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rtpbasepayload: rtpbasedepayload: Add source-info property

Browse source 
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947
This commit is contained in:
Stian Selnes 2018-10-10 14:16:12 -04:00 committed by Olivier Crête
parent 1b9ed134d1
commit f766b85b96
17 changed files with 972 additions and 43 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
docs/libs/gst-plugins-base-libs-docs.sgml
View file

@ -110,6 +110,7 @@
<filename>gstreamer-plugins-base-&GST_API_VERSION;.pc</filename> and adding
<filename>-lgstrtp-&GST_API_VERSION;</filename> to the library flags.
</para>
<xi:include href="xml/gstrtpmeta.xml" />
<xi:include href="xml/gstrtpbaseaudiopayload.xml" />
<xi:include href="xml/gstrtpbasedepayload.xml" />
<xi:include href="xml/gstrtpbasepayload.xml" />

25
docs/libs/gst-plugins-base-libs-sections.txt
View file

@ -1374,6 +1374,9 @@ GST_RTP_BASE_DEPAYLOAD_SRCPAD
gst_rtp_base_depayload_push
gst_rtp_base_depayload_push_list
gst_rtp_base_depayload_is_source_info_enabled
gst_rtp_base_depayload_set_source_info_enabled
<SUBSECTION Standard>
GstRTPBaseDepayloadPrivate
GST_TYPE_RTP_BASE_DEPAYLOAD
@ -1404,6 +1407,11 @@ gst_rtp_base_payload_push
gst_rtp_base_payload_push_list
gst_rtp_base_payload_set_options
gst_rtp_base_payload_set_outcaps
gst_rtp_base_payload_allocate_output_buffer
gst_rtp_base_payload_get_source_count
gst_rtp_base_payload_is_source_info_enabled
gst_rtp_base_payload_set_source_info_enabled
<SUBSECTION Standard>
GST_TYPE_RTP_BASE_PAYLOAD
GST_RTP_BASE_PAYLOAD
@ -1414,6 +1422,23 @@ GST_IS_RTP_BASE_PAYLOAD_CLASS
gst_rtp_base_payload_get_type
</SECTION>
<SECTION>
<FILE>gstrtpmeta</FILE>
<INCLUDE>gst/rtp/rtp.h</INCLUDE>
GstRTPSourceMeta
gst_buffer_add_rtp_source_meta
gst_buffer_get_rtp_source_meta
gst_rtp_source_meta_append_csrc
gst_rtp_source_meta_get_info
gst_rtp_source_meta_get_source_count
gst_rtp_source_meta_set_ssrc
GST_RTP_SOURCE_META_MAX_CSRC_COUNT
<SUBSECTION Standard>
gst_rtp_source_meta_api_get_type
GST_RTP_SOURCE_META_API_TYPE
GST_RTP_SOURCE_META_INFO
</SECTION>
<SECTION>
<FILE>gstrtcpbuffer</FILE>
<INCLUDE>gst/rtp/rtp.h</INCLUDE>

6
gst-libs/gst/rtp/Makefile.am
View file

@ -10,7 +10,8 @@ libgstrtpinclude_HEADERS = \
gstrtphdrext.h \
gstrtpbaseaudiopayload.h \
gstrtpbasepayload.h \
gstrtpbasedepayload.h
gstrtpbasedepayload.h \
gstrtpmeta.h
lib_LTLIBRARIES = libgstrtp-@GST_API_VERSION@.la
@ -20,7 +21,8 @@ libgstrtp_@GST_API_VERSION@_la_SOURCES = gstrtpbuffer.c \
gstrtphdrext.c \
gstrtpbaseaudiopayload.c \
gstrtpbasepayload.c \
gstrtpbasedepayload.c
gstrtpbasedepayload.c \
gstrtpmeta.c
built_sources = gstrtp-enumtypes.c
built_headers = gstrtp-enumtypes.h

27
gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
View file

@ -110,6 +110,7 @@ struct _GstRTPBaseAudioPayloadPrivate
guint cached_max_length;
guint cached_ptime_multiple;
guint cached_align;
guint cached_csrc_count;
gboolean buffer_list;
};
@ -453,7 +454,8 @@ gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload,
payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
@ -519,7 +521,7 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
payload_len, GST_TIME_ARGS (timestamp));
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
@ -628,7 +630,7 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
@ -653,8 +655,8 @@ gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
* mtu and min/max_ptime values. We cache those so that we don't have to redo
* all the calculations */
static gboolean
gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
basepayload, guint * min_payload_len, guint * max_payload_len,
gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload * basepayload,
guint csrc_count, guint * min_payload_len, guint * max_payload_len,
guint * align)
{
GstRTPBaseAudioPayload *payload;
@ -672,13 +674,16 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
/* check cached values */
/* check cached values. Since csrc_count may vary for each packet, we only
* check whether the new value exceeds the cached value and thus result in
* smaller payload. */
if (G_LIKELY (priv->cached_mtu == mtu
&& priv->cached_ptime_multiple ==
basepayload->ptime_multiple
&& priv->cached_ptime == basepayload->ptime
&& priv->cached_max_ptime == basepayload->max_ptime
&& priv->cached_min_ptime == basepayload->min_ptime)) {
&& priv->cached_min_ptime == basepayload->min_ptime
&& priv->cached_csrc_count >= csrc_count)) {
/* if nothing changed, return cached values */
*min_payload_len = priv->cached_min_length;
*max_payload_len = priv->cached_max_length;
@ -697,7 +702,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
maxptime_octets = G_MAXUINT;
}
/* MTU max */
max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, csrc_count);
/* round down to alignment */
max_mtu = ALIGN_DOWN (max_mtu, *align);
@ -733,6 +738,7 @@ gst_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
priv->cached_min_length = *min_payload_len;
priv->cached_max_length = *max_payload_len;
priv->cached_align = *align;
priv->cached_csrc_count = csrc_count;
return TRUE;
}
@ -875,8 +881,9 @@ gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
}
}
if (!gst_rtp_base_audio_payload_get_lengths (basepayload, &min_payload_len,
&max_payload_len, &align))
if (!gst_rtp_base_audio_payload_get_lengths (basepayload,
gst_rtp_base_payload_get_source_count (basepayload, buffer),
&min_payload_len, &max_payload_len, &align))
goto config_error;
GST_DEBUG_OBJECT (payload,

