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Revert "srtpdec: Add support for buffer list"

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This reverts commit ff11a1a8a0.

It can't be assumed that all buffers in a buffer list have the same SSRC or
are RTP or RTCP only. It has to be checked for every single buffer, and one
basically has to do the processing that is done by the default chain_list
implementation.
This commit is contained in:
Sebastian Dröge 2015-08-13 12:38:41 +02:00
parent 397b5d06ec
commit 88f85f6595
1 changed files with 1 additions and 141 deletions
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142
ext/srtp/gstsrtpdec.c
View file

@ -145,22 +145,6 @@ enum
PROP_REPLAY_WINDOW_SIZE
};
typedef struct _ValidateBufferItData
{
GstSrtpDecSsrcStream **stream;
GstSrtpDec *filter;
guint32 *ssrc;
gboolean *is_rtcp;
} ValidateBufferItData;
typedef struct _DecodeBufferItData
{
GstSrtpDec *filter;
GstPad *pad;
guint32 ssrc;
gboolean is_rtcp;
} DecodeBufferItData;
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
@ -226,11 +210,6 @@ static GstFlowReturn gst_srtp_dec_chain_rtp (GstPad * pad,
static GstFlowReturn gst_srtp_dec_chain_rtcp (GstPad * pad,
GstObject * parent, GstBuffer * buf);
static GstFlowReturn gst_srtp_dec_chain_list_rtp (GstPad * pad,
GstObject * parent, GstBufferList * buf_list);
static GstFlowReturn gst_srtp_dec_chain_list_rtcp (GstPad * pad,
GstObject * parent, GstBufferList * buf_list);
static GstStateChangeReturn gst_srtp_dec_change_state (GstElement * element,
GstStateChange transition);
@ -390,8 +369,6 @@ gst_srtp_dec_init (GstSrtpDec * filter)
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp));
gst_pad_set_chain_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtp));
gst_pad_set_chain_list_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_list_rtp));
filter->rtp_srcpad =
gst_pad_new_from_static_template (&rtp_src_template, "rtp_src");
@ -415,8 +392,6 @@ gst_srtp_dec_init (GstSrtpDec * filter)
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp));
gst_pad_set_chain_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtcp));
gst_pad_set_chain_list_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_list_rtcp));
filter->rtcp_srcpad =
gst_pad_new_from_static_template (&rtcp_src_template, "rtcp_src");
@ -1228,6 +1203,7 @@ gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
GST_OBJECT_LOCK (filter);
/* Check if this stream exists, if not create a new stream */
if (!(stream = validate_buffer (filter, buf, &ssrc, &is_rtcp))) {
GST_OBJECT_UNLOCK (filter);
GST_WARNING_OBJECT (filter, "Invalid buffer, dropping");
@ -1275,107 +1251,6 @@ drop_buffer:
return ret;
}
static gboolean
validate_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
{
ValidateBufferItData *data = user_data;
if (!(*data->stream =
validate_buffer (data->filter, *buffer, data->ssrc, data->is_rtcp))) {
GST_WARNING_OBJECT (data->filter, "Invalid buffer, dropping");
gst_buffer_replace (buffer, NULL);
}
return TRUE;
}
static gboolean
decode_buffer_it (GstBuffer ** buffer, guint index, gpointer user_data)
{
DecodeBufferItData *data = user_data;
if (!gst_srtp_dec_decode_buffer (data->filter, data->pad, *buffer,
data->is_rtcp, data->ssrc)) {
GST_WARNING_OBJECT (data->filter, "Error decoding buffer, dropping");
gst_buffer_replace (buffer, NULL);
}
return TRUE;
}
static GstFlowReturn
gst_srtp_dec_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * buf_list, gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstSrtpDecSsrcStream *stream = NULL;
GstFlowReturn ret = GST_FLOW_OK;
guint32 ssrc = 0;
ValidateBufferItData validate_data;
DecodeBufferItData decode_data;
validate_data.stream = &stream;
validate_data.filter = filter;
validate_data.ssrc = &ssrc;
validate_data.is_rtcp = &is_rtcp;
decode_data.filter = filter;
decode_data.pad = pad;
GST_OBJECT_LOCK (filter);
/* Check if this stream exists, if not create a new stream */
gst_buffer_list_foreach (buf_list, validate_buffer_it, &validate_data);
if (!gst_buffer_list_length (buf_list)) {
GST_OBJECT_LOCK (filter);
gst_buffer_list_unref (buf_list);
return GST_FLOW_OK;
}
if (!STREAM_HAS_CRYPTO (stream)) {
GST_OBJECT_UNLOCK (filter);
goto push_out;
}
decode_data.ssrc = ssrc;
decode_data.is_rtcp = is_rtcp;
gst_buffer_list_foreach (buf_list, decode_buffer_it, &decode_data);
GST_OBJECT_UNLOCK (filter);
if (!gst_buffer_list_length (buf_list)) {
gst_buffer_list_unref (buf_list);
return GST_FLOW_OK;
}
/* If all is well, we may have reached soft limit */
if (gst_srtp_get_soft_limit_reached ())
request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);
push_out:
/* Push buffer list to source pad */
if (is_rtcp) {
otherpad = filter->rtcp_srcpad;
if (!filter->rtcp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
filter->rtp_srcpad, TRUE);
} else {
otherpad = filter->rtp_srcpad;
if (!filter->rtp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
filter->rtcp_srcpad, FALSE);
}
GST_LOG_OBJECT (pad, "Pushing buffer chain of %d",
gst_buffer_list_length (buf_list));
ret = gst_pad_push_list (otherpad, buf_list);
return ret;
}
static GstFlowReturn
gst_srtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
@ -1388,21 +1263,6 @@ gst_srtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf)
return gst_srtp_dec_chain (pad, parent, buf, TRUE);
}
static GstFlowReturn
gst_srtp_dec_chain_list_rtp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_dec_chain_list (pad, parent, buf_list, FALSE);
}
static GstFlowReturn
gst_srtp_dec_chain_list_rtcp (GstPad * pad, GstObject * parent,
GstBufferList * buf_list)
{
return gst_srtp_dec_chain_list (pad, parent, buf_list, TRUE);
}
static GstStateChangeReturn
gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
{