mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-08 13:18:52 +00:00
184 lines
6 KiB
C
184 lines
6 KiB
C
/* GStreamer
|
|
* Copyright (C) 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
*
|
|
* 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/audio.h>
|
|
|
|
#include "gstaudiosegmentclip.h"
|
|
|
|
static GstStaticPadTemplate sink_pad_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
|
|
|
|
static GstStaticPadTemplate src_pad_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)));
|
|
|
|
static void gst_audio_segment_clip_reset (GstSegmentClip * self);
|
|
static GstFlowReturn gst_audio_segment_clip_clip_buffer (GstSegmentClip * self,
|
|
GstBuffer * buffer, GstBuffer ** outbuf);
|
|
static gboolean gst_audio_segment_clip_set_caps (GstSegmentClip * self,
|
|
GstCaps * caps);
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_audio_segment_clip_debug);
|
|
#define GST_CAT_DEFAULT gst_audio_segment_clip_debug
|
|
|
|
G_DEFINE_TYPE (GstAudioSegmentClip, gst_audio_segment_clip,
|
|
GST_TYPE_SEGMENT_CLIP);
|
|
GST_ELEMENT_REGISTER_DEFINE (audiosegmentclip, "audiosegmentclip",
|
|
GST_RANK_NONE, GST_TYPE_AUDIO_SEGMENT_CLIP);
|
|
|
|
static void
|
|
gst_audio_segment_clip_class_init (GstAudioSegmentClipClass * klass)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
GstSegmentClipClass *segment_clip_klass = GST_SEGMENT_CLIP_CLASS (klass);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_audio_segment_clip_debug, "audiosegmentclip", 0,
|
|
"audiosegmentclip element");
|
|
|
|
segment_clip_klass->reset = GST_DEBUG_FUNCPTR (gst_audio_segment_clip_reset);
|
|
segment_clip_klass->set_caps =
|
|
GST_DEBUG_FUNCPTR (gst_audio_segment_clip_set_caps);
|
|
segment_clip_klass->clip_buffer =
|
|
GST_DEBUG_FUNCPTR (gst_audio_segment_clip_clip_buffer);
|
|
|
|
gst_element_class_set_static_metadata (element_class,
|
|
"Audio buffer segment clipper",
|
|
"Filter/Audio",
|
|
"Clips audio buffers to the configured segment",
|
|
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &sink_pad_template);
|
|
gst_element_class_add_static_pad_template (element_class, &src_pad_template);
|
|
}
|
|
|
|
static void
|
|
gst_audio_segment_clip_init (GstAudioSegmentClip * self)
|
|
{
|
|
}
|
|
|
|
static void
|
|
gst_audio_segment_clip_reset (GstSegmentClip * base)
|
|
{
|
|
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
|
|
|
|
GST_DEBUG_OBJECT (self, "Resetting internal state");
|
|
|
|
self->rate = self->framesize = 0;
|
|
}
|
|
|
|
|
|
static GstFlowReturn
|
|
gst_audio_segment_clip_clip_buffer (GstSegmentClip * base, GstBuffer * buffer,
|
|
GstBuffer ** outbuf)
|
|
{
|
|
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
|
|
GstSegment *segment = &base->segment;
|
|
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
GstClockTime duration = GST_BUFFER_DURATION (buffer);
|
|
guint64 offset = GST_BUFFER_OFFSET (buffer);
|
|
guint64 offset_end = GST_BUFFER_OFFSET_END (buffer);
|
|
guint size = gst_buffer_get_size (buffer);
|
|
|
|
if (!self->rate || !self->framesize) {
|
|
GST_ERROR_OBJECT (self, "Not negotiated yet");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
|
|
if (segment->format != GST_FORMAT_DEFAULT &&
|
|
segment->format != GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (self, "Unsupported segment format %s",
|
|
gst_format_get_name (segment->format));
|
|
*outbuf = buffer;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
|
GST_WARNING_OBJECT (self, "Buffer without valid timestamp");
|
|
*outbuf = buffer;
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
*outbuf =
|
|
gst_audio_buffer_clip (buffer, segment, self->rate, self->framesize);
|
|
|
|
if (!*outbuf) {
|
|
GST_DEBUG_OBJECT (self, "Buffer outside the configured segment");
|
|
|
|
/* Now return unexpected if we're before/after the end */
|
|
if (segment->format == GST_FORMAT_TIME) {
|
|
if (segment->rate >= 0) {
|
|
if (segment->stop != -1 && timestamp >= segment->stop)
|
|
return GST_FLOW_EOS;
|
|
} else {
|
|
if (!GST_CLOCK_TIME_IS_VALID (duration))
|
|
duration =
|
|
gst_util_uint64_scale_int (size, GST_SECOND,
|
|
self->framesize * self->rate);
|
|
|
|
if (segment->start != -1 && timestamp + duration <= segment->start)
|
|
return GST_FLOW_EOS;
|
|
}
|
|
} else {
|
|
if (segment->rate >= 0) {
|
|
if (segment->stop != -1 && offset != -1 && offset >= segment->stop)
|
|
return GST_FLOW_EOS;
|
|
} else if (offset != -1 || offset_end != -1) {
|
|
if (offset_end == -1)
|
|
offset_end = offset + size / self->framesize;
|
|
|
|
if (segment->start != -1 && offset_end <= segment->start)
|
|
return GST_FLOW_EOS;
|
|
}
|
|
}
|
|
}
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
|
|
{
|
|
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
|
|
gboolean ret;
|
|
GstAudioInfo info;
|
|
gint rate, channels, width;
|
|
|
|
gst_audio_info_init (&info);
|
|
ret = gst_audio_info_from_caps (&info, caps);
|
|
|
|
if (ret) {
|
|
rate = GST_AUDIO_INFO_RATE (&info);
|
|
channels = GST_AUDIO_INFO_CHANNELS (&info);
|
|
width = GST_AUDIO_INFO_WIDTH (&info);
|
|
|
|
GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d",
|
|
rate, channels, width);
|
|
self->rate = rate;
|
|
self->framesize = (width / 8) * channels;
|
|
}
|
|
|
|
return ret;
|
|
}
|