94
gst-libs/gst/rtp/gstrtpbasedepayload.c
View file

@ -30,6 +30,7 @@
#endif
#include "gstrtpbasedepayload.h"
#include "gstrtpmeta.h"
GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
@ -57,6 +58,9 @@ struct _GstRTPBaseDepayloadPrivate
GstCaps *last_caps;
GstEvent *segment_event;
guint32 segment_seqnum; /* Note: this is a GstEvent seqnum */
gboolean source_info;
GstBuffer *input_buffer;
};
/* Filter signals and args */
@ -66,10 +70,13 @@ enum
LAST_SIGNAL
};
#define DEFAULT_SOURCE_INFO FALSE
enum
{
PROP_0,
PROP_STATS,
PROP_SOURCE_INFO,
PROP_LAST
};
@ -183,6 +190,18 @@ gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * klass)
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBaseDepayload:source-info:
*
* Add RTP source information found in RTP header as meta to output buffer.
*
* Since: 1.16
**/
g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
g_param_spec_boolean ("source-info", "RTP source information",
"Add RTP source information as buffer meta",
DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
gstelement_class->change_state = gst_rtp_base_depayload_change_state;
klass->packet_lost = gst_rtp_base_depayload_packet_lost;
@ -231,6 +250,7 @@ gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
priv->dts = -1;
priv->pts = -1;
priv->duration = -1;
priv->source_info = DEFAULT_SOURCE_INFO;
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
}
@ -442,6 +462,8 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
filter->need_newsegment = FALSE;
}
priv->input_buffer = in;
if (process_rtp_packet_func != NULL) {
out_buf = process_rtp_packet_func (filter, &rtp);
gst_rtp_buffer_unmap (&rtp);
@ -458,6 +480,7 @@ gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
}
gst_buffer_unref (in);
priv->input_buffer = NULL;
return ret;
@ -727,6 +750,30 @@ create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
return event;
}
static void
add_rtp_source_meta (GstBuffer * outbuf, GstBuffer * rtpbuf)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
GstRTPSourceMeta *meta;
guint32 ssrc;
if (!gst_rtp_buffer_map (rtpbuf, GST_MAP_READ, &rtp))
return;
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
meta = gst_buffer_add_rtp_source_meta (outbuf, &ssrc, NULL, 0);
if (meta != NULL) {
gint i;
gint csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
for (i = 0; i < csrc_count; i++) {
guint32 csrc = gst_rtp_buffer_get_csrc (&rtp, i);
gst_rtp_source_meta_append_csrc (meta, &csrc, 1);
}
}
gst_rtp_buffer_unmap (&rtp);
}
static gboolean
set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
{
@ -759,6 +806,9 @@ set_headers (GstBuffer ** buffer, guint idx, GstRTPBaseDepayload * depayload)
priv->dts = GST_CLOCK_TIME_NONE;
priv->duration = GST_CLOCK_TIME_NONE;
if (priv->source_info && priv->input_buffer)
add_rtp_source_meta (*buffer, priv->input_buffer);
return TRUE;
}
@ -959,7 +1009,15 @@ static void
gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPBaseDepayload *depayload;
depayload = GST_RTP_BASE_DEPAYLOAD (object);
switch (prop_id) {
case PROP_SOURCE_INFO:
gst_rtp_base_depayload_set_source_info_enabled (depayload,
g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -979,8 +1037,44 @@ gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
g_value_take_boxed (value,
gst_rtp_base_depayload_create_stats (depayload));
break;
case PROP_SOURCE_INFO:
g_value_set_boolean (value,
gst_rtp_base_depayload_is_source_info_enabled (depayload));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/**
* gst_rtp_base_depayload_set_source_info_enabled:
* @depayload: a #GstRTPBaseDepayload
* @enable: whether to add meta about RTP sources to buffer
*
* Enable or disable adding #GstRTPSourceMeta to depayloaded buffers.
*
* Since: 1.16
**/
void
gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
gboolean enable)
{
depayload->priv->source_info = enable;
}
/**
* gst_rtp_base_depayload_is_source_info_enabled:
* @depayload: a #GstRTPBaseDepayload
*
* Queries whether #GstRTPSourceMeta will be added to depayloaded buffers.
*
* Returns: %TRUE if source-info is enabled.
*
* Since: 1.16
**/
gboolean
gst_rtp_base_depayload_is_source_info_enabled (GstRTPBaseDepayload * depayload)
{
return depayload->priv->source_info;
}

7
gst-libs/gst/rtp/gstrtpbasedepayload.h
View file

@ -120,6 +120,13 @@ GstFlowReturn gst_rtp_base_depayload_push (GstRTPBaseDepayload *filter,
GST_RTP_API
GstFlowReturn gst_rtp_base_depayload_push_list (GstRTPBaseDepayload *filter, GstBufferList *out_list);
GST_RTP_API
gboolean gst_rtp_base_depayload_is_source_info_enabled (GstRTPBaseDepayload * depayload);
GST_RTP_API
void gst_rtp_base_depayload_set_source_info_enabled (GstRTPBaseDepayload * depayload,
gboolean enable);
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseDepayload, gst_object_unref)

