--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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:gstaudio
+ * @short_description: Support library for audio elements
+ *
+ * This library contains some helper functions for audio elements.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "audio.h"
+#include "audio-enumtypes.h"
+
+#include <gst/gststructure.h>
+
+/**
+ * gst_audio_frame_byte_size:
+ * @pad: the #GstPad to get the caps from
+ *
+ * Calculate byte size of an audio frame.
+ *
+ * Returns: the byte size, or 0 if there was an error
+ */
+int
+gst_audio_frame_byte_size (GstPad * pad)
+{
+ /* FIXME: this should be moved closer to the gstreamer core
+ * and be implemented for every mime type IMO
+ */
+
+ 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;
+}
+
+/**
+ * gst_audio_frame_length:
+ * @pad: the #GstPad to get the caps from
+ * @buf: the #GstBuffer
+ *
+ * Calculate length of buffer in frames.
+ *
+ * Returns: 0 if there's an error, or the number of frames if everything's ok
+ */
+long
+gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
+{
+ /* FIXME: this should be moved closer to the gstreamer core
+ * and be implemented for every mime type IMO
+ */
+ 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;
+}
+
+/**
+ * gst_audio_duration_from_pad_buffer:
+ * @pad: the #GstPad to get the caps from
+ * @buf: the #GstBuffer
+ *
+ * Calculate length in nanoseconds of audio buffer @buf based on capabilities of
+ * @pad.
+ *
+ * Returns: the length.
+ */
+GstClockTime
+gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)
+{
+ 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;
+}
+
+/**
+ * gst_audio_is_buffer_framed:
+ * @pad: the #GstPad to get the caps from
+ * @buf: the #GstBuffer
+ *
+ * Check if the buffer size is a whole multiple of the frame size.
+ *
+ * Returns: %TRUE if buffer size is multiple.
+ */
+gboolean
+gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
+{
+ 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);
+}
+
+/**
+ * gst_audio_structure_set_int:
+ * @structure: a #GstStructure
+ * @flag: a set of #GstAudioFieldFlag
+ *
+ * Do not use anymore.
+ *
+ * Deprecated: use gst_structure_set()
+ */
+#ifndef GST_REMOVE_DEPRECATED
+#ifdef GST_DISABLE_DEPRECATED
+typedef enum
+{
+ GST_AUDIO_FIELD_RATE = (1 << 0),
+ GST_AUDIO_FIELD_CHANNELS = (1 << 1),
+ GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
+ GST_AUDIO_FIELD_WIDTH = (1 << 3),
+ GST_AUDIO_FIELD_DEPTH = (1 << 4),
+ GST_AUDIO_FIELD_SIGNED = (1 << 5),
+} GstAudioFieldFlag;
+void
+gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag);
+#endif /* GST_DISABLE_DEPRECATED */
+
+void
+gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
+{
+ /* was added here:
+ * http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17
+ * but it is not used
+ */
+ 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);
+}
+#endif /* GST_REMOVE_DEPRECATED */
+
+/**
+ * gst_audio_buffer_clip:
+ * @buffer: The buffer to clip.
+ * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
+ * @rate: sample rate.
+ * @frame_size: size of one audio frame in bytes.
+ *
+ * Clip the the buffer to the given %GstSegment.
+ *
+ * After calling this function the caller does not own a reference to
+ * @buffer anymore.
+ *
+ * Returns: %NULL if the buffer is completely outside the configured segment,
+ * otherwise the clipped buffer is returned.
+ *
+ * If the buffer has no timestamp, it is assumed to be inside the segment and
+ * is not clipped
+ *
+ * Since: 0.10.14
+ */
+GstBuffer *
+gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
+ gint frame_size)
+{
+ GstBuffer *ret;
+ GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
+ guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
+ guint8 *data;
+ guint size;
+
+ gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =
+ TRUE;
+
+ g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||
+ segment->format == GST_FORMAT_DEFAULT, buffer);
+ g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);
+
+ if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
+ /* No timestamp - assume the buffer is completely in the segment */
+ return buffer;
+
+ /* Get copies of the buffer metadata to change later.
+ * Calculate the missing values for the calculations,
+ * they won't be changed later though. */
+
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_SIZE (buffer);
+
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
+ duration = GST_BUFFER_DURATION (buffer);
+ } else {
+ change_duration = FALSE;
+ duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);
+ }
+
+ if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
+ offset = GST_BUFFER_OFFSET (buffer);
+ } else {
+ change_offset = FALSE;
+ offset = 0;
+ }
+
+ if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {
+ offset_end = GST_BUFFER_OFFSET_END (buffer);
+ } else {
+ change_offset_end = FALSE;
+ offset_end = offset + size / frame_size;
+ }
+
+ if (segment->format == GST_FORMAT_TIME) {
+ /* Handle clipping for GST_FORMAT_TIME */
+
+ gint64 start, stop, cstart, cstop, diff;
+
+ start = timestamp;
+ stop = timestamp + duration;
+
+ if (gst_segment_clip (segment, GST_FORMAT_TIME,
+ start, stop, &cstart, &cstop)) {
+
+ diff = cstart - start;
+ if (diff > 0) {
+ timestamp = cstart;
+
+ if (change_duration)
+ duration -= diff;
+
+ diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
+ if (change_offset)
+ offset += diff;
+ data += diff * frame_size;
+ size -= diff * frame_size;
+ }
+
+ diff = stop - cstop;
+ if (diff > 0) {
+ /* duration is always valid if stop is valid */
+ duration -= diff;
+
+ diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
+ if (change_offset_end)
+ offset_end -= diff;
+ size -= diff * frame_size;
+ }
+ } else {
+ gst_buffer_unref (buffer);
+ return NULL;
+ }
+ } else {
+ /* Handle clipping for GST_FORMAT_DEFAULT */
+ gint64 start, stop, cstart, cstop, diff;
+
+ g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);
+
+ start = offset;
+ stop = offset_end;
+
+ if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,
+ start, stop, &cstart, &cstop)) {
+
+ diff = cstart - start;
+ if (diff > 0) {
+ offset = cstart;
+
+ timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);
+
+ if (change_duration)
+ duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
+
+ data += diff * frame_size;
+ size -= diff * frame_size;
+ }
+
+ diff = stop - cstop;
+ if (diff > 0) {
+ offset_end = cstop;
+
+ if (change_duration)
+ duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
+
+ size -= diff * frame_size;
+ }
+ } else {
+ gst_buffer_unref (buffer);
+ return NULL;
+ }
+ }
+
+ /* Get a metadata writable buffer and apply all changes */
+ ret = gst_buffer_make_metadata_writable (buffer);
+
+ GST_BUFFER_TIMESTAMP (ret) = timestamp;
+ GST_BUFFER_SIZE (ret) = size;
+ GST_BUFFER_DATA (ret) = data;
+
+ if (change_duration)
+ GST_BUFFER_DURATION (ret) = duration;
+ if (change_offset)
+ GST_BUFFER_OFFSET (ret) = offset;
+ if (change_offset_end)
+ GST_BUFFER_OFFSET_END (ret) = offset_end;
+
+ return ret;
+}
projects / mafwsubrenderer / blobdiff