Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.
fixes #585839
static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
- RTPSource * src, GstBuffer * buffer, gpointer user_data);
+ RTPSource * src, gpointer data, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
}
-/* called when the session manager has an RTP packet ready for further
- * processing */
+/* called when the session manager has an RTP packet or a list of packets
+ * ready for further processing */
static GstFlowReturn
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
* sending */
static GstFlowReturn
gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
- GstBuffer * buffer, gpointer user_data)
+ gpointer data, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
- GST_LOG_OBJECT (rtpsession, "sending RTP packet");
-
if (rtpsession->send_rtp_src) {
- result = gst_pad_push (rtpsession->send_rtp_src, buffer);
+ if (GST_IS_BUFFER (data)) {
+ GST_LOG_OBJECT (rtpsession, "sending RTP packet");
+ result = gst_pad_push (rtpsession->send_rtp_src, GST_BUFFER_CAST (data));
+ } else {
+ GST_LOG_OBJECT (rtpsession, "sending RTP list");
+ result = gst_pad_push_list (rtpsession->send_rtp_src,
+ GST_BUFFER_LIST_CAST (data));
+ }
} else {
- gst_buffer_unref (buffer);
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
result = GST_FLOW_OK;
}
return result;
return TRUE;
}
-/* Recieve an RTP packet to be send to the receivers, send to RTP session
- * manager and forward to send_rtp_src.
+/* Recieve an RTP packet or a list of packets to be send to the receivers,
+ * send to RTP session manager and forward to send_rtp_src.
*/
static GstFlowReturn
-gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
+gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data,
+ gboolean is_list)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
- GST_LOG_OBJECT (rtpsession, "received RTP packet");
+ GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ if (is_list) {
+ GstBuffer *buffer = NULL;
+
+ /* All groups in an list have the same timestamp.
+ * So, just take it from the first group. */
+ buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0);
+ if (buffer)
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ else
+ timestamp = -1;
+ } else {
+ timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
+ }
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment start value. */
ntpnstime =
}
current_time = gst_clock_get_time (priv->sysclock);
- ret = rtp_session_send_rtp (priv->session, buffer, current_time, ntpnstime);
+ ret =
+ rtp_session_send_rtp (priv->session, data, is_list, current_time,
+ ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
}
}
+static GstFlowReturn
+gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
+{
+ return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE);
+}
+
+static GstFlowReturn
+gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list)
+{
+ return gst_rtp_session_chain_send_rtp_common (pad, list, TRUE);
+}
+
/* Create sinkpad to receive RTP packets from senders. This will also create a
* srcpad for the RTP packets.
*/
"send_rtp_sink");
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
+ gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
+ gst_rtp_session_chain_send_rtp_list);
gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
gst_rtp_session_getcaps_send_rtp);
gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,
}
static GstFlowReturn
-source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
+source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
{
GstFlowReturn result = GST_FLOW_OK;
if (session->callbacks.send_rtp)
result =
- session->callbacks.send_rtp (session, source, buffer,
+ session->callbacks.send_rtp (session, source, data,
session->send_rtp_user_data);
- else
- gst_buffer_unref (buffer);
-
+ else {
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+ }
} else {
GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
result =
- session->callbacks.process_rtp (session, source, buffer,
- session->process_rtp_user_data);
+ session->callbacks.process_rtp (session, source,
+ GST_BUFFER_CAST (data), session->process_rtp_user_data);
else
- gst_buffer_unref (buffer);
+ gst_buffer_unref (GST_BUFFER_CAST (data));
}
RTP_SESSION_LOCK (session);
/**
* rtp_session_send_rtp:
* @sess: an #RTPSession
- * @buffer: an RTP buffer
+ * @data: pointer to either an RTP buffer or a list of RTP buffers
* @current_time: the current system time
* @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
* This is the buffer timestamp converted to NTP time.