186
gst-libs/gst/rtp/gstrtpbasepayload.c
View file

@ -29,6 +29,7 @@
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpbasepayload.h"
#include "gstrtpmeta.h"
GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug);
#define GST_CAT_DEFAULT (rtpbasepayload_debug)
@ -44,6 +45,9 @@ struct _GstRTPBasePayloadPrivate
gboolean pt_set;
gboolean source_info;
GstBuffer *input_meta_buffer;
guint64 base_offset;
gint64 base_rtime;
guint64 base_rtime_hz;
@ -84,6 +88,7 @@ enum
#define DEFAULT_PERFECT_RTPTIME TRUE
#define DEFAULT_PTIME_MULTIPLE 0
#define DEFAULT_RUNNING_TIME GST_CLOCK_TIME_NONE
#define DEFAULT_SOURCE_INFO FALSE
enum
{
@ -100,6 +105,7 @@ enum
PROP_PERFECT_RTPTIME,
PROP_PTIME_MULTIPLE,
PROP_STATS,
PROP_SOURCE_INFO,
PROP_LAST
};
@ -299,6 +305,19 @@ gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:source-info:
*
* Enable writing the CSRC field in allocated RTP header based on RTP source
* information found in the input buffer's #GstRTPSourceMeta.
*
* Since: 1.16
**/
g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
g_param_spec_boolean ("source-info", "RTP source information",
"Write CSRC based on buffer meta RTP source information",
DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
gstelement_class->change_state = gst_rtp_base_payload_change_state;
klass->get_caps = gst_rtp_base_payload_getcaps_default;
@ -351,6 +370,7 @@ gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class)
priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
priv->ssrc_random = (rtpbasepayload->ssrc == -1);
priv->pt_set = FALSE;
priv->source_info = DEFAULT_SOURCE_INFO;
rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
@ -633,6 +653,15 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
if (!rtpbasepayload->priv->negotiated)
goto not_negotiated;
if (rtpbasepayload->priv->source_info) {
/* Save a copy of meta (instead of taking an extra reference before
* handle_buffer) to make the meta available when allocating a output
* buffer. */
rtpbasepayload->priv->input_meta_buffer = gst_buffer_new ();
gst_buffer_copy_into (rtpbasepayload->priv->input_meta_buffer, buffer,
GST_BUFFER_COPY_META, 0, -1);
}
if (gst_pad_check_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
if (!gst_rtp_base_payload_negotiate (rtpbasepayload)) {
gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload));
@ -646,6 +675,8 @@ gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
gst_buffer_replace (&rtpbasepayload->priv->input_meta_buffer, NULL);
return ret;
/* ERRORS */
@ -1156,6 +1187,25 @@ map_failed:
}
}
static gboolean
foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
{
GType drop_api_type = (GType) GPOINTER_TO_INT (user_data);
const GstMetaInfo *info = (*meta)->info;
if (info->api == drop_api_type)
*meta = NULL;
return TRUE;
}
static gboolean
filter_meta (GstBuffer ** buffer, guint idx, gpointer user_data)
{
return gst_buffer_foreach_meta (*buffer, foreach_metadata_drop,
GINT_TO_POINTER (GST_RTP_SOURCE_META_API_TYPE));
}
/* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
* before the buffer is pushed. */
static GstFlowReturn
@ -1252,11 +1302,14 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
}
/* set ssrc, payload type, seq number, caps and rtptime */
/* remove unwanted meta */
if (is_list) {
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), set_headers, &data);
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), filter_meta, NULL);
} else {
GstBuffer *buf = GST_BUFFER_CAST (obj);
set_headers (&buf, 0, &data);
filter_meta (&buf, 0, NULL);
}
priv->next_seqnum = data.seqnum;
@ -1267,8 +1320,8 @@ gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
(is_list) ? -1 : gst_buffer_get_size (GST_BUFFER (obj)),
payload->seqnum, data.rtptime, GST_TIME_ARGS (data.pts));
if (g_atomic_int_compare_and_exchange (&payload->
priv->notified_first_timestamp, 1, 0)) {
if (g_atomic_int_compare_and_exchange (&payload->priv->
notified_first_timestamp, 1, 0)) {
g_object_notify (G_OBJECT (payload), "timestamp");
g_object_notify (G_OBJECT (payload), "seqnum");
}
@ -1351,6 +1404,62 @@ gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
return res;
}
/**
* gst_rtp_base_payload_allocate_output_buffer:
* @payload: a #GstRTPBasePayload
* @payload_len: the length of the payload
* @pad_len: the amount of padding
* @csrc_count: the minimum number of CSRC entries
*
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
* minimum @csrc_count CSRCs, a payload length of @payload_len and padding of
* @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional
* CSRCs may be allocated and filled with RTP source information.
*
* Returns: A newly allocated buffer that can hold an RTP packet with given
* parameters.
*
* Since: 1.16
*/
GstBuffer *
gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
guint payload_len, guint8 pad_len, guint8 csrc_count)
{
GstBuffer *buffer = NULL;
if (payload->priv->input_meta_buffer != NULL) {
GstRTPSourceMeta *meta =
gst_buffer_get_rtp_source_meta (payload->priv->input_meta_buffer);
if (meta != NULL) {
guint total_csrc_count, idx, i;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
total_csrc_count = csrc_count + meta->csrc_count +
(meta->ssrc_valid ? 1 : 0);
total_csrc_count = MIN (total_csrc_count, 15);
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len,
total_csrc_count);
gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp);
/* Skip CSRC fields requested by derived class and fill CSRCs from meta.
* Finally append the SSRC as a new CSRC. */
idx = csrc_count;
for (i = 0; i < meta->csrc_count && idx < 15; i++, idx++)
gst_rtp_buffer_set_csrc (&rtp, idx, meta->csrc[i]);
if (meta->ssrc_valid && idx < 15)
gst_rtp_buffer_set_csrc (&rtp, idx, meta->ssrc);
gst_rtp_buffer_unmap (&rtp);
}
}
if (buffer == NULL)
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len, csrc_count);
return buffer;
}
static GstStructure *
gst_rtp_base_payload_create_stats (GstRTPBasePayload * rtpbasepayload)
{
@ -1421,6 +1530,10 @@ gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
case PROP_PTIME_MULTIPLE:
rtpbasepayload->ptime_multiple = g_value_get_int64 (value);
break;
case PROP_SOURCE_INFO:
gst_rtp_base_payload_set_source_info_enabled (rtpbasepayload,
g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -1484,6 +1597,10 @@ gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
g_value_take_boxed (value,
gst_rtp_base_payload_create_stats (rtpbasepayload));
break;
case PROP_SOURCE_INFO:
g_value_set_boolean (value,
gst_rtp_base_payload_is_source_info_enabled (rtpbasepayload));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -1551,3 +1668,68 @@ gst_rtp_base_payload_change_state (GstElement * element,
}
return ret;
}
/**
* gst_rtp_base_payload_set_source_info_enabled:
* @payload: a #GstRTPBasePayload
* @enable: whether to add contributing sources to RTP packets
*
* Enable or disable adding contributing sources to RTP packets from
* #GstRTPSourceMeta.
*
* Since: 1.16
**/
void
gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
gboolean enable)
{
payload->priv->source_info = enable;
}
/**
* gst_rtp_base_payload_is_source_info_enabled:
* @payload: a #GstRTPBasePayload
*
* Queries whether the payloader will add contributing sources (CSRCs) to the
* RTP header from #GstRTPSourceMeta.
*
* Returns: %TRUE if source-info is enabled.
*
* Since: 1.16
**/
gboolean
gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)
{
return payload->priv->source_info;
}
/**
* gst_rtp_base_payload_get_source_count:
* @payload: a #GstRTPBasePayload
* @buffer: (transfer none): a #GstBuffer, typically the buffer to payload
*
* Count the total number of RTP sources found in the meta of @buffer, which
* will be automically added by gst_rtp_base_payload_allocate_output_buffer().
* If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
*
* Returns: The number of sources.
*
* Since: 1.16
**/
guint
gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
GstBuffer * buffer)
{
guint count = 0;
g_return_val_if_fail (buffer != NULL, 0);
if (gst_rtp_base_payload_is_source_info_enabled (payload)) {
GstRTPSourceMeta *meta = gst_buffer_get_rtp_source_meta (buffer);
if (meta != NULL)
count = gst_rtp_source_meta_get_source_count (meta);
}
return count;
}

