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gstaudiometa.c:382: Warning: GstAudio: gst_buffer_add_audio_meta: return value: Invalid non-constant return of bare structure or union; register as boxed type or (skip)
494 lines
15 KiB
C
494 lines
15 KiB
C
/* GStreamer
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* Copyright (C) <2011> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudiometa
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* @title: GstAudio meta
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* @short_description: Buffer metadata for audio downmix matrix handling
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*
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* #GstAudioDownmixMeta defines an audio downmix matrix to be send along with
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* audio buffers. These functions in this module help to create and attach the
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* meta as well as extracting it.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstaudiometa.h"
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static gboolean
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gst_audio_downmix_meta_init (GstMeta * meta, gpointer params,
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GstBuffer * buffer)
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{
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GstAudioDownmixMeta *dmeta = (GstAudioDownmixMeta *) meta;
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dmeta->from_position = dmeta->to_position = NULL;
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dmeta->from_channels = dmeta->to_channels = 0;
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dmeta->matrix = NULL;
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return TRUE;
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}
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static void
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gst_audio_downmix_meta_free (GstMeta * meta, GstBuffer * buffer)
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{
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GstAudioDownmixMeta *dmeta = (GstAudioDownmixMeta *) meta;
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g_free (dmeta->from_position);
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if (dmeta->matrix) {
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g_free (*dmeta->matrix);
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g_free (dmeta->matrix);
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}
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}
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static gboolean
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gst_audio_downmix_meta_transform (GstBuffer * dest, GstMeta * meta,
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GstBuffer * buffer, GQuark type, gpointer data)
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{
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GstAudioDownmixMeta *smeta, *dmeta;
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smeta = (GstAudioDownmixMeta *) meta;
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if (GST_META_TRANSFORM_IS_COPY (type)) {
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dmeta = gst_buffer_add_audio_downmix_meta (dest, smeta->from_position,
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smeta->from_channels, smeta->to_position, smeta->to_channels,
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(const gfloat **) smeta->matrix);
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if (!dmeta)
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return FALSE;
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} else {
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/* return FALSE, if transform type is not supported */
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return FALSE;
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}
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return TRUE;
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}
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/**
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* gst_buffer_get_audio_downmix_meta_for_channels:
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* @buffer: a #GstBuffer
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* @to_position: (array length=to_channels): the channel positions of
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* the destination
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* @to_channels: The number of channels of the destination
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*
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* Find the #GstAudioDownmixMeta on @buffer for the given destination
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* channel positions.
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*
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* Returns: (transfer none): the #GstAudioDownmixMeta on @buffer.
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*/
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GstAudioDownmixMeta *
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gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer * buffer,
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const GstAudioChannelPosition * to_position, gint to_channels)
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{
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gpointer state = NULL;
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GstMeta *meta;
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const GstMetaInfo *info = GST_AUDIO_DOWNMIX_META_INFO;
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while ((meta = gst_buffer_iterate_meta (buffer, &state))) {
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if (meta->info->api == info->api) {
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GstAudioDownmixMeta *ameta = (GstAudioDownmixMeta *) meta;
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if (ameta->to_channels == to_channels &&
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memcmp (ameta->to_position, to_position,
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sizeof (GstAudioChannelPosition) * to_channels) == 0)
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return ameta;
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}
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}
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return NULL;
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}
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/**
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* gst_buffer_add_audio_downmix_meta:
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* @buffer: a #GstBuffer
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* @from_position: (array length=from_channels): the channel positions
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* of the source
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* @from_channels: The number of channels of the source
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* @to_position: (array length=to_channels): the channel positions of
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* the destination
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* @to_channels: The number of channels of the destination
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* @matrix: The matrix coefficients.
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*
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* Attaches #GstAudioDownmixMeta metadata to @buffer with the given parameters.
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*
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* @matrix is an two-dimensional array of @to_channels times @from_channels
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* coefficients, i.e. the i-th output channels is constructed by multiplicating
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* the input channels with the coefficients in @matrix[i] and taking the sum
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* of the results.
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*
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* Returns: (transfer none): the #GstAudioDownmixMeta on @buffer.