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
+rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
GstClockTime current_time, guint64 ntpnstime)
{
GstFlowReturn result;
RTPSource *source;
gboolean prevsender;
+ gboolean valid_packet;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
- g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
- if (!gst_rtp_buffer_validate (buffer))
+ if (is_list) {
+ valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
+ } else {
+ valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
+ }
+
+ if (!valid_packet)
goto invalid_packet;
- GST_LOG ("received RTP packet for sending");
+ GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
RTP_SESSION_LOCK (sess);
source = sess->source;
prevsender = RTP_SOURCE_IS_SENDER (source);
/* we use our own source to send */
- result = rtp_source_send_rtp (source, buffer, ntpnstime);
+ result = rtp_source_send_rtp (source, data, is_list, ntpnstime);
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
/* ERRORS */
invalid_packet:
{
- gst_buffer_unref (buffer);
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}
*
* Returns: a #GstFlowReturn.
*/
-typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
+typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gpointer data, gpointer user_data);
/**
* RTPSessionSendRTCP:
GstClockTime current_time);
/* processing packets for sending */
-GstFlowReturn rtp_session_send_rtp (RTPSession *sess, GstBuffer *buffer,
+GstFlowReturn rtp_session_send_rtp (RTPSession *sess, gpointer data, gboolean is_list,
GstClockTime current_time, guint64 ntpnstime);
/* stopping the session */
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
-#define RTP_MAX_PROBATION_LEN 32
+#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
src->received_bye = TRUE;
}
+static GstBufferListItem
+set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
+{
+ *buffer = gst_buffer_make_writable (*buffer);
+ gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
+ return GST_BUFFER_LIST_SKIP_GROUP;
+}
+
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
- * @buffer: an RTP buffer
+ * @data: an RTP buffer or a list of RTP buffers
+ * @is_list: if @data is a buffer or list
* @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This
* is the buffer timestamp converted to NTP time.
*
- * Send an RTP @buffer originating from @src. This will make @src a sender.
- * This function takes ownership of @buffer and modifies the SSRC in the RTP
- * packet to that of @src when needed.
+ * Send @data (an RTP buffer or list of buffers) originating from @src.
+ * This will make @src a sender. This function takes ownership of @data and
+ * modifies the SSRC in the RTP packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
+rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
+ guint64 ntpnstime)
{
- GstFlowReturn result = GST_FLOW_OK;
+ GstFlowReturn result;
guint len;
guint32 rtptime;
guint64 ext_rtptime;
guint64 ntp_diff, rtp_diff;
guint64 elapsed;
+ GstBufferList *list = NULL;
+ GstBuffer *buffer = NULL;
+ guint packets;
+ guint32 ssrc;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
- g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
- len = gst_rtp_buffer_get_payload_len (buffer);
+ if (is_list) {
+ list = GST_BUFFER_LIST_CAST (data);
+ /* We can grab the caps from the first group, since all
+ * groups of a buffer list have same caps. */
+ buffer = gst_buffer_list_get (list, 0, 0);
+ if (!buffer)
+ goto no_buffer;
+ } else {
+ buffer = GST_BUFFER_CAST (data);
+ }
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
/* we are a sender now */
src->is_sender = TRUE;
+ if (is_list) {
+ /* Each group makes up a network packet. */
+ packets = gst_buffer_list_n_groups (list);
+ len = gst_rtp_buffer_list_get_payload_len (list);
+ } else {
+ packets = 1;
+ len = gst_rtp_buffer_get_payload_len (buffer);
+ }
+
/* update stats for the SR */
- src->stats.packets_sent++;
+ src->stats.packets_sent += packets;
src->stats.octets_sent += len;
src->bytes_sent += len;
src->bitrate = 0;
}
- rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ if (is_list) {
+ rtptime = gst_rtp_buffer_list_get_timestamp (list);