16
gst-libs/gst/rtp/gstrtpbasepayload.h
View file

@ -172,6 +172,22 @@ GST_RTP_API
GstFlowReturn gst_rtp_base_payload_push_list (GstRTPBasePayload *payload,
GstBufferList *list);
GST_RTP_API
GstBuffer * gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
guint payload_len, guint8 pad_len,
guint8 csrc_count);
GST_RTP_API
void gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
gboolean enable);
GST_RTP_API
gboolean gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload);
GST_RTP_API
guint gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
GstBuffer * buffer);
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBasePayload, gst_object_unref)
#endif

229
gst-libs/gst/rtp/gstrtpmeta.c Normal file
View file

@ -0,0 +1,229 @@
/* GStreamer
* Copyright (C) <2016> Stian Selnes <stian@pexip.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpmeta.h"
#include <string.h>
/**
* SECTION:gstrtpmeta
* @title: GstMeta for RTP
* @short_description: RTP related GstMeta
*
*/
/**
* gst_buffer_add_rtp_source_meta:
* @buffer: a #GstBuffer
* @ssrc: (allow-none) (transfer none): pointer to the SSRC
* @csrc: (allow-none) (transfer none): pointer to the CSRCs
* @csrc_count: number of elements in @csrc
*
* Attaches RTP source information to @buffer.
*
* Returns: (transfer none): the #GstRTPSourceMeta on @buffer.
*
* Since: 1.16
*/
GstRTPSourceMeta *
gst_buffer_add_rtp_source_meta (GstBuffer * buffer, const guint32 * ssrc,
const guint * csrc, guint csrc_count)
{
gint i;
GstRTPSourceMeta *meta;
g_return_val_if_fail (buffer != NULL, NULL);
g_return_val_if_fail (csrc_count <= GST_RTP_SOURCE_META_MAX_CSRC_COUNT, NULL);
g_return_val_if_fail (csrc_count == 0 || csrc != NULL, NULL);
meta = (GstRTPSourceMeta *) gst_buffer_add_meta (buffer,
GST_RTP_SOURCE_META_INFO, NULL);
if (!meta)
return NULL;
if (ssrc != NULL) {
meta->ssrc = *ssrc;
meta->ssrc_valid = TRUE;
} else {
meta->ssrc_valid = FALSE;
}
meta->csrc_count = csrc_count;
for (i = 0; i < csrc_count; i++) {
meta->csrc[i] = csrc[i];
}
return meta;
}
/**
* gst_buffer_get_rtp_source_meta:
* @buffer: a #GstBuffer
*
* Find the #GstRTPSourceMeta on @buffer.
*
* Returns: (transfer none): the #GstRTPSourceMeta or %NULL when there
* is no such metadata on @buffer.
*
* Since: 1.16
*/
GstRTPSourceMeta *
gst_buffer_get_rtp_source_meta (GstBuffer * buffer)
{
return (GstRTPSourceMeta *) gst_buffer_get_meta (buffer,
gst_rtp_source_meta_api_get_type ());
}
static gboolean
gst_rtp_source_meta_transform (GstBuffer * dst, GstMeta * meta,
GstBuffer * src, GQuark type, gpointer data)
{
if (GST_META_TRANSFORM_IS_COPY (type)) {
GstRTPSourceMeta *smeta = (GstRTPSourceMeta *) meta;
GstRTPSourceMeta *dmeta;
guint32 *ssrc = smeta->ssrc_valid ? &smeta->ssrc : NULL;
dmeta = gst_buffer_add_rtp_source_meta (dst, ssrc, smeta->csrc,
smeta->csrc_count);
if (dmeta == NULL)
return FALSE;
} else {
/* return FALSE, if transform type is not supported */
return FALSE;
}
return TRUE;
}
/**
* gst_rtp_source_meta_get_source_count:
* @meta: a #GstRTPSourceMeta
*
* Count the total number of RTP sources found in @meta, both SSRC and CSRC.
*
* Returns: The number of RTP sources
*
* Since: 1.16
*/
guint
gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta)
{
/* Never return more than a count of 15 so that the returned value
* conveniently can be used as argument 'csrc_count' in
* gst_rtp_buffer-functions. */
guint ssrc_count = meta->ssrc_valid ? 1 : 0;
return MIN (meta->csrc_count + ssrc_count, 15);
}
/**
* gst_rtp_source_meta_set_ssrc:
* @meta: a #GstRTPSourceMeta
* @ssrc: (allow-none) (transfer none): pointer to the SSRC
*
