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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, 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);
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_pad_template (element_class,
gst_static_pad_template_get (&sink_pad_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&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;
}
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