Searched refs:rtpgsmpay (Results 1 – 4 of 4) sorted by relevance
/third_party/gstreamer/gstplugins_good/gst/rtp/ |
D | gstrtpgsmpay.c | 65 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpgsmpay, "rtpgsmpay", 95 gst_rtp_gsm_pay_init (GstRTPGSMPay * rtpgsmpay) in gst_rtp_gsm_pay_init() argument 97 GST_RTP_BASE_PAYLOAD (rtpgsmpay)->clock_rate = 8000; in gst_rtp_gsm_pay_init() 98 GST_RTP_BASE_PAYLOAD_PT (rtpgsmpay) = GST_RTP_PAYLOAD_GSM; in gst_rtp_gsm_pay_init() 133 GstRTPGSMPay *rtpgsmpay; in gst_rtp_gsm_pay_handle_buffer() local 139 rtpgsmpay = GST_RTP_GSM_PAY (basepayload); in gst_rtp_gsm_pay_handle_buffer() 148 if (payload_len > GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay)) in gst_rtp_gsm_pay_handle_buffer() 157 gst_rtp_copy_audio_meta (rtpgsmpay, outbuf, buffer); in gst_rtp_gsm_pay_handle_buffer() 172 GST_ELEMENT_ERROR (rtpgsmpay, STREAM, ENCODE, (NULL), in gst_rtp_gsm_pay_handle_buffer() 174 GST_RTP_BASE_PAYLOAD_MTU (rtpgsmpay))); in gst_rtp_gsm_pay_handle_buffer()
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D | gstrtpelements.h | 55 GST_ELEMENT_REGISTER_DECLARE (rtpgsmpay);
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D | gstrtp.c | 55 ret |= GST_ELEMENT_REGISTER (rtpgsmpay, plugin); in plugin_init()
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/third_party/gstreamer/gstplugins_good/ |
D | ChangeLog | 24047 rtpgsmpay: fix accidental garbage data before actual payload 24053 gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink 40913 rtpgsmpay: Remove non-existing includes for now 40924 rtpgsmpay: Attach payload to the output buffer instead of copying it 86772 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling 86783 gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
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