* Sets @ssrc in @meta. If @ssrc is %NULL the ssrc of @meta will be unset.
*
* Returns: %TRUE on success, %FALSE otherwise.
*
* Since: 1.16
**/
gboolean
gst_rtp_source_meta_set_ssrc (GstRTPSourceMeta * meta, guint32 * ssrc)
{
if (ssrc != NULL) {
meta->ssrc = *ssrc;
meta->ssrc_valid = TRUE;
} else {
meta->ssrc_valid = FALSE;
}
return TRUE;
}
/**
* gst_rtp_source_meta_append_csrc:
* @meta: a #GstRTPSourceMeta
* @csrc: the csrcs to append
* @csrc_count: number of elements in @csrc
*
* Appends @csrc to the list of contributing sources in @meta.
*
* Returns: %TRUE if all elements in @csrc was added, %FALSE otherwise.
*
* Since: 1.16
**/
gboolean
gst_rtp_source_meta_append_csrc (GstRTPSourceMeta * meta, const guint32 * csrc,
guint csrc_count)
{
gint i;
guint new_csrc_count = meta->csrc_count + csrc_count;
if (new_csrc_count > GST_RTP_SOURCE_META_MAX_CSRC_COUNT)
return FALSE;
for (i = 0; i < csrc_count; i++)
meta->csrc[meta->csrc_count + i] = csrc[i];
meta->csrc_count = new_csrc_count;
return TRUE;
}
GType
gst_rtp_source_meta_api_get_type (void)
{
static volatile GType type = 0;
static const gchar *tags[] = { NULL };
if (g_once_init_enter (&type)) {
GType _type = gst_meta_api_type_register ("GstRTPSourceMetaAPI", tags);
g_once_init_leave (&type, _type);
}
return type;
}
static gboolean
gst_rtp_source_meta_init (GstMeta * meta, gpointer params, GstBuffer * buffer)
{
GstRTPSourceMeta *dmeta = (GstRTPSourceMeta *) meta;
dmeta->ssrc_valid = FALSE;
dmeta->csrc_count = 0;
return TRUE;
}
const GstMetaInfo *
gst_rtp_source_meta_get_info (void)
{
static const GstMetaInfo *rtp_source_meta_info = NULL;
if (g_once_init_enter (&rtp_source_meta_info)) {
const GstMetaInfo *meta = gst_meta_register (GST_RTP_SOURCE_META_API_TYPE,
"GstRTPSourceMeta",
sizeof (GstRTPSourceMeta),
gst_rtp_source_meta_init,
(GstMetaFreeFunction) NULL,
gst_rtp_source_meta_transform);
g_once_init_leave (&rtp_source_meta_info, meta);
}
return rtp_source_meta_info;
}

79
gst-libs/gst/rtp/gstrtpmeta.h Normal file
View file

@ -0,0 +1,79 @@
/* GStreamer
* Copyright (C) <2016> Stian Selnes <stian@pexip.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_RTP_META_H__
#define __GST_RTP_META_H__
#include <gst/gst.h>
#include <gst/rtp/rtp-prelude.h>
G_BEGIN_DECLS
#define GST_RTP_SOURCE_META_API_TYPE (gst_rtp_source_meta_api_get_type())
#define GST_RTP_SOURCE_META_INFO (gst_rtp_source_meta_get_info())
typedef struct _GstRTPSourceMeta GstRTPSourceMeta;
#define GST_RTP_SOURCE_META_MAX_CSRC_COUNT 15
/**
* GstRTPSourceMeta:
* @meta: parent #GstMeta
* @ssrc: the SSRC
* @ssrc_valid: whether @ssrc is set and valid
* @csrc: (allow-none): pointer to the CSRCs
* @csrc_count: number of elements in @csrc
*
* Meta describing the source(s) of the buffer.
*
* Since: 1.16
*/
struct _GstRTPSourceMeta
{
GstMeta meta;
guint32 ssrc;
gboolean ssrc_valid;
guint32 csrc[GST_RTP_SOURCE_META_MAX_CSRC_COUNT];
guint csrc_count;
};
GST_RTP_API
GType gst_rtp_source_meta_api_get_type (void);
GST_RTP_API
GstRTPSourceMeta * gst_buffer_add_rtp_source_meta (GstBuffer * buf, const guint32 * ssrc,
const guint32 * csrc, guint csrc_count);
GST_RTP_API
GstRTPSourceMeta * gst_buffer_get_rtp_source_meta (GstBuffer * buf);
GST_RTP_API
guint gst_rtp_source_meta_get_source_count (const GstRTPSourceMeta * meta);
GST_RTP_API
gboolean gst_rtp_source_meta_set_ssrc (GstRTPSourceMeta * meta, guint32 * ssrc);
GST_RTP_API
gboolean gst_rtp_source_meta_append_csrc (GstRTPSourceMeta * meta,
const guint32 * csrc, guint csrc_count);
GST_RTP_API
const GstMetaInfo * gst_rtp_source_meta_get_info (void);
G_END_DECLS
#endif /* __GST_RTP_META_H__ */

