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webrtc: produce stats for all relevant streams

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Instead of only using the last ssrc that was pushed into a sink pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>
This commit is contained in:
Matthew Waters 2022-02-21 14:02:52 +11:00 committed by GStreamer Marge Bot
parent 04de1a161f
commit 041eee6c2e
3 changed files with 88 additions and 113 deletions
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34
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -441,37 +441,6 @@ gst_webrtc_bin_pad_init (GstWebRTCBinPad * pad)
{
}
static GstPadProbeReturn
webrtc_bin_pad_buffer_cb (GstPad * pad, GstPadProbeInfo * info,
gpointer user_data)
{
GstWebRTCBinPad *wpad;
GstBuffer *buf;
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
if (info->type & GST_PAD_PROBE_TYPE_BUFFER) {
buf = GST_PAD_PROBE_INFO_BUFFER (info);
} else {
GstBufferList *list;
list = GST_PAD_PROBE_INFO_BUFFER_LIST (info);
buf = gst_buffer_list_get (list, 0);
}
if (buf == NULL)
return GST_PAD_PROBE_OK;
if (!gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf))
return GST_PAD_PROBE_OK;
wpad = GST_WEBRTC_BIN_PAD (pad);
wpad->last_ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
return GST_PAD_PROBE_OK;
}
static GstWebRTCBinPad *
gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
{
@ -493,9 +462,6 @@ gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query);
gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, webrtc_bin_pad_buffer_cb, NULL, NULL);
GST_DEBUG_OBJECT (pad, "new visible pad with direction %s",
direction == GST_PAD_SRC ? "src" : "sink");
return pad;

2
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.h
View file

@ -46,8 +46,6 @@ struct _GstWebRTCBinPad
GstWebRTCRTPTransceiver *trans;
gulong block_id;
guint32 last_ssrc;
GstCaps *received_caps;
};