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*/
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GstAudioDownmixMeta *
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gst_buffer_add_audio_downmix_meta (GstBuffer * buffer,
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const GstAudioChannelPosition * from_position, gint from_channels,
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const GstAudioChannelPosition * to_position, gint to_channels,
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const gfloat ** matrix)
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{
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GstAudioDownmixMeta *meta;
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gint i;
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g_return_val_if_fail (from_position != NULL, NULL);
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g_return_val_if_fail (from_channels > 0, NULL);
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g_return_val_if_fail (to_position != NULL, NULL);
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g_return_val_if_fail (to_channels > 0, NULL);
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g_return_val_if_fail (matrix != NULL, NULL);
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meta =
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(GstAudioDownmixMeta *) gst_buffer_add_meta (buffer,
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GST_AUDIO_DOWNMIX_META_INFO, NULL);
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meta->from_channels = from_channels;
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meta->to_channels = to_channels;
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meta->from_position =
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g_new (GstAudioChannelPosition, meta->from_channels + meta->to_channels);
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meta->to_position = meta->from_position + meta->from_channels;
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memcpy (meta->from_position, from_position,
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sizeof (GstAudioChannelPosition) * meta->from_channels);
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memcpy (meta->to_position, to_position,
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sizeof (GstAudioChannelPosition) * meta->to_channels);
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meta->matrix = g_new (gfloat *, meta->to_channels);
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meta->matrix[0] = g_new (gfloat, meta->from_channels * meta->to_channels);
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memcpy (meta->matrix[0], matrix[0], sizeof (gfloat) * meta->from_channels);
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for (i = 1; i < meta->to_channels; i++) {
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meta->matrix[i] = meta->matrix[0] + i * meta->from_channels;
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memcpy (meta->matrix[i], matrix[i], sizeof (gfloat) * meta->from_channels);
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}
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return meta;
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}
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GType
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gst_audio_downmix_meta_api_get_type (void)
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{
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static volatile GType type;
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static const gchar *tags[] =
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{ GST_META_TAG_AUDIO_STR, GST_META_TAG_AUDIO_CHANNELS_STR, NULL };
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if (g_once_init_enter (&type)) {
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GType _type = gst_meta_api_type_register ("GstAudioDownmixMetaAPI", tags);
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g_once_init_leave (&type, _type);
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}
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return type;
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}
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const GstMetaInfo *
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gst_audio_downmix_meta_get_info (void)
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{
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static const GstMetaInfo *audio_downmix_meta_info = NULL;
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if (g_once_init_enter ((GstMetaInfo **) & audio_downmix_meta_info)) {
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const GstMetaInfo *meta =
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gst_meta_register (GST_AUDIO_DOWNMIX_META_API_TYPE,
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"GstAudioDownmixMeta", sizeof (GstAudioDownmixMeta),
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gst_audio_downmix_meta_init, gst_audio_downmix_meta_free,
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gst_audio_downmix_meta_transform);
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g_once_init_leave ((GstMetaInfo **) & audio_downmix_meta_info,
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(GstMetaInfo *) meta);
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}
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return audio_downmix_meta_info;
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}
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static gboolean
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gst_audio_clipping_meta_init (GstMeta * meta, gpointer params,
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GstBuffer * buffer)
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{
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GstAudioClippingMeta *cmeta = (GstAudioClippingMeta *) meta;
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cmeta->format = GST_FORMAT_UNDEFINED;
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cmeta->start = cmeta->end = 0;
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return TRUE;
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}
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static gboolean
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gst_audio_clipping_meta_transform (GstBuffer * dest, GstMeta * meta,
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GstBuffer * buffer, GQuark type, gpointer data)
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{
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GstAudioClippingMeta *smeta, *dmeta;
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smeta = (GstAudioClippingMeta *) meta;
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if (GST_META_TRANSFORM_IS_COPY (type)) {
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GstMetaTransformCopy *copy = data;
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if (copy->region)
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return FALSE;
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dmeta =
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gst_buffer_add_audio_clipping_meta (dest, smeta->format, smeta->start,
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smeta->end);
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if (!dmeta)
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return FALSE;
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} else {
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/* TODO: Could implement an automatic transform for resampling */
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/* return FALSE, if transform type is not supported */
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return FALSE;
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}
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return TRUE;
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}
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/**
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* gst_buffer_add_audio_clipping_meta:
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* @buffer: a #GstBuffer
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* @format: GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
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* @start: Amount of audio to clip from start of buffer
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* @end: Amount of to clip from end of buffer
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*
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* Attaches #GstAudioClippingMeta metadata to @buffer with the given parameters.
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*
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* Returns: (transfer none): the #GstAudioClippingMeta on @buffer.