2
gst-libs/gst/rtp/meson.build
View file

@ -3,6 +3,7 @@ rtp_sources = [
'gstrtcpbuffer.c',
'gstrtppayloads.c',
'gstrtphdrext.c',
'gstrtpmeta.c',
'gstrtpbaseaudiopayload.c',
'gstrtpbasepayload.c',
'gstrtpbasedepayload.c'
@ -16,6 +17,7 @@ rtp_headers = [
'gstrtpbuffer.h',
'gstrtpdefs.h',
'gstrtphdrext.h',
'gstrtpmeta.h',
'gstrtppayloads.h',
'rtp-prelude.h',
'rtp.h',

1
gst-libs/gst/rtp/rtp.h
View file

@ -30,6 +30,7 @@
#include <gst/rtp/gstrtpbaseaudiopayload.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include <gst/rtp/gstrtpmeta.h>
#include <gst/rtp/gstrtp-enumtypes.h>
#endif /* __GST_RTP_H__ */

8
tests/check/Makefile.am
View file

@ -245,6 +245,7 @@ check_PROGRAMS = \
libs/rtp \
libs/rtpbasedepayload \
libs/rtpbasepayload \
libs/rtpmeta \
libs/rtsp \
libs/rtspconnection \
libs/sdp \
@ -581,6 +582,13 @@ libs_rtpbasedepayload_LDADD = \
$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
$(GST_BASE_LIBS) $(LDADD)
libs_rtpmeta_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(AM_CFLAGS)
libs_rtpmeta_LDADD = \
$(top_builddir)/gst-libs/gst/rtp/libgstrtp-@GST_API_VERSION@.la \
$(GST_BASE_LIBS) $(LDADD)
libs_rtsp_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(AM_CFLAGS)

1
tests/check/libs/.gitignore vendored
View file

@ -29,6 +29,7 @@ profile
rtp
rtpbasedepayload
rtpbasepayload
rtpmeta
rtsp
rtspconnection
sdp

114
tests/check/libs/rtpbasedepayload.c
View file

@ -23,8 +23,8 @@
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include <gst/check/gstharness.h>
#include <gst/rtp/rtp.h>
#define DEFAULT_CLOCK_RATE (42)
@ -317,22 +317,14 @@ validate_event (guint index, const gchar * name, const gchar * field, ...)
va_end (var_args);
}
#define push_rtp_buffer(state, field, ...) \
push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
#define push_rtp_buffer_fails(state, error, field, ...) \
push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
static void
push_rtp_buffer_full (State * state, GstFlowReturn expected,
const gchar * field, ...)
rtp_buffer_set_valist (GstBuffer * buf, const gchar * field, va_list var_args,
gboolean * extra_ref_)
{
GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
GstRTPBuffer rtp = { NULL };
gboolean mapped = FALSE;
gboolean extra_ref = FALSE;
va_list var_args;
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime pts = va_arg (var_args, GstClockTime);
@ -366,13 +358,18 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
gst_rtp_buffer_set_ssrc (&rtp, ssrc);
} else if (!g_strcmp0 (field, "extra-ref")) {
extra_ref = va_arg (var_args, gboolean);
if (extra_ref_)
*extra_ref_ = extra_ref;
} else if (!g_strcmp0 (field, "csrc")) {
guint idx = va_arg (var_args, guint);
guint csrc = va_arg (var_args, guint);
gst_rtp_buffer_set_csrc (&rtp, idx, csrc);
} else {
fail ("test cannot set unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
if (mapped) {
gst_rtp_buffer_unmap (&rtp);
@ -380,6 +377,34 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
if (extra_ref)
gst_buffer_ref (buf);
}
static void
rtp_buffer_set (GstBuffer * buf, const gchar * field, ...)
{
va_list var_args;
va_start (var_args, field);
rtp_buffer_set_valist (buf, field, var_args, NULL);
va_end (var_args);
}
#define push_rtp_buffer(state, field, ...) \
push_rtp_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
#define push_rtp_buffer_fails(state, error, field, ...) \
push_rtp_buffer_full ((state), (error), (field), __VA_ARGS__)
static void
push_rtp_buffer_full (State * state, GstFlowReturn expected,
const gchar * field, ...)
{
GstBuffer *buf = gst_rtp_buffer_new_allocate (0, 0, 0);
va_list var_args;
gboolean extra_ref = FALSE;
va_start (var_args, field);
rtp_buffer_set_valist (buf, field, var_args, &extra_ref);
va_end (var_args);
fail_unless_equals_int (gst_pad_push (state->srcpad, buf), expected);
@ -388,7 +413,7 @@ push_rtp_buffer_full (State * state, GstFlowReturn expected,
}
#define push_buffer(state, field, ...) \
push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
push_buffer_full ((state), GST_FLOW_OK, (field), __VA_ARGS__)
static void
push_buffer_full (State * state, GstFlowReturn expected,
@ -1237,7 +1262,64 @@ GST_START_TEST (rtp_base_depayload_clock_base_test)
destroy_depayloader (state);
}
GST_END_TEST static Suite *
GST_END_TEST
/* basedepayloader has a property source-info that will add
* GstRTPSourceMeta to the output buffer with RTP source information, such as
* SSRC and CSRCs. The is useful for letting downstream know about the origin
* of the stream. */
GST_START_TEST (rtp_base_depayload_source_info_test)
{
GstHarness *h;
GstRtpDummyDepay *depay;
GstBuffer *buffer;
GstRTPSourceMeta *meta;
guint seq = 0;
depay = rtp_dummy_depay_new ();
h = gst_harness_new_with_element (GST_ELEMENT_CAST (depay), "sink", "src");
gst_harness_set_src_caps_str (h, "application/x-rtp");
/* Property enabled should always add meta, also when there is only SSRC and
* no CSRC. */
g_object_set (depay, "source-info", TRUE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
buffer = gst_harness_push_and_pull (h, buffer);
fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
fail_unless (meta->ssrc_valid);
fail_unless_equals_int (meta->ssrc, 0x11);
fail_unless_equals_int (meta->csrc_count, 0);
gst_buffer_unref (buffer);
/* Both SSRC and CSRC should be added to the meta */
buffer = gst_rtp_buffer_new_allocate (0, 0, 2);
rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, "csrc", 0, 0x22,
"csrc", 1, 0x33, NULL);
buffer = gst_harness_push_and_pull (h, buffer);
fail_unless ((meta = gst_buffer_get_rtp_source_meta (buffer)));
fail_unless (meta->ssrc_valid);
fail_unless_equals_int (meta->ssrc, 0x11);
fail_unless_equals_int (meta->csrc_count, 2);
fail_unless_equals_int (meta->csrc[0], 0x22);
fail_unless_equals_int (meta->csrc[1], 0x33);
gst_buffer_unref (buffer);
/* Property disabled should never add meta */
g_object_set (depay, "source-info", FALSE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
rtp_buffer_set (buffer, "seq", seq++, "ssrc", 0x11, NULL);
buffer = gst_harness_push_and_pull (h, buffer);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
g_object_unref (depay);
gst_harness_teardown (h);
}
GST_END_TEST;
static Suite *
rtp_basepayloading_suite (void)
{
Suite *s = suite_create ("rtp_base_depayloading_test");
@ -1265,6 +1347,8 @@ rtp_basepayloading_suite (void)
tcase_add_test (tc_chain, rtp_base_depayload_play_speed_test);
tcase_add_test (tc_chain, rtp_base_depayload_clock_base_test);
tcase_add_test (tc_chain, rtp_base_depayload_source_info_test);
return s;
}