+ } else {
+ rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ }
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
src->last_ntpnstime = ntpnstime;
/* push packet */
- if (src->callbacks.push_rtp) {
- guint32 ssrc;
+ if (!src->callbacks.push_rtp)
+ goto no_callback;
+ if (is_list) {
+ ssrc = gst_rtp_buffer_list_get_ssrc (list);
+ } else {
ssrc = gst_rtp_buffer_get_ssrc (buffer);
- if (ssrc != src->ssrc) {
- /* the SSRC of the packet is not correct, make a writable buffer and
- * update the SSRC. This could involve a complete copy of the packet when
- * it is not writable. Usually the payloader will use caps negotiation to
- * get the correct SSRC from the session manager before pushing anything. */
- buffer = gst_buffer_make_writable (buffer);
-
- /* FIXME, we don't want to warn yet because we can't inform any payloader
- * of the changes SSRC yet because we don't implement pad-alloc. */
- GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
- src->ssrc);
- gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
+ }
+
+ if (ssrc != src->ssrc) {
+ /* the SSRC of the packet is not correct, make a writable buffer and
+ * update the SSRC. This could involve a complete copy of the packet when
+ * it is not writable. Usually the payloader will use caps negotiation to
+ * get the correct SSRC from the session manager before pushing anything. */
+
+ /* FIXME, we don't want to warn yet because we can't inform any payloader
+ * of the changes SSRC yet because we don't implement pad-alloc. */
+ GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
+ src->ssrc);
+
+ if (is_list) {
+ list = gst_buffer_list_make_writable (list);
+ gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
+ } else {
+ set_ssrc (&buffer, 0, 0, src);
}
- GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
- result = src->callbacks.push_rtp (src, buffer, src->user_data);
- } else {
- GST_WARNING ("no callback installed, dropping packet");
- gst_buffer_unref (buffer);
}
+ GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
+ src->stats.packets_sent);
+
+ result = src->callbacks.push_rtp (src, data, src->user_data);
return result;
+
+ /* ERRORS */
+no_buffer:
+ {
+ GST_WARNING ("no buffers in buffer list");
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+ return GST_FLOW_OK;
+ }
+no_callback:
+ {
+ GST_WARNING ("no callback installed, dropping packet");
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+ return GST_FLOW_OK;
+ }
}
/**
/* handling RTP */
GstFlowReturn rtp_source_process_rtp (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
-GstFlowReturn rtp_source_send_rtp (RTPSource *src, GstBuffer *buffer, guint64 ntpnstime);
+GstFlowReturn rtp_source_send_rtp (RTPSource *src, gpointer data, gboolean is_list, guint64 ntpnstime);
/* RTCP messages */
void rtp_source_process_bye (RTPSource *src, const gchar *reason);
--- /dev/null
+/* GStreamer
+ *
+ * Unit test for gstrtpbin sending rtp packets using GstBufferList.
+ * Copyright (C) 2009 Branko Subasic <branko dot subasic at axis dot com>
+ *
+ * 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.
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+
+
+/* This test makes sure that RTP packets sent as buffer lists are sent through
+ * the rtpbin as they are supposed to, and not corrupted in any way.
+ */
+
+
+#define TEST_CAPS \
+ "application/x-rtp, " \
+ "media=(string)video, " \
+ "clock-rate=(int)90000, " \
+ "encoding-name=(string)H264, " \
+ "profile-level-id=(string)4d4015, " \
+ "payload=(int)96, " \
+ "ssrc=(guint)2633237432, " \
+ "clock-base=(guint)1868267015, " \
+ "seqnum-base=(guint)54229"
+
+
+/* RTP headers and the first 2 bytes of the payload (FU indicator and FU header)
+ */
+static const guint8 rtp_header[2][14] = {
+ {0x80, 0x60, 0xbb, 0xb7, 0x5c, 0xe9, 0x09,
+ 0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x86},
+ {0x80, 0x60, 0xbb, 0xb8, 0x5c, 0xe9, 0x09,
+ 0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x46}
+};
+
+static const guint rtp_header_len[] = {
+ sizeof rtp_header[0],
+ sizeof rtp_header[1]
+};
+
+static GstBuffer *header_buffer[2] = { NULL, NULL };
+
+
+/* Some payload.
