diff options
| author | Sebastian Dröge <sebastian@centricular.com> | 2020-08-05 10:41:33 +0300 |
|---|---|---|
| committer | Sebastian Dröge <sebastian@centricular.com> | 2020-08-08 10:08:31 +0300 |
| commit | e9a0307b9429663433ceb4a93f71009c2d178607 (patch) | |
| tree | 268608d1f65d55174f4bc1d6aa473475385b5a14 /gst | |
| parent | 3296a03d73b9794aedf3e040b71fe39f3362bbbd (diff) | |
rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility
We didn't aggregate at all in previous versions and there are apparently
various RTP implementations that don't handle aggregation well at all.
As part of this also document that for RTSP it is recommended to keep it
set to "none" while for WebRTC it should be set to "zero-latency".
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692>
Diffstat (limited to 'gst')
| -rw-r--r-- | gst/rtp/gstrtph264pay.c | 17 | ||||
| -rw-r--r-- | gst/rtp/gstrtph265pay.c | 15 |
2 files changed, 29 insertions, 3 deletions
diff --git a/gst/rtp/gstrtph264pay.c b/gst/rtp/gstrtph264pay.c index 01a3659b1..67353283e 100644 --- a/gst/rtp/gstrtph264pay.c +++ b/gst/rtp/gstrtph264pay.c @@ -99,7 +99,7 @@ GST_STATIC_PAD_TEMPLATE ("src", #define DEFAULT_SPROP_PARAMETER_SETS NULL #define DEFAULT_CONFIG_INTERVAL 0 -#define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_ZERO_LATENCY +#define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_NONE enum { @@ -167,12 +167,25 @@ gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); + /** + * GstRtpH264Pay:aggregate-mode + * + * Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets. + * + * This can potentially reduce RTP packetization overhead but not all + * RTP implementations handle it correctly. + * + * For best compatibility, it is recommended to set this to "none" (the + * default) for RTSP and for WebRTC to "zero-latency". + * + * Since: 1.18 + */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_AGGREGATE_MODE, g_param_spec_enum ("aggregate-mode", "Attempt to use aggregate packets", "Bundle suitable SPS/PPS NAL units into STAP-A " - "aggregate packets. ", + "aggregate packets", GST_TYPE_RTP_H264_AGGREGATE_MODE, DEFAULT_AGGREGATE_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); diff --git a/gst/rtp/gstrtph265pay.c b/gst/rtp/gstrtph265pay.c index fc11e60fa..3793ad613 100644 --- a/gst/rtp/gstrtph265pay.c +++ b/gst/rtp/gstrtph265pay.c @@ -133,7 +133,7 @@ GST_STATIC_PAD_TEMPLATE ("src", ); #define DEFAULT_CONFIG_INTERVAL 0 -#define DEFAULT_AGGREGATE_MODE GST_RTP_H265_AGGREGATE_ZERO_LATENCY +#define DEFAULT_AGGREGATE_MODE GST_RTP_H265_AGGREGATE_NONE enum { @@ -192,6 +192,19 @@ gst_rtp_h265_pay_class_init (GstRtpH265PayClass * klass) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); + /** + * GstRtpH265Pay:aggregate-mode + * + * Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets. + * + * This can potentially reduce RTP packetization overhead but not all + * RTP implementations handle it correctly. + * + * For best compatibility, it is recommended to set this to "none" (the + * default) for RTSP and for WebRTC to "zero-latency". + * + * Since: 1.18 + */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_AGGREGATE_MODE, g_param_spec_enum ("aggregate-mode", |