109
tests/check/libs/rtpbasepayload.c
View file

@ -23,8 +23,8 @@
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/check/gstharness.h>
#include <gst/rtp/rtp.h>
#define DEFAULT_CLOCK_RATE (42)
#define BUFFER_BEFORE_LIST (10)
@ -121,7 +121,9 @@ gst_rtp_dummy_pay_handle_buffer (GstRTPBasePayload * pay, GstBuffer * buffer)
}
}
paybuffer = gst_rtp_buffer_new_allocate (0, 0, 0);
paybuffer =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (pay),
0, 0, 0);
GST_BUFFER_PTS (paybuffer) = GST_BUFFER_PTS (buffer);
GST_BUFFER_OFFSET (paybuffer) = GST_BUFFER_OFFSET (buffer);
@ -444,20 +446,11 @@ validate_buffers_received (guint received_buffers)
}
static void
validate_buffer (guint index, const gchar * field, ...)
validate_buffer_valist (GstBuffer * buf, const gchar * field, va_list var_args)
{
GstBuffer *buf;
GstRTPBuffer rtp = { NULL };
gboolean mapped = FALSE;
va_list var_args;
fail_if (index >= g_list_length (buffers));
buf = GST_BUFFER (g_list_nth_data (buffers, index));
fail_if (buf == NULL);
GST_TRACE ("%" GST_PTR_FORMAT, buf);
va_start (var_args, field);
while (field) {
if (!g_strcmp0 (field, "pts")) {
GstClockTime pts = va_arg (var_args, GstClockTime);
@ -489,19 +482,53 @@ validate_buffer (guint index, const gchar * field, ...)
} else if (!g_strcmp0 (field, "ssrc")) {
guint32 ssrc = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_ssrc (&rtp), ssrc);
} else if (!g_strcmp0 (field, "csrc")) {
guint idx = va_arg (var_args, guint);
guint csrc = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_csrc (&rtp, idx), csrc);
} else if (!g_strcmp0 (field, "csrc-count")) {
guint csrc_count = va_arg (var_args, guint);
fail_unless_equals_int (gst_rtp_buffer_get_csrc_count (&rtp),
csrc_count);
} else {
fail ("test cannot validate unknown buffer field '%s'", field);
}
}
field = va_arg (var_args, const gchar *);
}
va_end (var_args);
if (mapped) {
gst_rtp_buffer_unmap (&rtp);
}
}
static void
validate_buffer1 (GstBuffer * buf, const gchar * field, ...)
{
va_list var_args;
va_start (var_args, field);
validate_buffer_valist (buf, field, var_args);
va_end (var_args);
}
static void
validate_buffer (guint index, const gchar * field, ...)
{
GstBuffer *buf;
va_list var_args;
fail_if (index >= g_list_length (buffers));
buf = GST_BUFFER (g_list_nth_data (buffers, index));
fail_if (buf == NULL);
GST_TRACE ("%" GST_PTR_FORMAT, buf);
va_start (var_args, field);
validate_buffer_valist (buf, field, var_args);
va_end (var_args);
}
static void
get_buffer_field (guint index, const gchar * field, ...)
{
@ -1775,6 +1802,59 @@ GST_START_TEST (rtp_base_payload_property_stats_test)
GST_END_TEST;
/* basepayloader has a property source-info that makes it aware of RTP
* source information passed as GstRTPSourceMeta on the input buffers. All
* sources found in the meta will be added to the list of CSRCs in the RTP
* header. A useful scenario for this is, for instance, to signal which
* sources contributed to a mixed audio stream. */
GST_START_TEST (rtp_base_payload_property_source_info_test)
{
GstHarness *h;
GstRtpDummyPay *pay;
GstBuffer *buffer;
guint csrc_count = 2;
const guint32 csrc[] = { 0x11, 0x22 };
const guint32 ssrc = 0x33;
pay = rtp_dummy_pay_new ();
h = gst_harness_new_with_element (GST_ELEMENT_CAST (pay), "sink", "src");
gst_harness_set_src_caps_str (h, "application/x-rtp");
/* Input buffer has no meta, payloader should not add CSRC */
g_object_set (pay, "source-info", TRUE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
buffer = gst_harness_push_and_pull (h, buffer);
validate_buffer1 (buffer, "csrc-count", 0, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
/* Input buffer has meta, payloader should add CSRC */
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
fail_unless (gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc,
csrc_count));
buffer = gst_harness_push_and_pull (h, buffer);
/* The meta SSRC should be added as the last contributing source */
validate_buffer1 (buffer, "csrc-count", 3, "csrc", 0, csrc[0],
"csrc", 1, csrc[1], "csrc", 2, ssrc, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
/* When property is disabled, the meta should be ignored and no CSRC
* added. */
g_object_set (pay, "source-info", FALSE, NULL);
buffer = gst_rtp_buffer_new_allocate (0, 0, 0);
fail_unless (gst_buffer_add_rtp_source_meta (buffer, NULL, csrc, csrc_count));
buffer = gst_harness_push_and_pull (h, buffer);
validate_buffer1 (buffer, "csrc-count", 0, NULL);
fail_if (gst_buffer_get_rtp_source_meta (buffer));
gst_buffer_unref (buffer);
g_object_unref (pay);
gst_harness_teardown (h);
}
GST_END_TEST;
/* push a single buffer to the payloader which should successfully payload it
* into an RTP packet. besides the payloaded RTP packet there should be the
* three events initial events: stream-start, caps and segment. because of that
@ -1878,6 +1958,7 @@ rtp_basepayloading_suite (void)
tcase_add_test (tc_chain, rtp_base_payload_property_perfect_rtptime_test);
tcase_add_test (tc_chain, rtp_base_payload_property_ptime_multiple_test);
tcase_add_test (tc_chain, rtp_base_payload_property_stats_test);
tcase_add_test (tc_chain, rtp_base_payload_property_source_info_test);
tcase_add_test (tc_chain, rtp_base_payload_framerate_attribute);
tcase_add_test (tc_chain, rtp_base_payload_max_framerate_attribute);

