRAME_POINTER) -finline-functions -ffast-math $(GLIB_CFLAGS) $(XML_CFLAGS) $(GST_CFLAGS) diff --git a/gst-libs/audio/gstaudio.c b/gst-libs/audio/gstaudio.c new file mode 100644 index 0000000000..c18d6c6e78 --- /dev/null +++ b/gst-libs/audio/gstaudio.c @@ -0,0 +1,152 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * + * 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. + */ + +#include "gstaudio.h" + +int +gst_audio_frame_byte_size (GstPad* pad) +{ +/* calculate byte size of an audio frame + * this should be moved closer to the gstreamer core + * and be implemented for every mime type IMO + * returns 0 if there's an error, or the byte size if everything's ok + */ + + int width = 0; + int channels = 0; + + GstCaps *caps = NULL; + + /* get caps of pad */ + caps = GST_PAD_CAPS (pad); + + if (caps == NULL) + /* ERROR: could not get caps of pad */ + return 0; + + width = gst_caps_get_int (caps, "width"); + channels = gst_caps_get_int (caps, "channels"); + return (width / 8) * channels; +} + +long +gst_audio_frame_length (GstPad* pad, GstBuffer* buf) +/* calculate length of buffer in frames + * this should be moved closer to the gstreamer core + * and be implemented for every mime type IMO + * returns 0 if there's an error, or the number of frames if everything's ok + */ +{ + int frame_byte_size = 0; + + frame_byte_size = gst_audio_frame_byte_size (pad); + if (frame_byte_size == 0) + /* error */ + return 0; + /* FIXME: this function assumes the buffer size to be a whole multiple + * of the frame byte size + */ + return GST_BUFFER_SIZE (buf) / frame_byte_size; +} + +long +gst_audio_frame_rate (GstPad *pad) +/* + * calculate frame rate (based on caps of pad) + * returns 0 if failed, rate if success + */ +{ + GstCaps *caps = NULL; + + /* get caps of pad */ + caps = GST_PAD_CAPS (pad); + + if (caps == NULL) + /* ERROR: could not get caps of pad */ + return 0; + else + return gst_caps_get_int (caps, "rate"); +} + +double +gst_audio_length (GstPad* pad, GstBuffer* buf) +{ +/* calculate length in seconds + * of audio buffer buf + * based on capabilities of pad + */ + + long bytes = 0; + int width = 0; + int channels = 0; + long rate = 0L; + + double length; + + GstCaps *caps = NULL; + + /* get caps of pad */ + caps = GST_PAD_CAPS (pad); + if (caps == NULL) + { + /* ERROR: could not get caps of pad */ + length = 0.0; + } + else + { + bytes = GST_BUFFER_SIZE (buf); + width = gst_caps_get_int (caps, "width"); + channels = gst_caps_get_int (caps, "channels"); + rate = gst_caps_get_int (caps, "rate"); + + length = (bytes * 8.0) / (double) (rate * channels * width); + } + return length; +} + +long +gst_audio_highest_sample_value (GstPad* pad) +/* calculate highest possible sample value + * based on capabilities of pad + */ +{ + gboolean is_signed = FALSE; + gint width = 0; + GstCaps *caps = NULL; + + caps = GST_PAD_CAPS (pad); + // FIXME : Please change this to a better warning method ! + if (caps == NULL) + printf ("WARNING: gstaudio: could not get caps of pad !\n"); + width = gst_caps_get_int (caps, "width"); + is_signed = gst_caps_get_boolean (caps, "signed"); + if (is_signed) --width; + /* example : 16 bit, signed : samples between -32768 and 32767 */ + return ((long) (1 << width)); +} + +gboolean +gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf) +/* check if the buffer size is a whole multiple of the frame size */ +{ + if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0) + return TRUE; + else + return FALSE; +} diff --git a/gst-libs/audio/gstaudio.h b/gst-libs/audio/gstaudio.h new file mode 100644 index 0000000000..09ef3ec796 --- /dev/null +++ b/gst-libs/audio/gstaudio.h @@ -0,0 +1,109 @@ +/* Gnome-Streamer + * Copyright (C) <1999> Erik Walthinsen + * Library <2001> Thomas Vander Stichele + * + * 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. + */ + +#include + +/* for people that are looking at this source: the purpose of these defines is + * to make GstCaps a bit easier, in that you don't have to know all of the + * properties that need to be defined. you can just use these macros. currently + * (8/01) the only plugins that use these are the passthrough, speed, volume, + * and [de]interleave plugins. so. these are for convenience only, and do not + * specify the 'limits' of gstreamer. you might also use these definitions as a + * base for making your own caps, if need be. + * + * for example, to make a source pad that can output mono streams of either + * float or int: + + template = gst_padtemplate_new + ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, + gst_caps_append(gst_caps_new ("sink_int", "audio/raw", + GST_AUDIO_INT_PAD_TEMPLATE_PROPS), + gst_caps_new ("sink_float", "audio/raw", + GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)), + NULL); + + srcpad = gst_pad_new_from_template(template,"src"); + + * Andy Wingo, 18 August 2001 */ + +#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \ + gst_props_new (\ + "format", GST_PROPS_STRING ("int"),\ + "law", GST_PROPS_INT (0),\ + "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ + "signed", GST_PROPS_LIST (\ + GST_PROPS_BOOLEAN (TRUE),\ + GST_PROPS_BOOLEAN(FALSE)\ + ),\ + "width", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\ + "depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\ + "rate", GST_PROPS_INT_RANGE (4000, 96000),\ + "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\ + NULL) + +#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \ + gst_props_new (\ + "format", GST_PROPS_STRING ("int"),\ + "law", GST_PROPS_INT (0),\ + "endianness", GST_PROPS_INT (G_BYTE_ORDER),\ + "signed", GST_PROPS_LIST (\ + GST_PROPS_BOOLEAN (TRUE),\ + GST_PROPS_BOOLEAN(FALSE)\ + ),\ + "width", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\ + "depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\ + "rate", GST_PROPS_INT_RANGE (4000, 96000),\ + "channels", GST_PROPS_INT (1),\ + NULL) + +#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \ + gst_props_new (\ + "format", GST_PROPS_STRING ("float"),\ + "layout", GST_PROPS_STRING ("gfloat"),\ + "intercept", GST_PROPS_FLOAT (0.0),\ + "slope", GST_PROPS_FLOAT (1.0),\ + "rate", GST_PROPS_INT_RANGE (4000, 96000),\ + "channels", GST_PROPS_INT (1),\ + NULL) + +/* + * this library defines and implements some helper functions for audio + * handling + */ + +/* get byte size of audio frame (based on caps of pad */ +int gst_audio_frame_byte_size (GstPad* pad); + +/* get length in frames of buffer */ +long gst_audio_frame_length (GstPad* pad, GstBuffer* buf); + +/* get frame rate based on caps */ +long gst_audio_frame_rate (GstPad *pad); + +/* calculate length in seconds of audio buffer buf based on caps of pad */ +double gst_audio_length (GstPad* pad, GstBuffer* buf); + +/* calculate highest possible sample value based on capabilities of pad */ +long gst_audio_highest_sample_value (GstPad* pad); + +/* check if the buffer size is a whole multiple of the frame size */ +gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf); + +