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*
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* Since: 1.8
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*/
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GstAudioClippingMeta *
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gst_buffer_add_audio_clipping_meta (GstBuffer * buffer,
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GstFormat format, guint64 start, guint64 end)
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{
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GstAudioClippingMeta *meta;
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g_return_val_if_fail (format != GST_FORMAT_UNDEFINED, NULL);
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meta =
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(GstAudioClippingMeta *) gst_buffer_add_meta (buffer,
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GST_AUDIO_CLIPPING_META_INFO, NULL);
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meta->format = format;
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meta->start = start;
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meta->end = end;
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return meta;
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}
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GType
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gst_audio_clipping_meta_api_get_type (void)
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{
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static volatile GType type;
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static const gchar *tags[] =
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{ GST_META_TAG_AUDIO_STR, GST_META_TAG_AUDIO_RATE_STR, NULL };
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if (g_once_init_enter (&type)) {
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GType _type = gst_meta_api_type_register ("GstAudioClippingMetaAPI", tags);
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g_once_init_leave (&type, _type);
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}
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return type;
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}
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const GstMetaInfo *
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gst_audio_clipping_meta_get_info (void)
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{
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static const GstMetaInfo *audio_clipping_meta_info = NULL;
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if (g_once_init_enter ((GstMetaInfo **) & audio_clipping_meta_info)) {
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const GstMetaInfo *meta =
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gst_meta_register (GST_AUDIO_CLIPPING_META_API_TYPE,
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"GstAudioClippingMeta", sizeof (GstAudioClippingMeta),
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gst_audio_clipping_meta_init, NULL,
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gst_audio_clipping_meta_transform);
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g_once_init_leave ((GstMetaInfo **) & audio_clipping_meta_info,
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(GstMetaInfo *) meta);
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}
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return audio_clipping_meta_info;
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}
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static gboolean
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gst_audio_meta_init (GstMeta * meta, gpointer params, GstBuffer * buffer)
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{
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GstAudioMeta *ameta = (GstAudioMeta *) meta;
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gst_audio_info_init (&ameta->info);
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ameta->samples = 0;
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ameta->offsets = NULL;
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return TRUE;
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}
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static void
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gst_audio_meta_free (GstMeta * meta, GstBuffer * buffer)
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{
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GstAudioMeta *ameta = (GstAudioMeta *) meta;
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if (ameta->offsets && ameta->offsets != ameta->priv_offsets_arr)
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g_slice_free1 (ameta->info.channels * sizeof (gsize), ameta->offsets);
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}
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static gboolean
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gst_audio_meta_transform (GstBuffer * dest, GstMeta * meta,
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GstBuffer * buffer, GQuark type, gpointer data)
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{
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GstAudioMeta *smeta, *dmeta;
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smeta = (GstAudioMeta *) meta;
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if (GST_META_TRANSFORM_IS_COPY (type)) {
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dmeta = gst_buffer_add_audio_meta (dest, &smeta->info, smeta->samples,
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smeta->offsets);
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if (!dmeta)
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return FALSE;
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} else {
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/* return FALSE, if transform type is not supported */
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return FALSE;
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}
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return TRUE;
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}
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/**
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* gst_buffer_add_audio_meta:
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* @buffer: a #GstBuffer
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* @info: the audio properties of the buffer
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* @samples: the number of valid samples in the buffer
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* @offsets: (nullable): the offsets (in bytes) where each channel plane starts
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* in the buffer or %NULL to calculate it (see below); must be %NULL also
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* when @info->layout is %GST_AUDIO_LAYOUT_INTERLEAVED
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*
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* Allocates and attaches a #GstAudioMeta on @buffer, which must be writable
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* for that purpose. The fields of the #GstAudioMeta are directly populated
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* from the arguments of this function.
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*
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* When @info->layout is %GST_AUDIO_LAYOUT_NON_INTERLEAVED and @offsets is
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* %NULL, the offsets are calculated with a formula that assumes the planes are
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* tightly packed and in sequence:
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* offsets[channel] = channel * @samples * sample_stride
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*
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* It is not allowed for channels to overlap in memory,
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* i.e. for each i in [0, channels), the range
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* [@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap
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* with any other such range. This function will assert if the parameters
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* specified cause this restriction to be violated.
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*
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* It is, obviously, also not allowed to specify parameters that would cause
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* out-of-bounds memory access on @buffer. This is also checked, which means
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* that you must add enough memory on the @buffer before adding this meta.