110
tests/check/libs/rtpmeta.c Normal file
View file

@ -0,0 +1,110 @@
/* GStreamer RTP meta unit tests
* Copyright (C) 2016 Stian Selnes <stian@pexip.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/rtp/rtp.h>
GST_START_TEST (test_rtp_source_meta_set_get_sources)
{
GstBuffer *buffer;
GstRTPSourceMeta *meta;
guint32 ssrc = 1000, ssrc2 = 2000;
const guint32 csrc[] = {
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14
};
buffer = gst_buffer_new ();
meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 12);
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
fail_unless (meta->ssrc_valid);
fail_unless_equals_int (meta->ssrc, ssrc);
for (gint i = 0; i < 12; i++)
fail_unless_equals_int (meta->csrc[i], csrc[i]);
/* Unset the ssrc */
fail_unless (gst_rtp_source_meta_set_ssrc (meta, NULL));
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12);
fail_if (meta->ssrc_valid);
/* Set the ssrc again */
fail_unless (gst_rtp_source_meta_set_ssrc (meta, &ssrc2));
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
fail_unless (meta->ssrc_valid);
fail_unless_equals_int (meta->ssrc, ssrc2);
/* Append multiple csrcs */
fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[12], 2));
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
for (gint i = 0; i < 14; i++)
fail_unless_equals_int (meta->csrc[i], csrc[i]);
gst_buffer_unref (buffer);
}
GST_END_TEST;
GST_START_TEST (test_rtp_source_meta_set_get_max_sources)
{
GstBuffer *buffer;
GstRTPSourceMeta *meta;
guint32 ssrc = 1000;
const guint32 csrc[16] = { 0, };
buffer = gst_buffer_new ();
meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 14);
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
fail_unless_equals_int (meta->csrc_count, 14);
fail_unless (meta->ssrc_valid);
fail_unless_equals_int (meta->ssrc, ssrc);
/* Append one more csrc */
/* The source count should cap at 15 for convenient use with
* gst_rtp_buffer-functions! */
fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[14], 1));
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
fail_unless_equals_int (meta->csrc_count, 15);
/* Try to append one more csrc, but we've reached max */
fail_if (gst_rtp_source_meta_append_csrc (meta, &csrc[15], 1));
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
fail_unless_equals_int (meta->csrc_count, 15);
gst_buffer_unref (buffer);
}
GST_END_TEST;
static Suite *
rtp_meta_suite (void)
{
Suite *s = suite_create ("rtp_meta_tests");
TCase *tc_chain;
suite_add_tcase (s, (tc_chain = tcase_create ("GstRTPSourceMeta")));
tcase_add_test (tc_chain, test_rtp_source_meta_set_get_sources);
tcase_add_test (tc_chain, test_rtp_source_meta_set_get_max_sources);
return s;
}
GST_CHECK_MAIN (rtp_meta)