| /third_party/gstreamer/gstplugins_base/gst-libs/gst/rtp/ |
| D | gstrtpbaseaudiopayload.c | 275 rtpbaseaudiopayload) in gst_rtp_base_audio_payload_set_frame_based()
277 g_return_if_fail (rtpbaseaudiopayload != NULL); in gst_rtp_base_audio_payload_set_frame_based()
278 g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL); in gst_rtp_base_audio_payload_set_frame_based()
279 g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL); in gst_rtp_base_audio_payload_set_frame_based()
280 g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL); in gst_rtp_base_audio_payload_set_frame_based()
282 rtpbaseaudiopayload->priv->bytes_to_time = in gst_rtp_base_audio_payload_set_frame_based()
284 rtpbaseaudiopayload->priv->bytes_to_rtptime = in gst_rtp_base_audio_payload_set_frame_based()
286 rtpbaseaudiopayload->priv->time_to_bytes = in gst_rtp_base_audio_payload_set_frame_based()
299 rtpbaseaudiopayload) in gst_rtp_base_audio_payload_set_sample_based()
301 g_return_if_fail (rtpbaseaudiopayload != NULL); in gst_rtp_base_audio_payload_set_sample_based()
[all …]
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| D | gstrtpbaseaudiopayload.h | 85 … gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
88 … gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
94 … gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
97 … gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
101 … gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
107 … gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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| /third_party/gstreamer/gstplugins_good/gst/rtp/ |
| D | gstrtpsirenpay.c | 88 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_siren_pay_init() local
91 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpsirenpay); in gst_rtp_siren_pay_init()
98 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload); in gst_rtp_siren_pay_init()
105 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_siren_pay_setcaps() local
111 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_siren_pay_setcaps()
126 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, 20, 40); in gst_rtp_siren_pay_setcaps()
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| D | gstrtpg726pay.c | 123 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_g726_pay_init() local
125 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg726pay); in gst_rtp_g726_pay_init()
132 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_g726_pay_init()
140 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_g726_pay_setcaps() local
145 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (payload); in gst_rtp_g726_pay_setcaps()
161 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, in gst_rtp_g726_pay_setcaps()
166 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, in gst_rtp_g726_pay_setcaps()
171 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, in gst_rtp_g726_pay_setcaps()
176 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, in gst_rtp_g726_pay_setcaps()
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| D | gstrtpbvpay.c | 105 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_bv_pay_init() local
107 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay); in gst_rtp_bv_pay_init()
112 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload); in gst_rtp_bv_pay_init()
119 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_bv_pay_sink_setcaps() local
125 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_bv_pay_sink_setcaps()
150 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, in gst_rtp_bv_pay_sink_setcaps()
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| D | gstrtpg722pay.c | 100 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_g722_pay_init() local
102 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg722pay); in gst_rtp_g722_pay_init()
107 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_g722_pay_init()
122 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_g722_pay_setcaps() local
124 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); in gst_rtp_g722_pay_setcaps()
184 gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload, in gst_rtp_g722_pay_setcaps()
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| D | gstrtpilbcpay.c | 93 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_ilbc_pay_init() local
96 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpilbcpay); in gst_rtp_ilbc_pay_init()
105 gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload); in gst_rtp_ilbc_pay_init()
113 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_ilbc_pay_sink_setcaps() local
121 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload); in gst_rtp_ilbc_pay_sink_setcaps()
138 gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload, in gst_rtp_ilbc_pay_sink_setcaps()
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| D | gstrtppcmapay.c | 88 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_pcma_pay_init() local
90 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmapay); in gst_rtp_pcma_pay_init()
96 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_pcma_pay_init()
99 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1); in gst_rtp_pcma_pay_init()
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| D | gstrtppcmupay.c | 88 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_pcmu_pay_init() local
90 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtppcmupay); in gst_rtp_pcmu_pay_init()
96 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_pcmu_pay_init()
99 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, 1); in gst_rtp_pcmu_pay_init()
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| D | gstrtpL24pay.c | 117 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L24_pay_init() local
119 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL24pay); in gst_rtp_L24_pay_init()
122 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_L24_pay_init()
133 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L24_pay_setcaps() local
135 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); in gst_rtp_L24_pay_setcaps()
168 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, in gst_rtp_L24_pay_setcaps()
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| D | gstrtpL8pay.c | 118 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L8_pay_init() local
120 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL8pay); in gst_rtp_L8_pay_init()
123 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_L8_pay_init()
134 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L8_pay_setcaps() local
136 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); in gst_rtp_L8_pay_setcaps()
169 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, in gst_rtp_L8_pay_setcaps()
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| D | gstrtpL16pay.c | 127 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L16_pay_init() local
129 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpL16pay); in gst_rtp_L16_pay_init()
132 gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload); in gst_rtp_L16_pay_init()
143 GstRTPBaseAudioPayload *rtpbaseaudiopayload; in gst_rtp_L16_pay_setcaps() local
145 rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload); in gst_rtp_L16_pay_setcaps()
178 gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload, in gst_rtp_L16_pay_setcaps()
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| /third_party/gstreamer/gstplugins_base/ |
| D | ChangeLog | 56204 rtpbaseaudiopayload: Copy metadata in the (de)payloader, but only the relevant ones
57089 rtpbaseaudiopayload: Don't copy memory if not needed, just append payload to the RTP buffer
57752 rtpbaseaudiopayload: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
73009 rtpbaseaudiopayload: Avoid copying the data
82481 rtpbaseaudiopayload: add transfer annotation to get_adapter return
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