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*
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* Returns: (transfer none): the #GstAudioMeta that was attached on the @buffer
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*
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* Since: 1.16
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*/
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GstAudioMeta *
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gst_buffer_add_audio_meta (GstBuffer * buffer, const GstAudioInfo * info,
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gsize samples, gsize offsets[])
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{
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GstAudioMeta *meta;
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gint i;
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gsize plane_size;
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g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
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g_return_val_if_fail (info != NULL, NULL);
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g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (info), NULL);
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g_return_val_if_fail (GST_AUDIO_INFO_FORMAT (info) !=
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GST_AUDIO_FORMAT_UNKNOWN, NULL);
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g_return_val_if_fail (info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED
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|| !offsets, NULL);
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meta =
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(GstAudioMeta *) gst_buffer_add_meta (buffer, GST_AUDIO_META_INFO, NULL);
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meta->info = *info;
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meta->samples = samples;
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plane_size = samples * info->finfo->width / 8;
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if (info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
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#ifndef G_DISABLE_CHECKS
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gsize max_offset = 0;
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gint j;
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#endif
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if (G_UNLIKELY (info->channels > 8))
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meta->offsets = g_slice_alloc (info->channels * sizeof (gsize));
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else
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meta->offsets = meta->priv_offsets_arr;
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if (offsets) {
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for (i = 0; i < info->channels; i++) {
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meta->offsets[i] = offsets[i];
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#ifndef G_DISABLE_CHECKS
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max_offset = MAX (max_offset, offsets[i]);
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for (j = 0; j < info->channels; j++) {
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if (i != j && !(offsets[j] + plane_size <= offsets[i]
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|| offsets[i] + plane_size <= offsets[j])) {
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g_critical ("GstAudioMeta properties would cause channel memory "
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"areas to overlap! offsets: %" G_GSIZE_FORMAT " (%d), %"
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G_GSIZE_FORMAT " (%d) with plane size %" G_GSIZE_FORMAT,
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offsets[i], i, offsets[j], j, plane_size);
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gst_buffer_remove_meta (buffer, (GstMeta *) meta);
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return NULL;
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}
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}
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#endif
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}
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} else {
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/* default offsets assume channels are laid out sequentially in memory */
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for (i = 0; i < info->channels; i++)
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meta->offsets[i] = i * plane_size;
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#ifndef G_DISABLE_CHECKS
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max_offset = meta->offsets[info->channels - 1];
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#endif
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}
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#ifndef G_DISABLE_CHECKS
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if (max_offset + plane_size > gst_buffer_get_size (buffer)) {
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g_critical ("GstAudioMeta properties would cause "
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"out-of-bounds memory access on the buffer: max_offset %"
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G_GSIZE_FORMAT ", samples %" G_GSIZE_FORMAT ", bps %u, buffer size %"
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G_GSIZE_FORMAT, max_offset, samples, info->finfo->width / 8,
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gst_buffer_get_size (buffer));
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gst_buffer_remove_meta (buffer, (GstMeta *) meta);
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return NULL;
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}
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#endif
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}
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return meta;
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}
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GType
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gst_audio_meta_api_get_type (void)
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{
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static volatile GType type;
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static const gchar *tags[] = {
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GST_META_TAG_AUDIO_STR, GST_META_TAG_AUDIO_CHANNELS_STR,
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GST_META_TAG_AUDIO_RATE_STR, NULL
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};
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if (g_once_init_enter (&type)) {
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GType _type = gst_meta_api_type_register ("GstAudioMetaAPI", tags);
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g_once_init_leave (&type, _type);
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}
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return type;
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}
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const GstMetaInfo *
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gst_audio_meta_get_info (void)
|
|
{
|
|
static const GstMetaInfo *audio_meta_info = NULL;
|
|
|
|
if (g_once_init_enter ((GstMetaInfo **) & audio_meta_info)) {
|
|
const GstMetaInfo *meta = gst_meta_register (GST_AUDIO_META_API_TYPE,
|
|
"GstAudioMeta", sizeof (GstAudioMeta),
|
|
gst_audio_meta_init,
|
|
gst_audio_meta_free,
|
|
gst_audio_meta_transform);
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|
g_once_init_leave ((GstMetaInfo **) & audio_meta_info,
|
|
(GstMetaInfo *) meta);
|
|
}
|
|
return audio_meta_info;
|
|
}
|