diff options
Diffstat (limited to 'gst-libs/gst/audio/audio.c')
| -rw-r--r-- | gst-libs/gst/audio/audio.c | 433 |
1 files changed, 0 insertions, 433 deletions
diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c deleted file mode 100644 index 86b8885b..00000000 --- a/gst-libs/gst/audio/audio.c +++ /dev/null @@ -1,433 +0,0 @@ -/* 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; -#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; -} |