| /third_party/gstreamer/gstplugins_base/gst-libs/gst/rtp/ |
| D | gstrtpbasepayload.c | 144 static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload,
149 rtpbasepayload, GstPad * pad, GstCaps * filter);
152 rtpbasepayload, GstEvent * event);
156 rtpbasepayload, GstEvent * event);
160 rtpbasepayload, GstPad * pad, GstQuery * query);
507 gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class) in gst_rtp_base_payload_init() argument
512 rtpbasepayload->priv = priv = in gst_rtp_base_payload_init()
513 gst_rtp_base_payload_get_instance_private (rtpbasepayload); in gst_rtp_base_payload_init()
519 rtpbasepayload->srcpad = gst_pad_new_from_template (templ, "src"); in gst_rtp_base_payload_init()
520 gst_pad_set_event_function (rtpbasepayload->srcpad, in gst_rtp_base_payload_init()
[all …]
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| D | gstrtpbasedepayload.c | 127 static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
133 rtpbasepayload, GstRTPHeaderExtension * ext);
135 rtpbasepayload);
1122 gst_rtp_base_depayload_add_extension (GstRTPBaseDepayload * rtpbasepayload, in gst_rtp_base_depayload_add_extension() argument
1129 GST_OBJECT_LOCK (rtpbasepayload); in gst_rtp_base_depayload_add_extension()
1130 g_ptr_array_add (rtpbasepayload->priv->header_exts, gst_object_ref (ext)); in gst_rtp_base_depayload_add_extension()
1131 GST_OBJECT_UNLOCK (rtpbasepayload); in gst_rtp_base_depayload_add_extension()
1135 gst_rtp_base_depayload_clear_extensions (GstRTPBaseDepayload * rtpbasepayload) in gst_rtp_base_depayload_clear_extensions() argument
1137 GST_OBJECT_LOCK (rtpbasepayload); in gst_rtp_base_depayload_clear_extensions()
1138 g_ptr_array_set_size (rtpbasepayload->priv->header_exts, 0); in gst_rtp_base_depayload_clear_extensions()
[all …]
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| D | gstrtpbaseaudiopayload.c | 951 GstRTPBaseAudioPayload *rtpbasepayload; in gst_rtp_base_payload_audio_change_state() local
954 rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element); in gst_rtp_base_payload_audio_change_state()
958 rtpbasepayload->priv->cached_mtu = -1; in gst_rtp_base_payload_audio_change_state()
959 rtpbasepayload->priv->last_rtptime = -1; in gst_rtp_base_payload_audio_change_state()
960 rtpbasepayload->priv->last_timestamp = -1; in gst_rtp_base_payload_audio_change_state()
970 gst_adapter_clear (rtpbasepayload->priv->adapter); in gst_rtp_base_payload_audio_change_state()
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| /third_party/gstreamer/gstplugins_good/gst/rtp/ |
| D | gstrtpsirenpay.c | 87 GstRTPBasePayload *rtpbasepayload; in gst_rtp_siren_pay_init() local
90 rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpsirenpay); in gst_rtp_siren_pay_init()
95 rtpbasepayload->clock_rate = 16000; in gst_rtp_siren_pay_init()
102 gst_rtp_siren_pay_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps) in gst_rtp_siren_pay_setcaps() argument
110 rtpsirenpay = GST_RTP_SIREN_PAY (rtpbasepayload); in gst_rtp_siren_pay_setcaps()
111 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_siren_pay_setcaps()
123 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "SIREN", in gst_rtp_siren_pay_setcaps()
128 return gst_rtp_base_payload_set_outcaps (rtpbasepayload, NULL); in gst_rtp_siren_pay_setcaps()
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| D | gstrtpilbcpay.c | 92 GstRTPBasePayload *rtpbasepayload; in gst_rtp_ilbc_pay_init() local
95 rtpbasepayload = GST_RTP_BASE_PAYLOAD (rtpilbcpay); in gst_rtp_ilbc_pay_init()
100 rtpbasepayload->clock_rate = 8000; in gst_rtp_ilbc_pay_init()
109 gst_rtp_ilbc_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, in gst_rtp_ilbc_pay_sink_setcaps() argument
120 rtpilbcpay = GST_RTP_ILBC_PAY (rtpbasepayload); in gst_rtp_ilbc_pay_sink_setcaps()
121 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_ilbc_pay_sink_setcaps()
135 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "ILBC", in gst_rtp_ilbc_pay_sink_setcaps()
143 gst_rtp_base_payload_set_outcaps (rtpbasepayload, "mode", G_TYPE_STRING, in gst_rtp_ilbc_pay_sink_setcaps()
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| D | gstrtpbvpay.c | 116 gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps) in gst_rtp_bv_pay_sink_setcaps() argument