+ */
+static char *payload =
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
+ "0123456789ABSDEF0123456";
+
+static const guint payload_offset[] = {
+ 0, 498
+};
+
+static const guint payload_len[] = {
+ 498, 5
+};
+
+
+static GstBuffer *original_buffer = NULL;
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp"));
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp"));
+
+
+static GstBuffer *
+_create_original_buffer (void)
+{
+ GstCaps *caps;
+
+ if (original_buffer != NULL)
+ return original_buffer;
+
+ original_buffer = gst_buffer_new ();
+ fail_unless (original_buffer != NULL);
+
+ gst_buffer_set_data (original_buffer, (guint8 *) payload, strlen (payload));
+ GST_BUFFER_TIMESTAMP (original_buffer) =
+ gst_clock_get_internal_time (gst_system_clock_obtain ());
+
+ caps = gst_caps_from_string (TEST_CAPS);
+ fail_unless (caps != NULL);
+ gst_buffer_set_caps (original_buffer, caps);
+ gst_caps_unref (caps);
+
+ return original_buffer;
+}
+
+static GstBufferList *
+_create_buffer_list (void)
+{
+ GstBufferList *list;
+ GstBufferListIterator *it;
+ GstBuffer *orig_buffer;
+ GstBuffer *buffer;
+
+ orig_buffer = _create_original_buffer ();
+ fail_if (orig_buffer == NULL);
+
+ list = gst_buffer_list_new ();
+ fail_if (list == NULL);
+
+ it = gst_buffer_list_iterate (list);
+ fail_if (it == NULL);
+
+ /*** First group, i.e. first packet. **/
+ gst_buffer_list_iterator_add_group (it);
+
+ /* Create buffer with RTP header and add it to the 1st group */
+ buffer = gst_buffer_new ();
+ GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[0], rtp_header_len[0]);
+ GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+ GST_BUFFER_SIZE (buffer) = rtp_header_len[0];
+ gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
+ header_buffer[0] = buffer;
+ gst_buffer_list_iterator_add (it, buffer);
+
+ /* Create the payload buffer and add it to the 1st group
+ */
+ buffer =
+ gst_buffer_create_sub (orig_buffer, payload_offset[0], payload_len[0]);
+ fail_if (buffer == NULL);
+ gst_buffer_list_iterator_add (it, buffer);
+
+
+ /*** Second group, i.e. second packet. ***/
+
+ /* Create a new group to hold the rtp header and the payload */
+ gst_buffer_list_iterator_add_group (it);
+
+ /* Create buffer with RTP header and add it to the 2nd group */
+ buffer = gst_buffer_new ();
+ GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[1], rtp_header_len[1]);
+ GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+ GST_BUFFER_SIZE (buffer) = rtp_header_len[1];
+ gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
+ header_buffer[1] = buffer;
+
+ /* Add the rtp header to the buffer list */
+ gst_buffer_list_iterator_add (it, buffer);
+
+ /* Create the payload buffer and add it to the 2d group
+ */
+ buffer =
+ gst_buffer_create_sub (orig_buffer, payload_offset[1], payload_len[1]);
+ fail_if (buffer == NULL);
+ gst_buffer_list_iterator_add (it, buffer);
+
+ gst_buffer_list_iterator_free (it);
+
+ return list;
+}
+
+
+static void
+_check_header (GstBuffer * buffer, guint index)
+{
+ guint8 *data;
+
+ fail_if (buffer == NULL);
+ fail_unless (index < 2);
+
+ fail_unless (GST_BUFFER_SIZE (buffer) == rtp_header_len[index]);
+
+ /* Can't do a memcmp() on the whole header, cause the SSRC (bytes 8-11) will
+ * most likely be changed in gstrtpbin.
