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 gboolean
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
{
GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
gboolean ret;
GstStructure *s;
gint rate, channels, width;
s = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (s, "rate", &rate);
ret = ret && gst_structure_get_int (s, "channels", &channels);
ret = ret && gst_structure_get_int (s, "width", &width);
if (ret) {
GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d", rate,
channels, width);
self->rate = rate;
self->framesize = (width / 8) * channels;
}
return ret;
}
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;
}
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;
}
