"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&sink_factory));
+ gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
+ gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
gst_element_class_set_static_metadata (gstelement_class,
"Speex audio encoder", "Codec/Encoder/Audio",
"Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
/* arrange granulepos marking (and required perfect ts) */
gst_audio_encoder_set_mark_granule (benc, TRUE);
gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+ GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}
static gboolean
speex_bits_init (&enc->bits);
enc->tags = gst_tag_list_new_empty ();
enc->header_sent = FALSE;
+ enc->encoded_samples = 0;
return TRUE;
}
enc->state = NULL;
}
speex_bits_destroy (&enc->bits);
+ speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
gst_tag_list_unref (enc->tags);
enc->tags = NULL;
}
break;
}
+ case GST_EVENT_SEGMENT:
+ enc->encoded_samples = 0;
+ break;
default:
break;
}
gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
+ GstSegment *segment;
+ GstClockTime duration;
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
+ /* If encoding part of a frame, and we have no set stop time on
+ * the output segment, we update the segment stop time to reflect
+ * the last sample. This will let oggmux set the last page's
+ * granpos to tell a decoder the dummy samples should be clipped.
+ */
+ segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
+ GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
+ segment);
+ if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
+ int input_samples = bsize / (enc->channels * 2);
+ GST_DEBUG_OBJECT (enc,
+ "No stop time and partial frame, updating segment");
+ duration =
+ gst_util_uint64_scale (enc->encoded_samples + input_samples,
+ GST_SECOND, enc->rate);
+ segment->stop = segment->start + duration;
+ GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
+ segment);
+ gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ gst_event_new_segment (segment));
+ }
+
size = ((bsize / bytes) + 1) * bytes;
data0 = data = g_malloc0 (size);
memcpy (data, bdata, bsize);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples);
+ enc->encoded_samples += frame_size;
done:
g_free (data0);
/*
* (really really) FIXME: move into core (dixit tpm)
*/
-/**
+/*
* _gst_caps_set_buffer_array:
- * @caps: a #GstCaps
+ * @caps: (transfer full): a #GstCaps
* @field: field in caps to set
* @buf: header buffers
*
* Adds given buffers to an array of buffers set as the given @field
* on the given @caps. List of buffer arguments must be NULL-terminated.
*
- * Returns: input caps with a streamheader field added, or NULL if some error
+ * Returns: (transfer full): input caps with a streamheader field added, or NULL
+ * if some error occurred
*/
static GstCaps *
_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
buf = va_arg (va, GstBuffer *);
}
+ va_end (va);
gst_structure_set_value (structure, field, &array);
g_value_unset (&array);