*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* <refsect2>
* <title>Example pipelines</title>
* |[
- * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
+ * gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
* ]| Encode an Ogg/Speex file.
* </refsect2>
*/
GstBuffer * in_buf);
static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
-static GstFlowReturn
-gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer);
#define gst_speex_enc_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_AUDIO_ENCODER,
base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame);
- base_class->event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
- base_class->pre_push = GST_DEBUG_FUNCPTR (gst_speex_enc_pre_push);
+ base_class->sink_event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_float ("quality", "Quality", "Encoding quality",
"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_set_details_simple (gstelement_class, "Speex audio encoder",
- "Codec/Encoder/Audio",
+ 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>");
GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder");
/* 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);
- gst_tag_list_free (enc->tags);
+ speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
+ gst_tag_list_unref (enc->tags);
enc->tags = NULL;
- g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
- enc->headers = NULL;
gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags);
comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
0, "Encoded with GStreamer Speexenc");
- gst_tag_list_free (merged_tags);
+ gst_tag_list_unref (merged_tags);
GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
return TRUE;
}
-/* push out the buffer */
-static GstFlowReturn
-gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
-{
- GST_DEBUG_OBJECT (enc, "pushing output buffer of size %" G_GSIZE_FORMAT,
- gst_buffer_get_size (buffer));
-
- return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
-}
-
static gboolean
gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
}
break;
}
+ case GST_EVENT_SEGMENT:
+ enc->encoded_samples = 0;
+ break;
default:
break;
}
/* we only peeked, let base class handle it */
- return FALSE;
+ return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
static GstFlowReturn
gint frame_size = enc->frame_size;
gint bytes = frame_size * 2 * enc->channels, samples;
gint outsize, written, dtx_ret = 0;
- guint8 *data, *data0 = NULL, *bdata, *outdata;
+ GstMapInfo map;
+ guint8 *data, *data0 = NULL, *bdata;
gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
+ GstSegment *segment;
+ GstClockTime duration;
if (G_LIKELY (buf)) {
- bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ bdata = map.data;
+ bsize = map.size;
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);
- gst_buffer_unmap (buf, bdata, bsize);
+ gst_buffer_unmap (buf, &map);
bdata = NULL;
} else {
data = bdata;
outsize = speex_bits_nbytes (&enc->bits);
if (bdata)
- gst_buffer_unmap (buf, bdata, bsize);
+ gst_buffer_unmap (buf, &map);
#if 0
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
if ((GST_FLOW_OK != ret))
goto done;
#endif
- outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
- outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
+ outbuf = gst_buffer_new_allocate (NULL, outsize, NULL);
+ gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
- written = speex_bits_write (&enc->bits, (gchar *) outdata, outsize);
+ written = speex_bits_write (&enc->bits, (gchar *) map.data, outsize);
if (G_UNLIKELY (written < outsize)) {
GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
written = outsize;
}
- gst_buffer_unmap (outbuf, outdata, written);
+ gst_buffer_unmap (outbuf, &map);
+ gst_buffer_resize (outbuf, 0, written);
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
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,
g_assert (gst_buffer_is_writable (buf));
/* mark buffer */
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
buf = va_arg (va, GstBuffer *);
}
+ va_end (va);
gst_structure_set_value (structure, field, &array);
g_value_unset (&array);
GstCaps *caps;
guchar *data;
gint data_len;
+ GList *headers;
/* create header buffer */
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
+ gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
gst_caps_unref (caps);
/* push out buffers */
/* store buffers for later pre_push sending */
- g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
- enc->headers = NULL;
+ headers = NULL;
GST_DEBUG_OBJECT (enc, "storing header buffers");
- enc->headers = g_slist_prepend (enc->headers, buf2);
- enc->headers = g_slist_prepend (enc->headers, buf1);
+ headers = g_list_prepend (headers, buf2);
+ headers = g_list_prepend (headers, buf1);
+ gst_audio_encoder_set_headers (benc, headers);
enc->header_sent = TRUE;
}
return ret;
}
-static GstFlowReturn
-gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
-{
- GstSpeexEnc *enc;
- GstFlowReturn ret = GST_FLOW_OK;
-
- enc = GST_SPEEX_ENC (benc);
-
- /* FIXME 0.11 ? get rid of this special ogg stuff and have it
- * put and use 'codec data' in caps like anything else,
- * with all the usual out-of-band advantage etc */
- if (G_UNLIKELY (enc->headers)) {
- GSList *header = enc->headers;
-
- /* try to push all of these, if we lose one, might as well lose all */
- while (header) {
- if (ret == GST_FLOW_OK)
- ret = gst_speex_enc_push_buffer (enc, header->data);
- else
- gst_speex_enc_push_buffer (enc, header->data);
- header = g_slist_next (header);
- }
-
- g_slist_free (enc->headers);
- enc->headers = NULL;
- }
-
- return ret;
-}
-
static void
gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
projects / platform / upstream / gst-plugins-good.git / blobdiff