124 rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload); in gst_rtp_bv_pay_sink_setcaps()
125 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_bv_pay_sink_setcaps()
140 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16", in gst_rtp_bv_pay_sink_setcaps()
142 rtpbasepayload->clock_rate = 8000; in gst_rtp_bv_pay_sink_setcaps()
144 gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32", in gst_rtp_bv_pay_sink_setcaps()
146 rtpbasepayload->clock_rate = 16000; in gst_rtp_bv_pay_sink_setcaps()
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| D | gstrtpklvpay.h | 46 GstRTPBasePayload rtpbasepayload; member
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| /third_party/gstreamer/gstplugins_base/tests/check/ |
| D | meson.build | 21 [ 'libs/rtpbasepayload.c' ],
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| /third_party/gstreamer/gstplugins_bad/ |
| D | NEWS | 1935 - rtpbasepayload: Copy all buffer metadata instead of just GstMetas
2092 - rtpbasepayload: various header extension handling fixes
2145 - rtpbasepayload: Don’t write header extensions if there’s no
2147 - rtpbasepayload: always store input buffer meta before negotiation
2148 - rtpbasepayload: fix transfer annotation for push and push_list
2273 - rtpbasepayload: fix transfer annotation for push and push_list
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| /third_party/gstreamer/gstplugins_base/ |
| D | NEWS | 1935 - rtpbasepayload: Copy all buffer metadata instead of just GstMetas
2092 - rtpbasepayload: various header extension handling fixes
2145 - rtpbasepayload: Don’t write header extensions if there’s no
2147 - rtpbasepayload: always store input buffer meta before negotiation
2148 - rtpbasepayload: fix transfer annotation for push and push_list
2273 - rtpbasepayload: fix transfer annotation for push and push_list
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| D | ChangeLog | 247 rtpbasepayload: always store input buffer meta before negotiation
387 …rtpbasepayload: Don't write header extensions if there's no corresponding input buffer for the pac…
441 rtpbasepayload: fix transfer annotation for push and push_list
588 * tests/check/libs/rtpbasepayload.c:
589 rtpbasepayload: Remove dead twcc code
660 rtpbasepayload: Copy all buffer metadata instead of just GstMetas for the input meta buffer
1455 * tests/check/libs/rtpbasepayload.c:
2765 rtpbasepayload: don't write empty extension header
2783 rtpbasepayload: map RTP buffer READWRITE when setting headers
5204 Same property as the one I just added on rtpbasepayload.
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| /third_party/gstreamer/gstplugins_good/ |
| D | NEWS | 1935 - rtpbasepayload: Copy all buffer metadata instead of just GstMetas
2092 - rtpbasepayload: various header extension handling fixes
2145 - rtpbasepayload: Don’t write header extensions if there’s no
2147 - rtpbasepayload: always store input buffer meta before negotiation
2148 - rtpbasepayload: fix transfer annotation for push and push_list
2273 - rtpbasepayload: fix transfer annotation for push and push_list
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| /third_party/gstreamer/gstreamer/ |
| D | NEWS | 1935 - rtpbasepayload: Copy all buffer metadata instead of just GstMetas
2092 - rtpbasepayload: various header extension handling fixes
2145 - rtpbasepayload: Don’t write header extensions if there’s no
2147 - rtpbasepayload: always store input buffer meta before negotiation
2148 - rtpbasepayload: fix transfer annotation for push and push_list
2273 - rtpbasepayload: fix transfer annotation for push and push_list
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| /third_party/gstreamer/gst_libav/ |
| D | NEWS | 1935 - rtpbasepayload: Copy all buffer metadata instead of just GstMetas
2092 - rtpbasepayload: various header extension handling fixes
2145 - rtpbasepayload: Don’t write header extensions if there’s no
2147 - rtpbasepayload: always store input buffer meta before negotiation
2148 - rtpbasepayload: fix transfer annotation for push and push_list
2273 - rtpbasepayload: fix transfer annotation for push and push_list
|