+ */
+ fail_unless ((data = GST_BUFFER_DATA (buffer)) != NULL);
+ fail_unless_equals_uint64 (*(guint64 *) data, *(guint64 *) rtp_header[index]);
+ fail_unless (*(guint16 *) (data + 12) ==
+ *(guint16 *) (rtp_header[index] + 12));
+}
+
+
+static void
+_check_payload (GstBuffer * buffer, guint index)
+{
+ fail_if (buffer == NULL);
+ fail_unless (index < 2);
+
+ fail_unless (GST_BUFFER_SIZE (buffer) == payload_len[index]);
+ fail_if (GST_BUFFER_DATA (buffer) !=
+ (gpointer) (payload + payload_offset[index]));
+ fail_if (memcmp (GST_BUFFER_DATA (buffer), payload + payload_offset[index],
+ payload_len[index]));
+}
+
+
+static void
+_check_group (GstBufferListIterator * it, guint index, GstCaps * caps)
+{
+ GstBuffer *buffer;
+
+ fail_unless (it != NULL);
+ fail_unless (gst_buffer_list_iterator_n_buffers (it) == 2);
+ fail_unless (caps != NULL);
+
+ fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
+
+ fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
+ GST_BUFFER_TIMESTAMP (original_buffer));
+
+ fail_unless (gst_caps_is_equal (GST_BUFFER_CAPS (original_buffer),
+ GST_BUFFER_CAPS (buffer)));
+
+ _check_header (buffer, index);
+
+ fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
+ _check_payload (buffer, index);
+}
+
+
+static GstFlowReturn
+_sink_chain_list (GstPad * pad, GstBufferList * list)
+{
+ GstCaps *caps;
+ GstBufferListIterator *it;
+
+ caps = gst_caps_from_string (TEST_CAPS);
+ fail_unless (caps != NULL);
+
+ fail_unless (GST_IS_BUFFER_LIST (list));
+ fail_unless (gst_buffer_list_n_groups (list) == 2);
+
+ it = gst_buffer_list_iterate (list);
+ fail_if (it == NULL);
+
+ fail_unless (gst_buffer_list_iterator_next_group (it));
+ _check_group (it, 0, caps);
+
+ fail_unless (gst_buffer_list_iterator_next_group (it));
+ _check_group (it, 1, caps);
+
+ gst_caps_unref (caps);
+ gst_buffer_list_iterator_free (it);
+
+ gst_buffer_list_unref (list);
+
+ return GST_FLOW_OK;
+}
+
+
+static void
+_set_chain_functions (GstPad * pad)
+{
+ gst_pad_set_chain_list_function (pad, _sink_chain_list);
+}
+
+
+GST_START_TEST (test_bufferlist)
+{
+ GstElement *rtpbin;
+ GstPad *sinkpad;
+ GstPad *srcpad;
+ GstBufferList *list;
+
+ list = _create_buffer_list ();
+ fail_unless (list != NULL);
+
+ rtpbin = gst_check_setup_element ("gstrtpbin");
+
+ srcpad =
+ gst_check_setup_src_pad_by_name (rtpbin, &srctemplate, "send_rtp_sink_0");
+ fail_if (srcpad == NULL);
+ sinkpad =
+ gst_check_setup_sink_pad_by_name (rtpbin, &sinktemplate,
+ "send_rtp_src_0");
+ fail_if (sinkpad == NULL);
+
+ _set_chain_functions (sinkpad);
+
+ gst_pad_set_active (sinkpad, TRUE);
+ gst_element_set_state (rtpbin, GST_STATE_PLAYING);
+ fail_unless (gst_pad_push_list (srcpad, list) == GST_FLOW_OK);
+ gst_pad_set_active (sinkpad, FALSE);
+
+ gst_check_teardown_pad_by_name (rtpbin, "send_rtp_src_0");
+ gst_check_teardown_pad_by_name (rtpbin, "send_rtp_sink_0");
+ gst_check_teardown_element (rtpbin);
+}
+
+GST_END_TEST;
+
+
+
+static Suite *
+bufferlist_suite (void)
+{
+ Suite *s = suite_create ("BufferList");
+
+ TCase *tc_chain = tcase_create ("general");
+
+ /* time out after 30s. */
+ tcase_set_timeout (tc_chain, 10);
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_bufferlist);
+
+ return s;
+}
+
+GST_CHECK_MAIN (bufferlist);
projects / platform / upstream / gstreamer.git / commitdiff