/* GStreamer * 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "audio.h" #include "multichannel-enumtypes.h" #include /** * SECTION:gstaudio * @short_description: Support library for audio elements */ 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 -1 if there's an error (to avoid division by zero), * or the byte size if everything's ok */ int width = 0; int channels = 0; const GstCaps *caps = NULL; GstStructure *structure; /* get caps of pad */ caps = GST_PAD_CAPS (pad); if (caps == NULL) { /* ERROR: could not get caps of pad */ g_warning ("gstaudio: could not get caps of pad %s:%s\n", GST_DEBUG_PAD_NAME (pad)); return 0; } structure = gst_caps_get_structure (caps, 0); gst_structure_get_int (structure, "width", &width); gst_structure_get_int (structure, "channels", &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; } GstClockTime gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf) { /* calculate length in nanoseconds * of audio buffer buf * based on capabilities of pad */ long bytes = 0; int width = 0; int channels = 0; int rate = 0; GstClockTime length; const GstCaps *caps = NULL; GstStructure *structure; g_assert (GST_IS_BUFFER (buf)); /* get caps of pad */ caps = GST_PAD_CAPS (pad); if (caps == NULL) { /* ERROR: could not get caps of pad */ g_warning ("gstaudio: could not get caps of pad %s:%s\n", GST_DEBUG_PAD_NAME (pad)); length = GST_CLOCK_TIME_NONE; } else { structure = gst_caps_get_structure (caps, 0); bytes = GST_BUFFER_SIZE (buf); gst_structure_get_int (structure, "width", &width); gst_structure_get_int (structure, "channels", &channels); gst_structure_get_int (structure, "rate", &rate); g_assert (bytes != 0); g_assert (width != 0); g_assert (channels != 0); g_assert (rate != 0); length = (bytes * 8 * GST_SECOND) / (rate * channels * width); } return length; } 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; } /* _getcaps helper functions * sets structure fields to default for audio type * flag determines which structure fields to set to default * keep these functions in sync with the templates in audio.h */ /* private helper function * sets a list on the structure * pass in structure, fieldname for the list, type of the list values, * number of list values, and each of the values, terminating with NULL */ static void _gst_audio_structure_set_list (GstStructure * structure, const gchar * fieldname, GType type, int number, ...) { va_list varargs; GValue value = { 0 }; GArray *array; int j; g_return_if_fail (structure != NULL); g_value_init (&value, GST_TYPE_LIST); array = g_value_peek_pointer (&value); va_start (varargs, number); for (j = 0; j < number; ++j) { int i; gboolean b; GValue list_value = { 0 }; switch (type) { case G_TYPE_INT: i = va_arg (varargs, int); g_value_init (&list_value, G_TYPE_INT); g_value_set_int (&list_value, i); break; case G_TYPE_BOOLEAN: b = va_arg (varargs, gboolean); g_value_init (&list_value, G_TYPE_BOOLEAN); g_value_set_boolean (&list_value, b); break; default: g_warning ("_gst_audio_structure_set_list: LIST of given type not implemented."); } g_array_append_val (array, list_value); } gst_structure_set_value (structure, fieldname, &value); va_end (varargs); } void gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag) { if (flag & GST_AUDIO_FIELD_RATE) gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); if (flag & GST_AUDIO_FIELD_CHANNELS) gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); if (flag & GST_AUDIO_FIELD_ENDIANNESS) _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2, G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL); if (flag & GST_AUDIO_FIELD_WIDTH) _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32, NULL); if (flag & GST_AUDIO_FIELD_DEPTH) gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL); if (flag & GST_AUDIO_FIELD_SIGNED) _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE, FALSE, NULL); }