165
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
View file

@ -697,72 +697,6 @@ _get_stats_from_dtls_transport (GstWebRTCBin * webrtc,
return id;
}
static void
_get_stats_from_transport_channel (GstWebRTCBin * webrtc,
TransportStream * stream, const gchar * codec_id, guint ssrc,
guint clock_rate, GstStructure * s)
{
GstWebRTCDTLSTransport *transport;
GObject *rtp_session;
GObject *gst_rtp_session;
GstStructure *rtp_stats, *twcc_stats;
GValueArray *source_stats;
gchar *transport_id;
double ts;
int i;
gst_structure_get_double (s, "timestamp", &ts);
transport = stream->transport;
if (!transport)
return;
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
stream->session_id, &rtp_session);
g_object_get (rtp_session, "stats", &rtp_stats, NULL);
g_signal_emit_by_name (webrtc->rtpbin, "get-session",
stream->session_id, &gst_rtp_session);
g_object_get (gst_rtp_session, "twcc-stats", &twcc_stats, NULL);
gst_structure_get (rtp_stats, "source-stats", G_TYPE_VALUE_ARRAY,
&source_stats, NULL);
GST_DEBUG_OBJECT (webrtc, "retrieving rtp stream stats from transport %"
GST_PTR_FORMAT " rtp session %" GST_PTR_FORMAT " with %u rtp sources, "
"transport %" GST_PTR_FORMAT, stream, rtp_session, source_stats->n_values,
transport);
transport_id =
_get_stats_from_dtls_transport (webrtc, transport, twcc_stats, s);
/* construct stats objects */
for (i = 0; i < source_stats->n_values; i++) {
const GstStructure *stats;
const GValue *val = g_value_array_get_nth (source_stats, i);
guint stats_ssrc = 0;
stats = gst_value_get_structure (val);
/* skip foreign sources */
if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
ssrc == stats_ssrc)
_get_stats_from_rtp_source_stats (webrtc, stream, stats, codec_id,
transport_id, s);
else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
ssrc == stats_ssrc)
_get_stats_from_remote_rtp_source_stats (webrtc, stream, stats, ssrc,
clock_rate, codec_id, transport_id, s);
}
g_object_unref (rtp_session);
g_object_unref (gst_rtp_session);
gst_structure_free (rtp_stats);
if (twcc_stats)
gst_structure_free (twcc_stats);
g_value_array_free (source_stats);
g_free (transport_id);
}
/* https://www.w3.org/TR/webrtc-stats/#codec-dict* */
static gboolean
_get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
@ -860,33 +794,110 @@ _get_codec_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad,
return has_caps_ssrc;
}
struct transport_stream_stats
{
GstWebRTCBin *webrtc;
TransportStream *stream;
char *transport_id;
char *codec_id;
guint clock_rate;
GValueArray *source_stats;
GstStructure *s;
};
static gboolean
webrtc_stats_get_from_transport (SsrcMapItem * entry,
struct transport_stream_stats *ts_stats)
{
double ts;
int i;
gst_structure_get_double (ts_stats->s, "timestamp", &ts);
/* construct stats objects */
for (i = 0; i < ts_stats->source_stats->n_values; i++) {
const GstStructure *stats;
const GValue *val = g_value_array_get_nth (ts_stats->source_stats, i);
guint stats_ssrc = 0;
stats = gst_value_get_structure (val);
/* skip foreign sources */
if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
entry->ssrc == stats_ssrc)
_get_stats_from_rtp_source_stats (ts_stats->webrtc, ts_stats->stream,
stats, ts_stats->codec_id, ts_stats->transport_id, ts_stats->s);
else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
entry->ssrc == stats_ssrc)
_get_stats_from_remote_rtp_source_stats (ts_stats->webrtc,
ts_stats->stream, stats, entry->ssrc, ts_stats->clock_rate,
ts_stats->codec_id, ts_stats->transport_id, ts_stats->s);
}
/* we want to look at all the entries */
return FALSE;
}
static gboolean
_get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
TransportStream *stream;
gchar *codec_id;
struct transport_stream_stats ts_stats = { NULL, };
guint ssrc, clock_rate;
gboolean has_caps_ssrc;
GObject *rtp_session;
GObject *gst_rtp_session;
GstStructure *rtp_stats, *twcc_stats;
has_caps_ssrc = _get_codec_stats_from_pad (webrtc, pad, s, &codec_id, &ssrc,
_get_codec_stats_from_pad (webrtc, pad, s, &ts_stats.codec_id, &ssrc,
&clock_rate);
if (!wpad->trans)
goto out;
stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream;
if (!stream)
ts_stats.stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream;
if (!ts_stats.stream)
goto out;
if (!has_caps_ssrc)
ssrc = wpad->last_ssrc;
if (wpad->trans->mline == G_MAXUINT)
goto out;
_get_stats_from_transport_channel (webrtc, stream, codec_id, ssrc,
clock_rate, s);
if (!ts_stats.stream->transport)
goto out;
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
ts_stats.stream->session_id, &rtp_session);
g_object_get (rtp_session, "stats", &rtp_stats, NULL);
g_signal_emit_by_name (webrtc->rtpbin, "get-session",
ts_stats.stream->session_id, &gst_rtp_session);
g_object_get (gst_rtp_session, "twcc-stats", &twcc_stats, NULL);
gst_structure_get (rtp_stats, "source-stats", G_TYPE_VALUE_ARRAY,
&ts_stats.source_stats, NULL);
ts_stats.transport_id =
_get_stats_from_dtls_transport (webrtc, ts_stats.stream->transport,
twcc_stats, s);
GST_DEBUG_OBJECT (webrtc, "retrieving rtp stream stats from transport %"
GST_PTR_FORMAT " rtp session %" GST_PTR_FORMAT " with %u rtp sources, "
"transport %" GST_PTR_FORMAT, ts_stats.stream, rtp_session,
ts_stats.source_stats->n_values, ts_stats.stream->transport);
ts_stats.s = s;
transport_stream_find_ssrc_map_item (ts_stats.stream, &ts_stats,
(FindSsrcMapFunc) webrtc_stats_get_from_transport);
g_clear_object (&rtp_session);
g_clear_object (&gst_rtp_session);
gst_clear_structure (&rtp_stats);
gst_clear_structure (&twcc_stats);
g_value_array_free (ts_stats.source_stats);
ts_stats.source_stats = NULL;
g_clear_pointer (&ts_stats.transport_id, g_free);
out:
g_free (codec_id);
g_clear_pointer (&ts_stats.codec_id, g_free);
return TRUE;
}